Big delay in audio after answer queue call

December 6, 2018, 12:01 am

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@danial wrote:

Hi
I 'm using Asterisk 13.20 and Freepbx 10.13 on a virtual server(8 cores CPU , 16 GB RAM ) we have multi queue and call recording
here is my problem: When queue increase in call volume (about 20 waiting calls ) calls ditributed normally but after agents answerd calls , there will be long delays in hearing the caller’s voice.

In log files there is a related warning message:
[2018-12-05 15:14:30] WARNING[31027][C-0000c3f0] channel.c: Exceptionally long voice queue length queuing to Local/1057@from-queue-0000fa42;1

more info :
-we have two servers with same hardware and software config and both of them have this issue .
-these servers connected to remote mysql (LAN server) by ODBC to store CDRs.
-All clients and servers are on LAN network without firewall or NAT devices , we don’t use DNS for client registering or trunks.
-Asterisk still working with no crash or core dump files.

in same time “queue show” command responses with delay:

2018 has 18 calls (max unlimited) in ‘rrmemory’ strategy (218s holdtime, 72s talktime), W:6, C:962, A:159, SL:85.4% within 60s
Members:
1162 (Local/1162@from-queue/n from sip/20050) (ringinuse disabled) (dynamic) (In use) has taken 11 calls (last was 110 secs ago)
1123 (Local/1123@from-queue/n from sip/20058) (ringinuse disabled) (dynamic) (Not in use) has taken 13 calls (last was 110 secs ago)
1113 (Local/1113@from-queue/n from sip/20044) (ringinuse disabled) (dynamic) (in call) (In use) has taken 4 calls (last was 32 secs ago)
1191 (Local/1191@from-queue/n from sip/20055) (ringinuse disabled) (dynamic) (in call) (In use) has taken 5 calls (last was 261 secs ago)
1057 (Local/1057@from-queue/n from sip/20054) (ringinuse disabled) (dynamic) (In use) has taken 8 calls (last was 494 secs ago)
1114 (Local/1114@from-queue/n from sip/20053) (ringinuse disabled) (dynamic) (in call) (In use) has taken no calls yet
1128 (Local/1128@from-queue/n from sip/20052) (ringinuse disabled) (dynamic) (in call) (In use) has taken 45 calls (last was 117 secs ago)
1189 (Local/1189@from-queue/n from sip/20059) (ringinuse disabled) (dynamic) (In use) has taken 16 calls (last was 274 secs ago)
1030 (Local/1030@from-queue/n from sip/20051) (ringinuse disabled) (dynamic) (In use) has taken 15 calls (last was 152 secs ago)
1364 (Local/1364@from-queue/n from sip/) (ringinuse disabled) (dynamic) (Invalid) has taken no calls yet
1115 (Local/1115@from-queue/n from sip/20062) (ringinuse disabled) (dynamic) (in call) (In use) has taken 17 calls (last was 70 secs ago)
Callers:

SIP/kam-dispatch-0001c313 (wait: 3:33, prio: 0)

SIP/kam-dispatch-0001c317 (wait: 3:22, prio: 0)

SIP/kam-dispatch-0001c318 (wait: 3:13, prio: 0)

SIP/kam-dispatch-0001c31f (wait: 2:59, prio: 0)

SIP/kam-dispatch-0001c324 (wait: 2:31, prio: 0)

SIP/kam-dispatch-0001c329 (wait: 2:11, prio: 0)

SIP/kam-dispatch-0001c331 (wait: 1:55, prio: 0)

SIP/kam-dispatch-0001c337 (wait: 1:42, prio: 0)

SIP/kam-dispatch-0001c33b (wait: 1:19, prio: 0)

SIP/kam-dispatch-0001c33c (wait: 1:11, prio: 0)

SIP/kam-dispatch-0001c33e (wait: 1:07, prio: 0)

SIP/kam-dispatch-0001c340 (wait: 1:01, prio: 0)

SIP/kam-dispatch-0001c347 (wait: 0:40, prio: 0)

SIP/kam-dispatch-0001c349 (wait: 0:34, prio: 0)

SIP/kam-dispatch-0001c34a (wait: 0:30, prio: 0)

SIP/kam-dispatch-0001c34b (wait: 0:23, prio: 0)

SIP/kam-dispatch-0001c34d (wait: 0:03, prio: 0)

SIP/kam-dispatch-0001c34f (wait: 0:01, prio: 0)

thanks in advance.

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Using Reverse Proxy breaks Ajax - ajaxRequest declined - Referrer - GET /admin/ajax.php?command=authping HTTP/1.1" 403 43

December 6, 2018, 4:12 am

≫ Next: FreePBX + Cisco 7945 'Hello World'

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@Onedutch wrote:

Hello,

I fresh installed 14.0.5.2 on a CentOS7 system with a dedicated WAN ip with the help of page (newbies are not allowed to add website links)

Connecting to the WAN ip over http (port 80) works flawless. Just like others , i found some sites, i would like to use a reveser proxy. So a client connects to an FQDN freepbx.WEBSITE (notice the https) this is the reverse proxy with an other WAN ip as the freepbx CentOS host. The Reverse proxy forwards the request to the CentOS WAN ip over http 80. CentOS only accepts request from the reverse proxy IP.

Logging on works fine, after logging on You see the red bar at the top of the screen "ajaxRequest declined - Referrer " in the apache2 access log you will find a 403 (denied) GET /admin/ajax.php?command=authping " 403 43

I enabled freepbx.WEBSITE (so port 80 no ssl) no change, didn’t work either. Consulting the other webpages with similair issues i did find a answer. On the page ( community-dot-freepbx-dot-org/t/can-one-disable-freepbxs-bad-referrer-check/22136/3) SkykingOH mentions a setting on the “advanced settings module” unsure which setting.

After reading and reading, I think the Ajax component is ‘hardcoded’ expecting a referrer from the localhost CentOS and or the CentOS ip and not the https freepbx.WEBSITE . In the apache2 access log you see other GET requests working flawless /admin/config.php for example.

The apache2 error log mentions : [authz_core:error] [pid 15237] [client IPfromReverseProxy:40205] AH01630: client denied by server configuration: /var/www/html/admin/index.html

Anyone has any clue? Perhaps it has something to do with the ?

Best Regards,
Tom

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FreePBX + Cisco 7945 'Hello World'

December 6, 2018, 5:29 am

≫ Next: Extension in use but not in gui

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@yaronr wrote:

Hi
I’m considering using VOIP at home, instead of PSTN, and I want to do a quick ‘hello world’ test, to get a feel of this approach. I would greatly appreciate some help, as I’m new to this space.
For this test, I’m running FreePBX on a VM (standard installation), on a MacOS. MacOS Firewall disabled, VM running on host networking (no NAT). Seems to be running OK.
I have a Cisco 7945 phone.
So far, I’ve set up a TFTP on a NAS, successfully upgraded the phone (via TFTP) to SIP45.9-4-2SR3-1S.

Below is my SEP.cnf.xml - which according to FTP logs gets served to the phone. I’ve tried to keep it minimal.
I’ve added a user account to FreePBX (as reflected by the cnf.xml).
Phone tries to register, but nothing happens (not even an error) - it keeps trying to register, and I don’t know how to proceed.
Nothing on the phone logs (except failure to update locale - which is because I don’t have the locale files on tftp).

<?xml version="1.0" encoding="UTF-8"?>

<deviceProtocol>SIP</deviceProtocol>
<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword>

<devicePool>
    <dateTimeSetting>
        <dateTemplate>D/M/Y</dateTemplate>
        <timeZone>Jerusalem Standard/Daylight Time</timeZone>
        <ntps>
            <ntp>
                <name>timeserver.iix.net.il</name>
                <ntpMode>Unicast</ntpMode>
            </ntp>
        </ntps>
    </dateTimeSetting>
    <callManagerGroup>
        <members>
            <member priority="0">
                <callManager>
                    <ports>
                        <sipPort>5060</sipPort>
                    </ports>
                    <processNodeName>127.0.0.1</processNodeName> <!-- I've also tried the FreePBX IP -->

                </callManager>
            </member>
        </members>
    </callManagerGroup>

</devicePool>
<sipProfile>
    <natEnabled>false</natEnabled>
    <natAddress></natAddress>

    <sipProxies>
        <backupProxy></backupProxy> 
		<backupProxyPort></backupProxyPort> 
		<emergencyProxy></emergencyProxy> 
		<emergencyProxyPort></emergencyProxyPort> 
		<outboundProxy></outboundProxy> 
		<outboundProxyPort></outboundProxyPort> 
        <registerWithProxy>true</registerWithProxy>
    </sipProxies>
    
    <preferredCodec>g711alaw</preferredCodec>
    <phoneLabel>Y a r o n</phoneLabel>
    
    <sipCallFeatures>
        <cnfJoinEnabled>true</cnfJoinEnabled>
        <callForwardURI>x--serviceuri-cfwdall</callForwardURI>
        <callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
        <callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
        <callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
        <meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
        <abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
        <rfc2543Hold>false</rfc2543Hold>
        <callHoldRingback>2</callHoldRingback>
        <localCfwdEnable>true</localCfwdEnable>
        <semiAttendedTransfer>true</semiAttendedTransfer>
        <anonymousCallBlock>2</anonymousCallBlock>
        <callerIdBlocking>2</callerIdBlocking>
        <dndControl>0</dndControl>
        <remoteCcEnable>true</remoteCcEnable>
    </sipCallFeatures>
    <sipStack>
        <sipInviteRetx>6</sipInviteRetx>
        <sipRetx>10</sipRetx>
        <timerInviteExpires>180</timerInviteExpires>
        <timerRegisterExpires>180</timerRegisterExpires>
        <!-- Force short registration timeout to keep NAT connection alive -->
        <timerRegisterDelta>5</timerRegisterDelta>
        <timerKeepAliveExpires>120</timerKeepAliveExpires>
        <timerSubscribeExpires>120</timerSubscribeExpires>
        <timerSubscribeDelta>5</timerSubscribeDelta>
        <timerT1>500</timerT1>
        <timerT2>4000</timerT2>
        <maxRedirects>70</maxRedirects>
        <remotePartyID>false</remotePartyID>
        <userInfo>None</userInfo>
    </sipStack>
    
    <sipLines>
        <!-- Add lines here -->
        <line button="1">
            <featureID>9</featureID>
            <featureLabel>Line 1</featureLabel>
            <!-- Displays next to Line Number -->
            <name>0001</name>
            <!-- SIP username -->
            <displayName>Line 2 caller ID</displayName>
            <!-- Name to display on outbound caller ID -->
            <contact>1</contact>
            <!-- SIP username again -->
            <proxy>10.1.0.34</proxy>
            <!-- SIP server -->
            <port>5060</port>
            <!-- AUTH -->
            <authName>Yaron</authName>
            <!-- auth SIP username same as <name>-->
            <authPassword>*****</authPassword>
            <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
            <messagesNumber>*97</messagesNumber>              

        </line>
        <line button="2">
            <featureID>2</featureID>
            <featureLabel>Line 2</featureLabel>
            <speedDialNumber>86555</speedDialNumber>
        </line>
    </sipLines>
    <dialTemplate>dialplan.xml</dialTemplate>
</sipProfile>

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Extension in use but not in gui

December 6, 2018, 11:02 am

≫ Next: New FreePBX install "All circuits are busy now" message

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@sentinelace wrote:

I have an extension stuck in the DB. It’s not in the gui, but when I try to add extension 138 for example, it says it’s in use. I ran the following and still have the same result

USE asterisk;
DELETE FROM sip WHERE id = ‘xxx’;
DELETE FROM users WHERE extension = ‘xxx’;
DELETE FROM devices WHERE id = ‘xxx’;

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New FreePBX install "All circuits are busy now" message

December 7, 2018, 1:27 am

≫ Next: Freepbx 14 and Cisco 7940, how can I record calls

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@mpforum wrote:

Hey all,

I’ve set up a new FreePBX install and am finding that whenever I dial an outbound number I get the “All circuits are busy now” message.

