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Extension Error for extension that never existed

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@jhayes wrote:

We are continually getting errors for similar to the one below. The endpoint 7496 is a valid extension and works without issue. The extension portion of the error is not a number that we use in our inbound or outbound rules. I have check every table in the asterisk DB and cannot locate these numbers anywhere. The only thing I believe it may be (though have no idea where to look) is a file was uploaded of number for a broadcast calling campaign that played a recorded message to our clients. This campaign was removed over a year ago.


[2017-08-16 11:37:26] NOTICE[112985]: res_pjsip_exten_state.c:358 new_subscribe: Endpoint '7496' state subscription failed: Extension '2394813321' does not exist in context 'from-internal' or has no associated hint
[2017-08-16 11:37:26] NOTICE[112985]: res_pjsip_exten_state.c:358 new_subscribe: Endpoint '7496' state subscription failed: Extension '8667149301' does not exist in context 'from-internal' or has no associated hint
[2017-08-16 11:37:26] NOTICE[112985]: res_pjsip_exten_state.c:358 new_subscribe: Endpoint '7496' state subscription failed: Extension '8002662278' does not exist in context 'from-internal' or has no associated hint
[2017-08-16 11:37:26] NOTICE[108333]: res_pjsip_exten_state.c:358 new_subscribe: Endpoint '7496' state subscription failed: Extension '2392098869' does not exist in context 'from-internal' or has no associated hint
[2017-08-16 11:37:26] NOTICE[112985]: res_pjsip_exten_state.c:358 new_subscribe: Endpoint '7496' state subscription failed: Extension '8883818581' does not exist in context 'from-internal' or has no associated hint
[2017-08-16 11:37:26] NOTICE[108333]: res_pjsip_exten_state.c:358 new_subscribe: Endpoint '7496' state subscription failed: Extension '2392261800' does not exist in context 'from-internal' or has no associated hint
[2017-08-16 11:37:26] NOTICE[112985]: res_pjsip_exten_state.c:358 new_subscribe: Endpoint '7496' state subscription failed: Extension '8003093163' does not exist in context 'from-internal' or has no associated hint
[2017-08-16 11:37:26] NOTICE[108333]: res_pjsip_exten_state.c:358 new_subscribe: Endpoint '7496' state subscription failed: Extension '8003182596' does not exist in context 'from-internal' or has no associated hint
[2017-08