I set up a DID number to check that my connection is good and have no issues there with inbound working a treat.

When I try to make a call I get the following in the log file:

[2018-12-07 20:13:02] VERBOSE[3370] pbx_variables.c: Setting global variable ‘SIPDOMAIN’ to ‘192.168.2.99’
[2018-12-07 20:13:02] VERBOSE[3370] netsock2.c: Using SIP RTP Audio TOS bits 184
[2018-12-07 20:13:02] VERBOSE[3370] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.
[2018-12-07 20:13:02] VERBOSE[3370] netsock2.c: Using SIP RTP Audio CoS mark 5
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [011614594444@from-internal:1] Macro(“PJSIP/1000-0000000b”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:1] Set(“PJSIP/1000-0000000b”, “TOUCH_MONITOR=1544173982.11”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:2] Set(“PJSIP/1000-0000000b”, “AMPUSER=1000”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:3] GotoIf(“PJSIP/1000-0000000b”, “0?report”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:4] ExecIf(“PJSIP/1000-0000000b”, “1?Set(REALCALLERIDNUM=1000)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:5] Set(“PJSIP/1000-0000000b”, “AMPUSER=1000”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:6] GotoIf(“PJSIP/1000-0000000b”, “0?limit”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:7] Set(“PJSIP/1000-0000000b”, “AMPUSERCIDNAME=1000”) in new stack
_[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:8] ExecIf(“PJSIP/1000-0000000b”, "0?Set(_CIDMASQUERADING=TRUE)") in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:9] GotoIf(“PJSIP/1000-0000000b”, “0?report”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:10] Set(“PJSIP/1000-0000000b”, “AMPUSERCID=1000”) in new stack
_[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:11] Set(“PJSIP/1000-0000000b”, "_DIAL_OPTIONS=HhTtr") in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:12] Set(“PJSIP/1000-0000000b”, “CALLERID(all)=“1000” <1000>”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:13] GotoIf(“PJSIP/1000-0000000b”, “0?limit”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:14] ExecIf(“PJSIP/1000-0000000b”, “1?Set(GROUP(concurrency_limit)=1000)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:15] ExecIf(“PJSIP/1000-0000000b”, “0?Set(CHANNEL(language)=)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:16] NoOp(“PJSIP/1000-0000000b”, “Macro Depth is 1”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:17] GotoIf(“PJSIP/1000-0000000b”, “1?report2:macroerror”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx_builtins.c: Goto (macro-user-callerid,s,18)
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:18] GotoIf(“PJSIP/1000-0000000b”, “1?continue”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx_builtins.c: Goto (macro-user-callerid,s,37)
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:37] Set(“PJSIP/1000-0000000b”, “CALLERID(number)=1000”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:38] Set(“PJSIP/1000-0000000b”, “CALLERID(name)=1000”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:39] GotoIf(“PJSIP/1000-0000000b”, “0?cnum”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:40] Set(“PJSIP/1000-0000000b”, “CDR(cnam)=1000”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:41] Set(“PJSIP/1000-0000000b”, “CDR(cnum)=1000”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-user-callerid:42] Set(“PJSIP/1000-0000000b”, “CHANNEL(language)=en”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [011614594444@from-internal:2] Gosub(“PJSIP/1000-0000000b”, “sub-record-check,s,1(out,011614594444,dontcare)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@sub-record-check:1] GotoIf(“PJSIP/1000-0000000b”, “0?initialized”) in new stack
_[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@sub-record-check:2] Set(“PJSIP/1000-0000000b”, "_REC_STATUS=INITIALIZED") in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@sub-record-check:3] Set(“PJSIP/1000-0000000b”, “NOW=1544173982”) in new stack
_[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@sub-record-check:4] Set(“PJSIP/1000-0000000b”, "_DAY=07") in new stack
_[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@sub-record-check:5] Set(“PJSIP/1000-0000000b”, "_MONTH=12") in new stack
_[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@sub-record-check:6] Set(“PJSIP/1000-0000000b”, "_YEAR=2018") in new stack
_[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@sub-record-check:7] Set(“PJSIP/1000-0000000b”, "_TIMESTR=20181207-201302") in new stack
_[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@sub-record-check:8] Set(“PJSIP/1000-0000000b”, "_FROMEXTEN=1000") in new stack
_[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@sub-record-check:9] Set(“PJSIP/1000-0000000b”, "_MON_FMT=wav") in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@sub-record-check:10] NoOp(“PJSIP/1000-0000000b”, “Recordings initialized”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@sub-record-check:11] ExecIf(“PJSIP/1000-0000000b”, “0?Set(ARG3=dontcare)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@sub-record-check:12] Set(“PJSIP/1000-0000000b”, “REC_POLICY_MODE_SAVE=”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@sub-record-check:13] ExecIf(“PJSIP/1000-0000000b”, “0?Set(REC_STATUS=NO)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@sub-record-check:14] GotoIf(“PJSIP/1000-0000000b”, “3?checkaction”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx_builtins.c: Goto (sub-record-check,s,17)
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@sub-record-check:17] GotoIf(“PJSIP/1000-0000000b”, “1?sub-record-check,out,1”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx_builtins.c: Goto (sub-record-check,out,1)
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [out@sub-record-check:1] NoOp(“PJSIP/1000-0000000b”, “Outbound Recording Check from 1000 to 011614594444”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [out@sub-record-check:2] Set(“PJSIP/1000-0000000b”, “RECMODE=yes”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [out@sub-record-check:3] ExecIf(“PJSIP/1000-0000000b”, “0?Goto(routewins)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [out@sub-record-check:4] ExecIf(“PJSIP/1000-0000000b”, “0?Goto(routewins)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [out@sub-record-check:5] Gosub(“PJSIP/1000-0000000b”, “recordcheck,1(yes,out,011614594444)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp(“PJSIP/1000-0000000b”, “Starting recording check against yes”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [recordcheck@sub-record-check:2] Goto(“PJSIP/1000-0000000b”, “yes”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx_builtins.c: Goto (sub-record-check,recordcheck,9)
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [recordcheck@sub-record-check:9] ExecIf(“PJSIP/1000-0000000b”, “0?Return()”) in new stack
_[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [recordcheck@sub-record-check:10] Set(“PJSIP/1000-0000000b”, "_REC_POLICY_MODE=YES") in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [recordcheck@sub-record-check:11] Goto(“PJSIP/1000-0000000b”, “startrec”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx_builtins.c: Goto (sub-record-check,recordcheck,16)
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [recordcheck@sub-record-check:16] NoOp(“PJSIP/1000-0000000b”, “Starting recording: out, 011614594444”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [recordcheck@sub-record-check:17] Set(“PJSIP/1000-0000000b”, “AUDIOHOOK_INHERIT(MixMonitor)=yes”) in new stack
[2018-12-07 20:13:02] ERROR[22389][C-00000007] pbx_functions.c: Function AUDIOHOOK_INHERIT not registered
_[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [recordcheck@sub-record-check:18] Set(“PJSIP/1000-0000000b”, "_CALLFILENAME=out-011614594444-1000-20181207-201302-1544173982.11") in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [recordcheck@sub-record-check:19] MixMonitor(“PJSIP/1000-0000000b”, “2018/12/07/out-011614594444-1000-20181207-201302-1544173982.11.wav,abi(LOCAL_MIXMON_ID),”) in new stack
_[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [recordcheck@sub-record-check:20] Set(“PJSIP/1000-0000000b”, "_MIXMON_ID=0x7ff4b0029ab0") in new stack
_[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [recordcheck@sub-record-check:21] Set(“PJSIP/1000-0000000b”, "_RECORD_ID=PJSIP/1000-0000000b") in new stack
_[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [recordcheck@sub-record-check:22] Set(“PJSIP/1000-0000000b”, "_REC_STATUS=RECORDING") in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [recordcheck@sub-record-check:23] Set(“PJSIP/1000-0000000b”, “CDR(recordingfile)=out-011614594444-1000-20181207-201302-1544173982.11.wav”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [recordcheck@sub-record-check:24] Return(“PJSIP/1000-0000000b”, “”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [out@sub-record-check:6] Return(“PJSIP/1000-0000000b”, “”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [011614594444@from-internal:3] ExecIf(“PJSIP/1000-0000000b”, “0 ?Set(CDR(accountcode)=)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [011614594444@from-internal:4] Set(“PJSIP/1000-0000000b”, “MOHCLASS=default”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [011614594444@from-internal:5] ExecIf(“PJSIP/1000-0000000b”, “1?Set(TRUNKCIDOVERRIDE=<131010>)”) in new stack
_[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [011614594444@from-internal:6] Set(“PJSIP/1000-0000000b”, "NODEST=") in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [011614594444@from-internal:7] Macro(“PJSIP/1000-0000000b”, “dialout-trunk,1,011614594444,off”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:1] Set(“PJSIP/1000-0000000b”, “DIAL_TRUNK=1”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:2] ExecIf(“PJSIP/1000-0000000b”, “0?Set(DIAL_OPTIONS=Hhtr)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:3] GosubIf(“PJSIP/1000-0000000b”, “0?sub-pincheck,s,1()”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:4] ExecIf(“PJSIP/1000-0000000b”, “0?Set(CALLERID(num)=1000)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:5] GotoIf(“PJSIP/1000-0000000b”, “0?disabletrunk,1”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:6] Set(“PJSIP/1000-0000000b”, “DIAL_NUMBER=011614594444”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:7] Set(“PJSIP/1000-0000000b”, “DIAL_TRUNK_OPTIONS=HhTtr”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:8] Set(“PJSIP/1000-0000000b”, “OUTBOUND_GROUP=OUT_1”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:9] Set(“PJSIP/1000-0000000b”, “DIAL_TRUNK_OPTIONS=T”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:10] GotoIf(“PJSIP/1000-0000000b”, “0?nomax”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:11] GotoIf(“PJSIP/1000-0000000b”, “0?chanfull”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:12] GotoIf(“PJSIP/1000-0000000b”, “0?skipoutcid”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:13] Macro(“PJSIP/1000-0000000b”, “outbound-callerid,1”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outbound-callerid:1] NoOp(“PJSIP/1000-0000000b”, “1000”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outbound-callerid:2] NoOp(“PJSIP/1000-0000000b”, “”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outbound-callerid:3] NoOp(“PJSIP/1000-0000000b”, “off”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outbound-callerid:4] ExecIf(“PJSIP/1000-0000000b”, “0?Set(CALLERPRES(name-pres)=)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outbound-callerid:5] ExecIf(“PJSIP/1000-0000000b”, “0?Set(CALLERPRES(num-pres)=)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outbound-callerid:6] ExecIf(“PJSIP/1000-0000000b”, “0?Set(REALCALLERIDNUM=1000)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outbound-callerid:7] ExecIf(“PJSIP/1000-0000000b”, “0?Set(AMPUSER=1000)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outbound-callerid:8] GotoIf(“PJSIP/1000-0000000b”, “1?normcid”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx_builtins.c: Goto (macro-outbound-callerid,s,12)
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outbound-callerid:12] Set(“PJSIP/1000-0000000b”, “USEROUTCID=”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outbound-callerid:13] Set(“PJSIP/1000-0000000b”, “EMERGENCYCID=”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outbound-callerid:14] Set(“PJSIP/1000-0000000b”, “TRUNKOUTCID=131010”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outbound-callerid:15] GotoIf(“PJSIP/1000-0000000b”, “1?trunkcid”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx_builtins.c: Goto (macro-outbound-callerid,s,20)
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outbound-callerid:20] ExecIf(“PJSIP/1000-0000000b”, “1?Set(CALLERID(all)=131010)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outbound-callerid:21] ExecIf(“PJSIP/1000-0000000b”, “0?Set(CALLERID(all)=)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outbound-callerid:22] ExecIf(“PJSIP/1000-0000000b”, “1?Set(CALLERID(all)=<131010>)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outbound-callerid:23] ExecIf(“PJSIP/1000-0000000b”, “0?Set(CALLERPRES(name-pres)=prohib_passed_screen)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outbound-callerid:24] ExecIf(“PJSIP/1000-0000000b”, “0?Set(CALLERPRES(num-pres)=prohib_passed_screen)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outbound-callerid:25] Set(“PJSIP/1000-0000000b”, “CDR(outbound_cnum)=131010”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outbound-callerid:26] Set(“PJSIP/1000-0000000b”, “CDR(outbound_cnam)=”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:14] GosubIf(“PJSIP/1000-0000000b”, “0?sub-flp-1,s,1()”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:15] Set(“PJSIP/1000-0000000b”, “OUTNUM=011614594444”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:16] Set(“PJSIP/1000-0000000b”, “custom=PJSIP”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:17] ExecIf(“PJSIP/1000-0000000b”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:18] ExecIf(“PJSIP/1000-0000000b”, “0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:19] Macro(“PJSIP/1000-0000000b”, “dialout-trunk-predial-hook,”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“PJSIP/1000-0000000b”, “”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:20] GotoIf(“PJSIP/1000-0000000b”, “0?skipcrm”) in new stack
_[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:21] Set(“PJSIP/1000-0000000b”, "_CRM_DIRECTION=OUTBOUND") in new stack
_[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:22] Set(“PJSIP/1000-0000000b”, "_CRM_DESTINATION=011614594444") in new stack
_[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:23] Set(“PJSIP/1000-0000000b”, "_CRM_SOURCE=1000") in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:24] AGI(“PJSIP/1000-0000000b”, “sangomacrm.agi”) in new stack
[2018-12-07 20:13:02] VERBOSE[22389][C-00000007] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2018-12-07 20:13:02] VERBOSE[22396][C-00000007] app_mixmonitor.c: Begin MixMonitor Recording PJSIP/1000-0000000b
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] res_agi.c: <PJSIP/1000-0000000b>AGI Script sangomacrm.agi completed, returning 0
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:25] Set(“PJSIP/1000-0000000b”, “CHANNEL(hangup_handler_push)=crm-hangup,s,1”) in new stack
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:26] NoOp(“PJSIP/1000-0000000b”, “CRM Finished”) in new stack
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:27] GotoIf(“PJSIP/1000-0000000b”, “0?bypass,1”) in new stack
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:28] ExecIf(“PJSIP/1000-0000000b”, “1?Set(CONNECTEDLINE(num,i)=011614594444)”) in new stack
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:29] ExecIf(“PJSIP/1000-0000000b”, “1?Set(CONNECTEDLINE(name,i)=CID:131010)”) in new stack
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:30] ExecIf(“PJSIP/1000-0000000b”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)131010)”) in new stack
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:31] GotoIf(“PJSIP/1000-0000000b”, “0?customtrunk”) in new stack
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:32] Dial(“PJSIP/1000-0000000b”, “PJSIP/011614594444@Flow_Route,300,Tb(func-apply-sipheaders^s^1)”) in new stack
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] app_stack.c: PJSIP/Flow_Route-0000000c Internal Gosub(func-apply-sipheaders,s,1) start
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:1] ExecIf(“PJSIP/Flow_Route-0000000c”, “0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:2] NoOp(“PJSIP/Flow_Route-0000000c”, “Applying SIP Headers to channel”) in new stack
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:3] Set(“PJSIP/Flow_Route-0000000c”, “SIPHEADERKEYS=”) in new stack
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:4] ExecIf(“PJSIP/Flow_Route-0000000c”, “0?Set(Rheader=1)”) in new stack
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:5] While(“PJSIP/Flow_Route-0000000c”, “0”) in new stack
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] app_while.c: Jumping to priority 9
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:10] ExecIf(“PJSIP/Flow_Route-0000000c”, “0?SIPRemoveHeader(Alert-Info:)”) in new stack
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:11] ExecIf(“PJSIP/Flow_Route-0000000c”, “0?Set(PJSIP_HEADER(remove,Alert-Info)=)”) in new stack
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:12] Return(“PJSIP/Flow_Route-0000000c”, “”) in new stack
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] app_stack.c: Spawn extension (from-pstn, 011614594444, 1) exited non-zero on ‘PJSIP/Flow_Route-0000000c’
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] app_stack.c: PJSIP/Flow_Route-0000000c Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
[2018-12-07 20:13:05] VERBOSE[22389][C-00000007] app_dial.c: Called PJSIP/011614594444@Flow_Route
[2018-12-07 20:13:06] VERBOSE[22389][C-00000007] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2018-12-07 20:13:06] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:33] NoOp(“PJSIP/1000-0000000b”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21”) in new stack
[2018-12-07 20:13:06] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-dialout-trunk:34] GotoIf(“PJSIP/1000-0000000b”, “0?continue,1:s-CHANUNAVAIL,1”) in new stack
[2018-12-07 20:13:06] VERBOSE[22389][C-00000007] pbx_builtins.c: Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
[2018-12-07 20:13:06] VERBOSE[22389][C-00000007] pbx.c: Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set(“PJSIP/1000-0000000b”, “RC=21”) in new stack
[2018-12-07 20:13:06] VERBOSE[22389][C-00000007] pbx.c: Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto(“PJSIP/1000-0000000b”, “21,1”) in new stack
[2018-12-07 20:13:06] VERBOSE[22389][C-00000007] pbx_builtins.c: Goto (macro-dialout-trunk,21,1)
[2018-12-07 20:13:06] VERBOSE[22389][C-00000007] pbx.c: Executing [21@macro-dialout-trunk:1] Goto(“PJSIP/1000-0000000b”, “continue,1”) in new stack
[2018-12-07 20:13:06] VERBOSE[22389][C-00000007] pbx_builtins.c: Goto (macro-dialout-trunk,continue,1)
[2018-12-07 20:13:06] VERBOSE[22389][C-00000007] pbx.c: Executing [continue@macro-dialout-trunk:1] NoOp(“PJSIP/1000-0000000b”, “TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks”) in new stack
[2018-12-07 20:13:06] VERBOSE[22389][C-00000007] pbx.c: Executing [continue@macro-dialout-trunk:2] ExecIf(“PJSIP/1000-0000000b”, “1?Set(CALLERID(number)=1000)”) in new stack
[2018-12-07 20:13:06] VERBOSE[22389][C-00000007] pbx.c: Executing [011614594444@from-internal:8] Macro(“PJSIP/1000-0000000b”, “outisbusy,”) in new stack
[2018-12-07 20:13:06] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outisbusy:1] Progress(“PJSIP/1000-0000000b”, “”) in new stack
[2018-12-07 20:13:06] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outisbusy:2] GotoIf(“PJSIP/1000-0000000b”, “0?emergency,1”) in new stack
[2018-12-07 20:13:06] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outisbusy:3] GotoIf(“PJSIP/1000-0000000b”, “0?intracompany,1”) in new stack
[2018-12-07 20:13:06] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-outisbusy:4] Playback(“PJSIP/1000-0000000b”, “all-circuits-busy-now&please-try-call-later, noanswer”) in new stack
[2018-12-07 20:13:06] VERBOSE[22389][C-00000007] file.c: <PJSIP/1000-0000000b> Playing ‘all-circuits-busy-now.g722’ (language ‘en’)
[2018-12-07 20:13:08] VERBOSE[22389][C-00000007] file.c: <PJSIP/1000-0000000b> Playing ‘please-try-call-later.ulaw’ (language ‘en’)
[2018-12-07 20:13:09] VERBOSE[22389][C-00000007] pbx.c: Executing [h@from-internal:1] Macro(“PJSIP/1000-0000000b”, “hangupcall”) in new stack
[2018-12-07 20:13:09] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/1000-0000000b”, “1?theend”) in new stack
[2018-12-07 20:13:09] VERBOSE[22389][C-00000007] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2018-12-07 20:13:09] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/1000-0000000b”, “0?Set(CDR(recordingfile)=)”) in new stack
[2018-12-07 20:13:09] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“PJSIP/1000-0000000b”, " monior file= /var/spool/asterisk/monitor/2018/12/07/out-011614594444-1000-20181207-201302-1544173982.11.wav") in new stack
[2018-12-07 20:13:09] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-hangupcall:5] AGI(“PJSIP/1000-0000000b”, “attendedtransfer-rec-restart.php,/var/spool/asterisk/monitor/2018/12/07/out-011614594444-1000-20181207-201302-1544173982.11.wav”) in new stack
[2018-12-07 20:13:09] VERBOSE[22389][C-00000007] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
[2018-12-07 20:13:09] VERBOSE[22389][C-00000007] res_agi.c: <PJSIP/1000-0000000b>AGI Script attendedtransfer-rec-restart.php completed, returning 0
[2018-12-07 20:13:09] VERBOSE[22389][C-00000007] pbx.c: Executing [s@macro-hangupcall:6] Hangup(“PJSIP/1000-0000000b”, “”) in new stack
[2018-12-07 20:13:09] VERBOSE[22389][C-00000007] app_macro.c: Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘PJSIP/1000-0000000b’ in macro ‘hangupcall’
[2018-12-07 20:13:09] VERBOSE[22389][C-00000007] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/1000-0000000b’
[2018-12-07 20:13:09] VERBOSE[22389][C-00000007] app_stack.c: PJSIP/1000-0000000b Internal Gosub(crm-hangup,s,1) start
[2018-12-07 20:13:09] VERBOSE[22389][C-00000007] pbx.c: Executing [s@crm-hangup:1] NoOp(“PJSIP/1000-0000000b”, “Sending Hangup to CRM”) in new stack
[2018-12-07 20:13:09] VERBOSE[22389][C-00000007] pbx.c: Executing [s@crm-hangup:2] NoOp(“PJSIP/1000-0000000b”, “HANGUP CAUSE: 21”) in new stack
_[2018-12-07 20:13:09] VERBOSE[22389][C-00000007] pbx.c: Executing [s@crm-hangup:3] ExecIf(“PJSIP/1000-0000000b”, "0?Set(_CRM_VOICEMAIL=)") in new stack
[2018-12-07 20:13:09] VERBOSE[22389][C-00000007] pbx.c: Executing [s@crm-hangup:4] NoOp(“PJSIP/1000-0000000b”, “MASTER CHANNEL: 1544173982.11 = 1544173982.11”) in new stack
[2018-12-07 20:13:09] VERBOSE[22389][C-00000007] pbx.c: Executing [s@crm-hangup:5] GotoIf(“PJSIP/1000-0000000b”, “0?return”) in new stack
_[2018-12-07 20:13:09] VERBOSE[22389][C-00000007] pbx.c: Executing [s@crm-hangup:6] Set(“PJSIP/1000-0000000b”, "_CRM_HANGUP=1") in new stack
[2018-12-07 20:13:09] VERBOSE[22389][C-00000007] pbx.c: Executing [s@crm-hangup:7] AGI(“PJSIP/1000-0000000b”, “sangomacrm.agi”) in new stack
[2018-12-07 20:13:09] VERBOSE[22389][C-00000007] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2018-12-07 20:13:11] VERBOSE[22389][C-00000007] res_agi.c: <PJSIP/1000-0000000b>AGI Script sangomacrm.agi completed, returning 0
[2018-12-07 20:13:11] VERBOSE[22389][C-00000007] pbx.c: Executing [s@crm-hangup:8] Return(“PJSIP/1000-0000000b”, “”) in new stack
[2018-12-07 20:13:11] VERBOSE[22389][C-00000007] app_stack.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/1000-0000000b’
[2018-12-07 20:13:11] VERBOSE[22389][C-00000007] app_stack.c: PJSIP/1000-0000000b Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
[2018-12-07 20:13:11] VERBOSE[22396][C-00000007] app_mixmonitor.c: MixMonitor close filestream (mixed)
[2018-12-07 20:13:11] VERBOSE[22396][C-00000007] app_mixmonitor.c: End MixMonitor Recording PJSIP/1000-0000000b

The number I have dialed is a random Australian number, but is a valid format.

SIP provider is Flowroute. Using PJSip with the default port (5060). Authentication is Outbound and registration is send.

Dial patters are:
011.
1NXXNXXXXXX
NXXNXXXXXX
NXXXXXX

I’m still fairly new to this so any help would be amazing.

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Freepbx 14 and Cisco 7940, how can I record calls

December 7, 2018, 3:48 am

≫ Next: Remote PJSIP Endpoints Straight to VM

≪ Previous: New FreePBX install "All circuits are busy now" message

@necto_random wrote:

Hellow, I have Freepbx 14, asterisk 13,
I read all documentation, but my calls can’t writing.
I can call to any phones, I register phones with pjsip,
But when i write extensions for my pjsip endpoints, phones when i call do not use extensions(Application->Extensions).
I should record calls, but my calls use only Asterisk.
I need any help.

CDR Report
Tue, Jan 1 2008 11:21 PM CHAN_START necto.random3 505 DEFAULT 507 default PJSIP/505-00000004
Tue, Jan 1 2008 11:21 PM CHAN_START DEFAULT s default PJSIP/507-00000005
Tue, Jan 1 2008 11:21 PM ANSWER 507 DEFAULT 507 default AppDial PJSIP/507-00000005
Tue, Jan 1 2008 11:21 PM ANSWER necto.random3 505 505 507 DEFAULT 507 default Dial PJSIP/505-00000004
Tue, Jan 1 2008 11:21 PM BRIDGE_ENTER 507 DEFAULT default AppDial PJSIP/507-00000005
Tue, Jan 1 2008 11:21 PM BRIDGE_ENTER necto.random3 505 505 507 DEFAULT 507 default Dial PJSIP/505-00000004
Tue, Jan 1 2008 11:21 PM BRIDGE_EXIT 507 DEFAULT default AppDial PJSIP/507-00000005
Tue, Jan 1 2008 11:21 PM HANGUP 507 DEFAULT default AppDial PJSIP/507-00000005
Tue, Jan 1 2008 11:21 PM BRIDGE_EXIT necto.random3 505 505 507 DEFAULT 507 default Dial PJSIP/505-00000004
Tue, Jan 1 2008 11:21 PM CHAN_END 507 DEFAULT default AppDial PJSIP/507-00000005
Tue, Jan 1 2008 11:21 PM HANGUP necto.random3 505 505 507 DEFAULT h default PJSIP/505-00000004
Tue, Jan 1 2008 11:21 PM CHAN_END necto.random3 505 505 507 DEFAULT h default PJSIP/505-00000004
Tue, Jan 1 2008 11:21 PM LINKEDID_END necto.random3 505 505 507 DEFAULT h default PJSIP/505-00000004

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Remote PJSIP Endpoints Straight to VM

December 7, 2018, 10:58 am

≫ Next: Trunk dont register with Vitelity

≪ Previous: Freepbx 14 and Cisco 7940, how can I record calls

@hardocp wrote:

I am running Asterisk 16.0.0 and Freepbx 14.0.5.2

My current setup is as follows – Asterisk is hosted on a cloud VM with a public IP

Asterisk is running fine as are the endpoints

All endpoints connect to Asterisk via PJSIP and i am able to make outbound calls and complete an echo test from each endpoint

Here is an example of what my endpoints look like in the CLI

Endpoint: 121/121 Not in use 0 of inf
InAuth: 121-auth/121
Aor: 121 1
Contact: 121/sip:121@68.195.13.5:47329;transport=TL c0eea25396 Avail 17.906
Transport: 0.0.0.0-tls tls 3 96 0.0.0.0:5061

Fine

My issue is that when i try to call from extension to extension or if i make a call to my DID and then try to connect to an extension the result is straight to VM

Here is the call log:

== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio CoS mark 5
– Executing [125@from-internal:1] GotoIf(“PJSIP/121-0000000d”, “1?ext-local,125,1:followme-check,125,1”) in new stack
– Goto (ext-local,125,1)
– Executing [125@ext-local:1] Set(“PJSIP/121-0000000d”, “__RINGTIMER=15”) in new stack
– Executing [125@ext-local:2] Macro(“PJSIP/121-0000000d”, “exten-vm,125,125,0,0,0”) in new stack
– Executing [s@macro-exten-vm:1] Macro(“PJSIP/121-0000000d”, “user-callerid,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“PJSIP/121-0000000d”, “TOUCH_MONITOR=1544208882.13”) in new stack
– Executing [s@macro-user-callerid:2] Set(“PJSIP/121-0000000d”, “AMPUSER=121”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“PJSIP/121-0000000d”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“PJSIP/121-0000000d”, “1?Set(REALCALLERIDNUM=121)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“PJSIP/121-0000000d”, “AMPUSER=121”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“PJSIP/121-0000000d”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:7] Set(“PJSIP/121-0000000d”, “AMPUSERCIDNAME=xxx”) in new stack
– Executing [s@macro-user-callerid:8] ExecIf(“PJSIP/121-0000000d”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
– Executing [s@macro-user-callerid:9] GotoIf(“PJSIP/121-0000000d”, “0?report”) in new stack
– Executing [s@macro-user-callerid:10] Set(“PJSIP/121-0000000d”, “AMPUSERCID=121”) in new stack
– Executing [s@macro-user-callerid:11] Set(“PJSIP/121-0000000d”, “__DIAL_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-user-callerid:12] Set(“PJSIP/121-0000000d”, “CALLERID(all)=“Asher Toporovsky” <121>”) in new stack
– Executing [s@macro-user-callerid:13] GotoIf(“PJSIP/121-0000000d”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:14] ExecIf(“PJSIP/121-0000000d”, “0?Set(GROUP(concurrency_limit)=121)”) in new stack
– Executing [s@macro-user-callerid:15] NoOp(“PJSIP/121-0000000d”, “Macro Depth is 2”) in new stack
– Executing [s@macro-user-callerid:16] GotoIf(“PJSIP/121-0000000d”, “1?report2:macroerror”) in new stack
– Goto (macro-user-callerid,s,17)
– Executing [s@macro-user-callerid:17] GotoIf(“PJSIP/121-0000000d”, “0?continue”) in new stack
– Executing [s@macro-user-callerid:18] Set(“PJSIP/121-0000000d”, “__TTL=64”) in new stack
– Executing [s@macro-user-callerid:19] GotoIf(“PJSIP/121-0000000d”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,35)
– Executing [s@macro-user-callerid:35] Set(“PJSIP/121-0000000d”, “CALLERID(number)=121”) in new stack
– Executing [s@macro-user-callerid:36] Set(“PJSIP/121-0000000d”, “CALLERID(name)=Asher Toporovsky”) in new stack
– Executing [s@macro-user-callerid:37] GotoIf(“PJSIP/121-0000000d”, “0?cnum”) in new stack
– Executing [s@macro-user-callerid:38] Set(“PJSIP/121-0000000d”, “CDR(cnam)=Asher Toporovsky”) in new stack
– Executing [s@macro-user-callerid:39] Set(“PJSIP/121-0000000d”, “CDR(cnum)=121”) in new stack
– Executing [s@macro-user-callerid:40] Set(“PJSIP/121-0000000d”, “CHANNEL(language)=en”) in new stack
– Executing [s@macro-exten-vm:2] Set(“PJSIP/121-0000000d”, “RingGroupMethod=none”) in new stack
– Executing [s@macro-exten-vm:3] Set(“PJSIP/121-0000000d”, “__EXTTOCALL=125”) in new stack
– Executing [s@macro-exten-vm:4] Set(“PJSIP/121-0000000d”, “__PICKUPMARK=125”) in new stack
– Executing [s@macro-exten-vm:5] Set(“PJSIP/121-0000000d”, “RT=15”) in new stack
– Executing [s@macro-exten-vm:6] Gosub(“PJSIP/121-0000000d”, “sub-record-check,s,1(exten,125,dontcare)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“PJSIP/121-0000000d”, “0?initialized”) in new stack
– Executing [s@sub-record-check:2] Set(“PJSIP/121-0000000d”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:3] Set(“PJSIP/121-0000000d”, “NOW=1544208882”) in new stack
– Executing [s@sub-record-check:4] Set(“PJSIP/121-0000000d”, “__DAY=07”) in new stack
– Executing [s@sub-record-check:5] Set(“PJSIP/121-0000000d”, “__MONTH=12”) in new stack
– Executing [s@sub-record-check:6] Set(“PJSIP/121-0000000d”, “__YEAR=2018”) in new stack
– Executing [s@sub-record-check:7] Set(“PJSIP/121-0000000d”, “__TIMESTR=20181207-135442”) in new stack
– Executing [s@sub-record-check:8] Set(“PJSIP/121-0000000d”, “__FROMEXTEN=121”) in new stack
– Executing [s@sub-record-check:9] Set(“PJSIP/121-0000000d”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:10] NoOp(“PJSIP/121-0000000d”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“PJSIP/121-0000000d”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“PJSIP/121-0000000d”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“PJSIP/121-0000000d”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“PJSIP/121-0000000d”, “5?checkaction”) in new stack
– Goto (sub-record-check,s,17)
– Executing [s@sub-record-check:17] GotoIf(“PJSIP/121-0000000d”, “1?sub-record-check,exten,1”) in new stack
– Goto (sub-record-check,exten,1)
– Executing [exten@sub-record-check:1] NoOp(“PJSIP/121-0000000d”, “Exten Recording Check between 121 and 125”) in new stack
– Executing [exten@sub-record-check:2] Set(“PJSIP/121-0000000d”, “CALLTYPE=internal”) in new stack
– Executing [exten@sub-record-check:3] ExecIf(“PJSIP/121-0000000d”, “0?Set(CALLTYPE=)”) in new stack
– Executing [exten@sub-record-check:4] Set(“PJSIP/121-0000000d”, “CALLEE=dontcare”) in new stack
– Executing [exten@sub-record-check:5] ExecIf(“PJSIP/121-0000000d”, “0?Set(CALLEE=dontcare)”) in new stack
– Executing [exten@sub-record-check:6] GotoIf(“PJSIP/121-0000000d”, “0?callee”) in new stack
– Executing [exten@sub-record-check:7] GotoIf(“PJSIP/121-0000000d”, “1?caller”) in new stack
– Goto (sub-record-check,exten,13)
– Executing [exten@sub-record-check:13] Set(“PJSIP/121-0000000d”, “RECMODE=dontcare”) in new stack
– Executing [exten@sub-record-check:14] ExecIf(“PJSIP/121-0000000d”, “0?Set(RECMODE=dontcare)”) in new stack
– Executing [exten@sub-record-check:15] ExecIf(“PJSIP/121-0000000d”, “1?Set(RECMODE=dontcare)”) in new stack
– Executing [exten@sub-record-check:16] Gosub(“PJSIP/121-0000000d”, “recordcheck,1(dontcare,internal,125)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“PJSIP/121-0000000d”, “Starting recording check against dontcare”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“PJSIP/121-0000000d”, “dontcare”) in new stack
– Goto (sub-record-check,recordcheck,3)
– Executing [recordcheck@sub-record-check:3] Return(“PJSIP/121-0000000d”, “”) in new stack
– Executing [exten@sub-record-check:17] Return(“PJSIP/121-0000000d”, “”) in new stack
– Executing [s@macro-exten-vm:7] GotoIf(“PJSIP/121-0000000d”, “1?macrodial”) in new stack
– Goto (macro-exten-vm,s,13)
– Executing [s@macro-exten-vm:13] GosubIf(“PJSIP/121-0000000d”, “0?clrheader,1()”) in new stack
– Executing [s@macro-exten-vm:14] Macro(“PJSIP/121-0000000d”, “dial-one,15,HhTtr,125”) in new stack
– Executing [s@macro-dial-one:1] Set(“PJSIP/121-0000000d”, “DEXTEN=125”) in new stack
– Executing [s@macro-dial-one:2] ExecIf(“PJSIP/121-0000000d”, “0?Set(__EXTTOCALL=125)”) in new stack
– Executing [s@macro-dial-one:3] Set(“PJSIP/121-0000000d”, “DIALSTATUS_CW=”) in new stack
– Executing [s@macro-dial-one:4] GosubIf(“PJSIP/121-0000000d”, “0?screen,1()”) in new stack
– Executing [s@macro-dial-one:5] GosubIf(“PJSIP/121-0000000d”, “0?cf,1()”) in new stack
– Executing [s@macro-dial-one:6] GotoIf(“PJSIP/121-0000000d”, “1?skip1”) in new stack
– Goto (macro-dial-one,s,9)
– Executing [s@macro-dial-one:9] GotoIf(“PJSIP/121-0000000d”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:10] GotoIf(“PJSIP/121-0000000d”, “0?continue”) in new stack
– Executing [s@macro-dial-one:11] Set(“PJSIP/121-0000000d”, “EXTHASCW=ENABLED”) in new stack
– Executing [s@macro-dial-one:12] GotoIf(“PJSIP/121-0000000d”, “0?next1:cwinusebusy”) in new stack
– Goto (macro-dial-one,s,24)
– Executing [s@macro-dial-one:24] GotoIf(“PJSIP/121-0000000d”, “0?next3:continue”) in new stack
– Goto (macro-dial-one,s,26)
– Executing [s@macro-dial-one:26] GotoIf(“PJSIP/121-0000000d”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:27] GosubIf(“PJSIP/121-0000000d”, “1?dstring,1():dlocal,1()”) in new stack
– Executing [dstring@macro-dial-one:1] Set(“PJSIP/121-0000000d”, “DSTRING=”) in new stack
– Executing [dstring@macro-dial-one:2] Set(“PJSIP/121-0000000d”, “DEVICES=125”) in new stack
– Executing [dstring@macro-dial-one:3] ExecIf(“PJSIP/121-0000000d”, “0?Return()”) in new stack
– Executing [dstring@macro-dial-one:4] ExecIf(“PJSIP/121-0000000d”, “0?Set(DEVICES=25)”) in new stack
– Executing [dstring@macro-dial-one:5] Set(“PJSIP/121-0000000d”, “LOOPCNT=1”) in new stack
– Executing [dstring@macro-dial-one:6] Set(“PJSIP/121-0000000d”, “ITER=1”) in new stack
– Executing [dstring@macro-dial-one:7] Set(“PJSIP/121-0000000d”, “THISDIAL=PJSIP/125”) in new stack
– Executing [dstring@macro-dial-one:8] GosubIf(“PJSIP/121-0000000d”, “1?zap2dahdi,1()”) in new stack
– Executing [zap2dahdi@macro-dial-one:1] ExecIf(“PJSIP/121-0000000d”, “0?Return()”) in new stack
– Executing [zap2dahdi@macro-dial-one:2] Set(“PJSIP/121-0000000d”, “NEWDIAL=”) in new stack
– Executing [zap2dahdi@macro-dial-one:3] Set(“PJSIP/121-0000000d”, “LOOPCNT2=1”) in new stack
– Executing [zap2dahdi@macro-dial-one:4] Set(“PJSIP/121-0000000d”, “ITER2=1”) in new stack
– Executing [zap2dahdi@macro-dial-one:5] Set(“PJSIP/121-0000000d”, “THISPART2=PJSIP/125”) in new stack
– Executing [zap2dahdi@macro-dial-one:6] ExecIf(“PJSIP/121-0000000d”, “0?Set(THISPART2=DAHDIIP/125)”) in new stack
– Executing [zap2dahdi@macro-dial-one:7] Set(“PJSIP/121-0000000d”, “NEWDIAL=PJSIP/125&”) in new stack
– Executing [zap2dahdi@macro-dial-one:8] Set(“PJSIP/121-0000000d”, “ITER2=2”) in new stack
– Executing [zap2dahdi@macro-dial-one:9] GotoIf(“PJSIP/121-0000000d”, “0?begin2”) in new stack
– Executing [zap2dahdi@macro-dial-one:10] Set(“PJSIP/121-0000000d”, “THISDIAL=PJSIP/125”) in new stack
– Executing [zap2dahdi@macro-dial-one:11] Return(“PJSIP/121-0000000d”, “”) in new stack
– Executing [dstring@macro-dial-one:9] GotoIf(“PJSIP/121-0000000d”, “0?docheck”) in new stack
– Executing [dstring@macro-dial-one:10] NoOp(“PJSIP/121-0000000d”, “Debug: Found PJSIP Destination PJSIP/125”) in new stack
– Executing [dstring@macro-dial-one:11] GotoIf(“PJSIP/121-0000000d”, “0?doset”) in new stack
– Executing [dstring@macro-dial-one:12] NoOp(“PJSIP/121-0000000d”, “Debug: Updating PJSIP Destination with PJSIP_DIAL_CONTACTS”) in new stack
– Executing [dstring@macro-dial-one:13] Set(“PJSIP/121-0000000d”, “THISDIAL=PJSIP/125/sip:125@68.195.13.5:62349;transport=TLS”) in new stack
– Executing [dstring@macro-dial-one:14] ExecIf(“PJSIP/121-0000000d”, “0?Set(DIALSTATUS=CHANUNAVAIL)”) in new stack
– Executing [dstring@macro-dial-one:15] GotoIf(“PJSIP/121-0000000d”, “0?skipset”) in new stack
– Executing [dstring@macro-dial-one:16] Set(“PJSIP/121-0000000d”, “DSTRING=PJSIP/125/sip:125@68.195.13.5:62349;transport=TLS&”) in new stack
– Executing [dstring@macro-dial-one:17] Set(“PJSIP/121-0000000d”, “ITER=2”) in new stack
– Executing [dstring@macro-dial-one:18] GotoIf(“PJSIP/121-0000000d”, “0?begin”) in new stack
– Executing [dstring@macro-dial-one:19] ExecIf(“PJSIP/121-0000000d”, “0?Return()”) in new stack
– Executing [dstring@macro-dial-one:20] Set(“PJSIP/121-0000000d”, “DSTRING=PJSIP/125/sip:125@68.195.13.5:62349;transport=TLS”) in new stack
– Executing [dstring@macro-dial-one:21] Return(“PJSIP/121-0000000d”, “”) in new stack
– Executing [s@macro-dial-one:28] GotoIf(“PJSIP/121-0000000d”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:29] GotoIf(“PJSIP/121-0000000d”, “0?skiptrace”) in new stack
– Executing [s@macro-dial-one:30] GosubIf(“PJSIP/121-0000000d”, “1?ctset,1():ctclear,1()”) in new stack
– Executing [ctset@macro-dial-one:1] Set(“PJSIP/121-0000000d”, “DB(CALLTRACE/125)=121”) in new stack
– Executing [ctset@macro-dial-one:2] Return(“PJSIP/121-0000000d”, “”) in new stack
– Executing [s@macro-dial-one:31] Set(“PJSIP/121-0000000d”, “D_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-dial-one:32] GosubIf(“PJSIP/121-0000000d”, “0?func-set-sipheader,s,1(Alert-Info,)”) in new stack
– Executing [s@macro-dial-one:33] NoOp(“PJSIP/121-0000000d”, "Blind Transfer: , Attended Transfer: , User: 121, Alert Info: ") in new stack
– Executing [s@macro-dial-one:34] ExecIf(“PJSIP/121-0000000d”, “1?Set(ALERT_INFO=)”) in new stack
– Executing [s@macro-dial-one:35] ExecIf(“PJSIP/121-0000000d”, “0?Set(ALERT_INFO=)”) in new stack
– Executing [s@macro-dial-one:36] ExecIf(“PJSIP/121-0000000d”, “0?Set(ALERT_INFO=)”) in new stack
– Executing [s@macro-dial-one:37] ExecIf(“PJSIP/121-0000000d”, “0?Set(ALERT_INFO=Normal;volume=)”) in new stack
– Executing [s@macro-dial-one:38] ExecIf(“PJSIP/121-0000000d”, “0?Set(ALERT_INFO=Normal;volume=)”) in new stack
– Executing [s@macro-dial-one:39] GosubIf(“PJSIP/121-0000000d”, “0?func-set-sipheader,s,1(Alert-Info,)”) in new stack
– Executing [s@macro-dial-one:40] ExecIf(“PJSIP/121-0000000d”, “0?Set(CHANNEL(musicclass)=)”) in new stack
– Executing [s@macro-dial-one:41] GosubIf(“PJSIP/121-0000000d”, “0?qwait,1()”) in new stack
– Executing [s@macro-dial-one:42] Set(“PJSIP/121-0000000d”, “__CWIGNORE=”) in new stack
– Executing [s@macro-dial-one:43] Set(“PJSIP/121-0000000d”, “__KEEPCID=TRUE”) in new stack
– Executing [s@macro-dial-one:44] GotoIf(“PJSIP/121-0000000d”, “0?usegoto,1”) in new stack
– Executing [s@macro-dial-one:45] GotoIf(“PJSIP/121-0000000d”, “0?godial”) in new stack
– Executing [s@macro-dial-one:46] Gosub(“PJSIP/121-0000000d”, “sub-presencestate-display,s,1(125)”) in new stack
– Executing [s@sub-presencestate-display:1] Goto(“PJSIP/121-0000000d”, “state-not_set,1”) in new stack
– Goto (sub-presencestate-display,state-not_set,1)
– Executing [state-not_set@sub-presencestate-display:1] Set(“PJSIP/121-0000000d”, “PRESENCESTATE_DISPLAY=”) in new stack
– Executing [state-not_set@sub-presencestate-display:2] Return(“PJSIP/121-0000000d”, “”) in new stack
– Executing [s@macro-dial-one:47] Set(“PJSIP/121-0000000d”, “CONNECTEDLINE(name,i)=Jack Sutton”) in new stack
– Executing [s@macro-dial-one:48] Set(“PJSIP/121-0000000d”, “CONNECTEDLINE(num)=125”) in new stack
– Executing [s@macro-dial-one:49] Set(“PJSIP/121-0000000d”, “D_OPTIONS=HhTtrI”) in new stack
– Executing [s@macro-dial-one:50] Macro(“PJSIP/121-0000000d”, “dialout-one-predial-hook,”) in new stack
– Executing [s@macro-dialout-one-predial-hook:1] MacroExit(“PJSIP/121-0000000d”, “”) in new stack
– Executing [s@macro-dial-one:51] ExecIf(“PJSIP/121-0000000d”, “0?Set(D_OPTIONS=HhtrII)”) in new stack
– Executing [s@macro-dial-one:52] NoOp(“PJSIP/121-0000000d”, “”) in new stack
– Executing [s@macro-dial-one:53] ExecIf(“PJSIP/121-0000000d”, “0?Set(D_OPTIONS=HhTtrIg)”) in new stack
– Executing [s@macro-dial-one:54] Dial(“PJSIP/121-0000000d”, “PJSIP/125/sip:125@68.195.13.5:62349;transport=TLS,15,HhTtrIb(func-apply-sipheaders^s^1)”) in new stack
– PJSIP/125-0000000e Internal Gosub(func-apply-sipheaders,s,1) start
– Executing [s@func-apply-sipheaders:1] NoOp(“PJSIP/125-0000000e”, “Applying SIP Headers to channel”) in new stack
– Executing [s@func-apply-sipheaders:2] Set(“PJSIP/125-0000000e”, “SIPHEADERKEYS=”) in new stack
– Executing [s@func-apply-sipheaders:3] ExecIf(“PJSIP/125-0000000e”, “0?Set(Rheader=1)”) in new stack
– Executing [s@func-apply-sipheaders:4] While(“PJSIP/125-0000000e”, “0”) in new stack
– Jumping to priority 8
– Executing [s@func-apply-sipheaders:9] ExecIf(“PJSIP/125-0000000e”, “0?SIPRemoveHeader(Alert-Info:)”) in new stack
– Executing [s@func-apply-sipheaders:10] ExecIf(“PJSIP/125-0000000e”, “0?Set(PJSIP_HEADER(remove,Alert-Info)=)”) in new stack
– Executing [s@func-apply-sipheaders:11] Return(“PJSIP/125-0000000e”, “”) in new stack
== Spawn extension (from-internal, 125, 1) exited non-zero on ‘PJSIP/125-0000000e’
– PJSIP/125-0000000e Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
– Called PJSIP/125/sip:125@68.195.13.5:62349;transport=TLS
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio CoS mark 5
– Connected line update to PJSIP/121-0000000d prevented.
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [s@macro-dial-one:55] ExecIf(“PJSIP/121-0000000d”, “0?MacroExit()”) in new stack
– Executing [s@macro-dial-one:56] ExecIf(“PJSIP/121-0000000d”, “0?Set(DIALSTATUS=)”) in new stack
– Executing [s@macro-dial-one:57] GosubIf(“PJSIP/121-0000000d”, “0?s-CHANUNAVAIL,1()”) in new stack
– Executing [s@macro-dial-one:58] MacroExit(“PJSIP/121-0000000d”, “”) in new stack
– Executing [s@macro-exten-vm:15] Set(“PJSIP/121-0000000d”, “SV_DIALSTATUS=CHANUNAVAIL”) in new stack
– Executing [s@macro-exten-vm:16] GosubIf(“PJSIP/121-0000000d”, “0?docfu,1()”) in new stack
– Executing [s@macro-exten-vm:17] GosubIf(“PJSIP/121-0000000d”, “0?docfb,1()”) in new stack
– Executing [s@macro-exten-vm:18] Set(“PJSIP/121-0000000d”, “DIALSTATUS=CHANUNAVAIL”) in new stack
– Executing [s@macro-exten-vm:19] ExecIf(“PJSIP/121-0000000d”, “0?MacroExit()”) in new stack
– Executing [s@macro-exten-vm:20] GotoIf(“PJSIP/121-0000000d”, “0?s-CHANUNAVAIL,1”) in new stack
– Executing [s@macro-exten-vm:21] Macro(“PJSIP/121-0000000d”, “vm,125,CHANUNAVAIL,”) in new stack
– Executing [s@macro-vm:1] Macro(“PJSIP/121-0000000d”, “user-callerid,SKIPTTL”) in new stack
– Executing [s@macro-user-callerid:1] Set(“PJSIP/121-0000000d”, “TOUCH_MONITOR=1544208882.13”) in new stack
– Executing [s@macro-user-callerid:2] Set(“PJSIP/121-0000000d”, “AMPUSER=121”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“PJSIP/121-0000000d”, “16?report”) in new stack
– Goto (macro-user-callerid,s,15)
– Executing [s@macro-user-callerid:15] NoOp(“PJSIP/121-0000000d”, “Macro Depth is 3”) in new stack
– Executing [s@macro-user-callerid:16] GotoIf(“PJSIP/121-0000000d”, “1?report2:macroerror”) in new stack
– Goto (macro-user-callerid,s,17)
– Executing [s@macro-user-callerid:17] GotoIf(“PJSIP/121-0000000d”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,35)
– Executing [s@macro-user-callerid:35] Set(“PJSIP/121-0000000d”, “CALLERID(number)=121”) in new stack
– Executing [s@macro-user-callerid:36] Set(“PJSIP/121-0000000d”, “CALLERID(name)=Asher Toporovsky”) in new stack
– Executing [s@macro-user-callerid:37] GotoIf(“PJSIP/121-0000000d”, “0?cnum”) in new stack
– Executing [s@macro-user-callerid:38] Set(“PJSIP/121-0000000d”, “CDR(cnam)=Asher Toporovsky”) in new stack
– Executing [s@macro-user-callerid:39] Set(“PJSIP/121-0000000d”, “CDR(cnum)=121”) in new stack
– Executing [s@macro-user-callerid:40] Set(“PJSIP/121-0000000d”, “CHANNEL(language)=en”) in new stack
– Executing [s@macro-vm:2] Set(“PJSIP/121-0000000d”, “VMGAIN=”) in new stack
– Executing [s@macro-vm:3] Macro(“PJSIP/121-0000000d”, “blkvm-check,”) in new stack
– Executing [s@macro-blkvm-check:1] Set(“PJSIP/121-0000000d”, “GOSUB_RETVAL=”) in new stack
– Executing [s@macro-blkvm-check:2] ExecIf(“PJSIP/121-0000000d”, “0?Set(GOSUB_RETVAL=TRUE)”) in new stack
– Executing [s@macro-blkvm-check:3] MacroExit(“PJSIP/121-0000000d”, “”) in new stack
– Executing [s@macro-vm:4] GotoIf(“PJSIP/121-0000000d”, “1?vmx,1”) in new stack
– Goto (macro-vm,vmx,1)
– Executing [vmx@macro-vm:1] Set(“PJSIP/121-0000000d”, “MEXTEN=125”) in new stack
– Executing [vmx@macro-vm:2] Set(“PJSIP/121-0000000d”, “MMODE=CHANUNAVAIL”) in new stack
– Executing [vmx@macro-vm:3] Set(“PJSIP/121-0000000d”, “RETVM=”) in new stack
– Executing [vmx@macro-vm:4] Set(“PJSIP/121-0000000d”, “MODE=unavail”) in new stack
– Executing [vmx@macro-vm:5] Macro(“PJSIP/121-0000000d”, “get-vmcontext,125”) in new stack
– Executing [s@macro-get-vmcontext:1] Set(“PJSIP/121-0000000d”, “VMCONTEXT=default”) in new stack
– Executing [s@macro-get-vmcontext:2] GotoIf(“PJSIP/121-0000000d”, “0?200:300”) in new stack
– Goto (macro-get-vmcontext,s,300)
– Executing [s@macro-get-vmcontext:300] NoOp(“PJSIP/121-0000000d”, “”) in new stack
– Executing [vmx@macro-vm:6] Set(“PJSIP/121-0000000d”, “MODE=unavail”) in new stack
– Executing [vmx@macro-vm:7] NoOp(“PJSIP/121-0000000d”, “MODE IS: unavail”) in new stack
– Executing [vmx@macro-vm:8] GotoIf(“PJSIP/121-0000000d”, “1?chknomsg”) in new stack
– Goto (macro-vm,vmx,10)
– Executing [vmx@macro-vm:10] GotoIf(“PJSIP/121-0000000d”, “0?s-CHANUNAVAIL,1”) in new stack
– Executing [vmx@macro-vm:11] GotoIf(“PJSIP/121-0000000d”, “1?notdirect”) in new stack
– Goto (macro-vm,vmx,13)
– Executing [vmx@macro-vm:13] NoOp(“PJSIP/121-0000000d”, "Checking if ext 125 is enabled: ") in new stack
– Executing [vmx@macro-vm:14] GotoIf(“PJSIP/121-0000000d”, “1?s-CHANUNAVAIL,1”) in new stack
– Goto (macro-vm,s-CHANUNAVAIL,1)
– Executing [s-CHANUNAVAIL@macro-vm:1] Macro(“PJSIP/121-0000000d”, “get-vmcontext,125”) in new stack
– Executing [s@macro-get-vmcontext:1] Set(“PJSIP/121-0000000d”, “VMCONTEXT=default”) in new stack
– Executing [s@macro-get-vmcontext:2] GotoIf(“PJSIP/121-0000000d”, “0?200:300”) in new stack
– Goto (macro-get-vmcontext,s,300)
– Executing [s@macro-get-vmcontext:300] NoOp(“PJSIP/121-0000000d”, “”) in new stack
– Executing [s-CHANUNAVAIL@macro-vm:2] VoiceMail(“PJSIP/121-0000000d”, “125@default,u”) in new stack
> 0x7ff2bc31e6a0 – Strict RTP learning after remote address set to: 192.168.2.15:12744
> 0x7ff2bc31e6a0 – Strict RTP qualifying stream type: audio
> 0x7ff2bc31e6a0 – Strict RTP switching source address to 68.195.13.5:44971
– <PJSIP/121-0000000d> Playing ‘/var/spool/asterisk/voicemail/default/125/unavail.slin’ (language ‘en’)
> 0x7ff2bc31e6a0 – Strict RTP learning complete - Locking on source address 68.195.13.5:44971
== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on ‘PJSIP/121-0000000d’ in macro ‘vm’
== Spawn extension (macro-exten-vm, s, 21) exited non-zero on ‘PJSIP/121-0000000d’ in macro ‘exten-vm’
== Spawn extension (ext-local, 125, 2) exited non-zero on ‘PJSIP/121-0000000d’
– Executing [h@ext-local:1] Macro(“PJSIP/121-0000000d”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/121-0000000d”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/121-0000000d”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“PJSIP/121-0000000d”, " monior file= ") in new stack
– Executing [s@macro-hangupcall:5] AGI(“PJSIP/121-0000000d”, “attendedtransfer-rec-restart.php,”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
– <PJSIP/121-0000000d>AGI Script attendedtransfer-rec-restart.php completed, returning 0
– Executing [s@macro-hangupcall:6] Hangup(“PJSIP/121-0000000d”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘PJSIP/121-0000000d’ in macro ‘hangupcall’
== Spawn extension (ext-local, h, 1) exited non-zero on ‘PJSIP/121-0000000d’

I am sure its some minor configuration issue in my PJSIP settings – but for the life of me i cannot zero in on it

If someone can help lend an extra hand to get this working i would really appreciate it

thanks

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Trunk dont register with Vitelity

December 7, 2018, 4:44 pm

≫ Next: CallerID HTTP error

≪ Previous: Remote PJSIP Endpoints Straight to VM

@Dhadz wrote:

Hi!!!

I had a problem with a Freepbx 13, i had a trunk with vitelity and voipms.

Vitelity is not registering but voipms does.

this server is behind a router but theres no firewall setting blockingas far is i know.

I used other servers inside router and with public ip and i can register vitelity.

i dunno if there´s something more that can block or have to be missconfigured to block register with an specific server.

AFAIK to regsiter you only need user account, pass, ip service and protocol (5060 in this case)

Any info you need or any suggestion would be apreciated.

Thanks in advance

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CallerID HTTP error

December 8, 2018, 7:53 am

≫ Next: Voicemail notification and MWI erratic behavior

≪ Previous: Trunk dont register with Vitelity

@HenkdeBoer wrote:

Hi!

I am trying to setup the CallerID Lookup Source with our server. It is working, but I get an “” error in the screen, so it is not requesting the url.

When I go to my server, it shows there has been an request with the right url: “GET /phone/index.php?number={NUMBER} HTTP/1.1” 301 517 “-” “asterisk-libcurl-agent/1.0”. And when I go to /etc/asterisk/extensions_additional.conf it also shows the right url:

[cidlookup]
include => cidlookup-custom
exten => cidlookup_1,1,Set(CALLERID(name)=${CURL(http://domain:80/phone/index.php?number=${CALLERID(num)})})
exten => cidlookup_1,n,Return()
exten => cidlookup_return,1,ExecIf($["${DB(cidname/${CALLERID(num)})}" != “”]?Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}))
exten => cidlookup_return,n,Return()

; end of [cidlookup]

When I copy the exact url from above, it shows me the name. What am I doing wrong? Can I see anywhere the exact url that has been requested or get the exact response?

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Voicemail notification and MWI erratic behavior

December 8, 2018, 1:47 pm

≫ Next: When Dialing an outside number XXX-XXX-XXXX I reach the Incoming Greeting

≪ Previous: CallerID HTTP error

@Ollie wrote:

On occasion, an extension will receive a voicemail(s) but will not notify the user either via the screen or MWI. Then, after a period of time the user will receive the notification and the mwi will light up. If the user does not check the new voicemails, the notifications may disappear for a period of time then return. Rebooting the phone will immediately show the new voicemail notifications.

Below is hopefully some useful info pertaining to our setup in relation to this issue. Happy to provide any additional logs or settings. Thanks for looking

PBX Firmware: 12.7.5-1807-1.sng7
Asterisk: 13.22.0
All phones: Yealink T29G firmware 46.83.0.35
We are not running the commercial module Voicemail Notifications.
We have two trunks connected via chan_sip and all extensions are pjsip

Freepbx extension settings:
Extension → Advanced: MWI Subscription Type = Auto

MWI basefile settings for yealink template in EPM:
account.1.subscribe_mwi = line1Mwi
account.1.subscribe_mwi_expires = 3600
account.1.subscribe_mwi_to_vm = 0

screenshot of related t29g phone settings:

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When Dialing an outside number XXX-XXX-XXXX I reach the Incoming Greeting

December 8, 2018, 2:16 pm

≫ Next: SIP ALG problem Powerful Linksys EA900

≪ Previous: Voicemail notification and MWI erratic behavior

@1reason wrote:

When a user dials a number to call someone using an outgoing truck, sometimes instead of hearing the expected ringing of the other party, the incoming IVR greeting. It doesn’t happen every time, albeit it happens almost every day now (maybe 30% or more lately)

What to do?

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SIP ALG problem Powerful Linksys EA900

December 9, 2018, 4:12 am

≫ Next: Responsive Firewall - Driving me Crazy!

≪ Previous: When Dialing an outside number XXX-XXX-XXXX I reach the Incoming Greeting

@cometa_s wrote:

**Linksys Router SIP ALG Problem.

Hello, I have Powerful Linksys EA900 Smart WiFi Router, I m using Elastix 2.5.
From outside External Extension sometime have audio issues oneway audio,
All forums and blogs I see everyone says Disable SIP ALG, when I uncheck (Disable) SIP ALG on my router noway any audio,
I Have forwarded ports to my PBX (SIP UDP 5060-5070) and (RTP UDP 10000-20000)
what can I doo? If SIP ALG Enabled works two-way audio but sometime issues oneway audio,
Also SPI Firewall have disabled.

Thanks**

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Responsive Firewall - Driving me Crazy!

December 9, 2018, 3:34 pm

≫ Next: Custom Context - Include context in other context

≪ Previous: SIP ALG problem Powerful Linksys EA900

@ittechgroup wrote:

Hello all,
I have changed my SIP port to a high number 4xxxx. I use Vitelity as my SIP provider. As long as i use a register string, I am able to connect to Vitelity and have had no problems. I am running out of channels and Vitelity told me that I need to move to IP Authentication. For this i need to change my PORT back to 5060 or forward my 4xxxx port to 5060 only for vitelity’s servers.

I tried to add port forwarding in firewall-4.rules. The port forwarding for 5060 doesn’t work because i keep getting a busy signal for incoming calls. Is this even possible with FreePBX 13 built in firewall?

If port forwarding isn’t going to work, I read that port 5060 can be configured to only respond to DNS string. So if i can setup the firewall to only respond to mypbxHost.domain.com instead of an IP address, this should prevent all the hackers from finding my pbx server by ip scans.

Changing the port to a high port number really lowered the number of attacks on my pbx servers. Now that vitelity will only work on 5060 with IP authentication, I need to figure out a way to minimize the hack attempts. The responsive firewall does a good job at blocking the hackers but on one server I had like 45 IPs that it was actively blocking. All these take up cpu and memory resources.

I just want to be able to minimize the exposure to my pbx system on the web utilizing the built in Firewall.

What do you guys suggest, I do?

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Custom Context - Include context in other context

December 10, 2018, 3:26 am

≫ Next: Need help with multiple devices on same extension

≪ Previous: Responsive Firewall - Driving me Crazy!

@essides wrote:

Hi there,

I’m trying to use custom contexts module , but I don’t know how to include contexts in other contexts that I already created.

Example:

I’ve created two contexts ( with their own dial patterns)
-mobile
-local-calls

And I would like to create a third context to include other contexts.

[billing-dep]
include => mobile
include => local-calls

The main idea is create departaments and define where they can call.

Thanks you.

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Need help with multiple devices on same extension

December 10, 2018, 6:03 am

≫ Next: Destination over "busy" onyl for external calls?

≪ Previous: Custom Context - Include context in other context

@Tommaso wrote:

Hello everyone!
There are already a lot of other tread with the same question, but i’m too inexpert to figure out the real way to achieve this: having a desk phone registered to one extension and add a softphone to the same extension!

I just need that they works one or the other! I mean that the softphone must allow a user to launch calls from outlook and websites (i already managed to use a “click-to-dial” addon, but it works only for my extension wich is registered only to a softphone), while the desk phone must be able to make and receive calls as it is already doing

I read about using PJSIP “contacts” feature, but i’m stuck on classic SIP so i cant use this function, and i read about “Device & User Mode” but the dislaimer on the wiki make me feel i’m going to only damage my system by playng with this mode…

any advices?

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Destination over "busy" onyl for external calls?

December 10, 2018, 6:16 am

≫ Next: Answering 2 calls at the same time

≪ Previous: Need help with multiple devices on same extension

@Tommaso wrote:

I’m aware of the possibility to set up a destination if an extension is busy… but this feature its half good and half bad for me!
I would like to make that internal calls will hear the classic “busy tone” if they call a busy extension, but i need that if someone call the office and skip the IVR by typing and extension number, if the extension they call is busy they will be redirected to the IVR or to a ring group!

I was just thinking about giving to those who already know our extension numbers, the numbers for the user Queues!
So, they call the office, the IVR plays, they call the queue so if the user is free they will be instantly make the phone ring, otherwise they will be put in a queue and the trick is done! but as i just said it would be only a trick (and the customers have to remember this new thing…)

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Answering 2 calls at the same time

December 10, 2018, 8:54 am

≫ Next: External Callers Setting Call Forwarding

≪ Previous: Destination over "busy" onyl for external calls?

@Matthew99 wrote:

Hi all

i recently introduced queues so i could glean some better reporting stats with the use of Asternic.

i have recently had a few reports of someone receiving a call and when they answered it they had actually answered 2 calls at the same time.

the queue structure we have is it calls support then fails over to a ring all queue, now this person is also in the ring all queue too.

it appears they went to answer the call initially but missed it the call then bounced through to everyone and when she went to answer answered another call coming into the queue along with the previous call so could hear 2 calls at once

we are using bria softphones, anyone ever come across this or have any ideas where i can start, i’ve checked the queue settings and nothing seems problematic here.

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External Callers Setting Call Forwarding

December 10, 2018, 10:28 am

≫ Next: Erorr with Freepbx gui (Non-Commercial)

≪ Previous: Answering 2 calls at the same time

@mlevy823 wrote:

We are having an issue with external callers setting up call forwarding for extensions that point towards external numbers. My guess is that these people are setting up call forwarding for our extensions so that they can then direct-dial those extensions and call whatever external number they forwarded through us.

I was wondering if there is any way to limit the call forwarding feature codes for only internal numbers, much like what was done with Asterisk In-Call Transferring getting an Advanced Option to prevent external callers from dialing *2 and ##.

I have tried looking everywhere for an option to limit the call forwarding feature codes but I have had no luck and I have simply disabled them, which is unfortunate because we would like to use them. Any help would be greatly appreciated, thanks!

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Erorr with Freepbx gui (Non-Commercial)

December 10, 2018, 7:32 pm

≫ Next: Removing Photo Thumbnail from YeaLink t46s

≪ Previous: External Callers Setting Call Forwarding

@irinetsys wrote:

Just want to say upfront thank you for anyone that is able to provided guidance with this issue that I am having with my freepbx system. I have having an issue where I can not at all apply configurations to the gui web interface.

I have removed all the commercial modules and when I go to apply the configuration I get this error shown below.

{code}
Error(s) have occured, the following is the retrieve_conf output:
exit: 1
Unable to continue. Undefined index: GENERATE_LEGACY_QUEUE_CODES in /var/www/html/admin/modules/queues/functions.inc/dialplan.php on line 313
#0 /var/www/html/admin/modules/queues/functions.inc/dialplan.php(313): Whoops\Run->handleError(8, ‘Undefined index…’, ‘/var/www/html/a…’, 313, Array)
#1 /var/www/html/admin/libraries/BMO/DialplanHooks.class.php(95): queues_get_config(‘asterisk’)
#2 /var/lib/asterisk/bin/retrieve_conf(860): FreePBX\DialplanHooks->processHooks(‘asterisk’, Array)
#3 {main}
{code}

Here are my installed modules

{code}
[root@pbx1 asterisk]# fwconsole ma list
No repos specified, using: [standard] from last GUI settings

±--------------------±-----------±--------±--------+
| Module | Version | Status | License |
±--------------------±-----------±--------±--------+
| accountcodepreserve | 13.0.2.2 | Enabled | GPLv2 |
| amd | 13.0.2 | Enabled | GPLv3+ |
| announcement | 13.0.7.3 | Enabled | GPLv3+ |
| arimanager | 13.0.4 | Enabled | GPLv3+ |
| asteriskinfo | 13.0.7.1 | Enabled | GPLv3+ |
| backup | 14.0.10.1 | Enabled | GPLv3+ |
| blacklist | 14.0.1 | Enabled | GPLv3+ |
| builtin | | Enabled | |
| bulkhandler | 13.0.14.8 | Enabled | GPLv3+ |
| calendar | 14.0.2.7 | Enabled | GPLv3+ |
| callback | 13.0.5.3 | Enabled | GPLv3+ |
| callforward | 14.0.1.3 | Enabled | AGPLv3+ |
| callrecording | 14.0.10 | Enabled | AGPLv3+ |
| callwaiting | 14.0.1.1 | Enabled | GPLv3+ |
| campon | 13.0.4.1 | Enabled | GPLv3+ |
| cdr | 14.0.5.14 | Enabled | GPLv3+ |
| cel | 14.0.2.8 | Enabled | GPLv3+ |
| certman | 14.0.3.1 | Enabled | AGPLv3+ |
| cidlookup | 14.0.1.7 | Enabled | GPLv3+ |
| conferences | 13.0.23.13 | Enabled | GPLv3+ |
| configedit | 13.0.7.1 | Enabled | AGPLv3+ |
| contactmanager | 14.0.4.10 | Enabled | GPLv3+ |
| core | 14.0.18.44 | Enabled | GPLv3+ |
| customappsreg | 13.0.5.5 | Enabled | GPLv3+ |
| dahdiconfig | 14.0.1.3 | Enabled | GPLv3+ |
| dashboard | 14.0.4.2 | Enabled | AGPLv3+ |
| daynight | 14.0.1 | Enabled | GPLv3+ |
| dictate | 13.0.5 | Enabled | GPLv3+ |
| directory | 13.0.19.7 | Enabled | GPLv3+ |
| disa | 13.0.6.10 | Enabled | AGPLv3+ |
| donotdisturb | 14.0.1.1 | Enabled | GPLv3+ |
| dundicheck | 2.11.0.3 | Enabled | GPLv3+ |
| extensionsettings | 13.0.4 | Enabled | GPLv3+ |
| fax | 14.0.2.6 | Enabled | GPLv3+ |
| findmefollow | 14.0.1.20 | Enabled | GPLv3+ |
| firewall | 13.0.57.1 | Enabled | AGPLv3+ |
| framework | 14.0.5.7 | Enabled | GPLv2+ |
| fw_langpacks | 14.0.1 | Enabled | GPLv3+ |
| hotelwakeup | 14.0.1.4 | Enabled | GPLv2 |
| iaxsettings | 14.0.1.4 | Enabled | AGPLv3 |
| infoservices | 13.0.1.3 | Enabled | GPLv2+ |
| irc | 13.0.1 | Enabled | GPLv3+ |
| ivr | 14.0.4 | Enabled | GPLv3+ |
| languages | 14.0.1.3 | Enabled | GPLv3+ |
| logfiles | 13.0.10.5 | Enabled | GPLv3+ |
| manager | 13.0.2.5 | Enabled | GPLv2+ |
| miscapps | 13.0.3.1 | Enabled | GPLv3+ |
| miscdests | 13.0.5 | Enabled | GPLv3+ |
| motif | 13.0.3.2 | Enabled | GPLv3+ |
| music | 13.0.22.5 | Enabled | GPLv3+ |
| outroutemsg | 13.0.2.1 | Enabled | GPLv3+ |
| paging | 14.0.6 | Enabled | GPLv3+ |
| parking | 13.0.19.9 | Enabled | GPLv3+ |
| pbdirectory | 2.11.0.6 | Enabled | GPLv3+ |
| phonebook | 13.0.6.1 | Enabled | GPLv3+ |
| phpinfo | 13.0.2 | Enabled | GPLv2+ |
| pinsets | 13.0.12 | Enabled | GPLv3+ |
| pm2 | 13.0.5.1 | Enabled | AGPLv3+ |
| presencestate | 14.0.1.7 | Enabled | GPLv3+ |
| printextensions | 13.0.3.1 | Enabled | GPLv3+ |
| queuemetrics | 2.11.0.3 | Enabled | GPLv3+ |
| queueprio | 13.0.3 | Enabled | GPLv3+ |
| queues | 14.0.2.22 | Enabled | GPLv2+ |
| recordings | 13.0.30.12 | Enabled | GPLv3+ |
| restapi | 13.0.21.2 | Enabled | AGPLv3 |
| ringgroups | 14.0.1.5 | Enabled | GPLv3+ |
| setcid | 13.0.6.2 | Enabled | GPLv3+ |
| sipsettings | 14.0.27.7 | Enabled | AGPLv3+ |
| soundlang | 14.0.5 | Enabled | GPLv3+ |
| speeddial | 2.11.0.4 | Enabled | GPLv3+ |
| superfecta | 14.0.7 | Enabled | GPLv2+ |
| timeconditions | 14.0.2.15 | Enabled | GPLv3+ |
| tts | 13.0.10 | Enabled | GPLv3+ |
| ttsengines | 13.0.7.3 | Enabled | AGPLv3 |
| userman | 14.0.3.46 | Enabled | AGPLv3+ |
| vmblast | 13.0.8 | Enabled | GPLv3+ |
| voicemail | 14.0.4.1 | Enabled | GPLv3+ |
| weakpasswords | 13.0.2 | Enabled | GPLv3+ |
±--------------------±-----------±--------±--------+
{code}

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Removing Photo Thumbnail from YeaLink t46s

December 4, 2018, 3:20 pm

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@SperoHealth wrote:

Hello,

I am trying to figure out how to remove the photo thumbnail on my YeaLink phone during call status (outgoing and incoming calls) which causes numbers to scroll and it just takes up a lot of room on the phone. Any insight would be helpful. Here is an image of what I mean

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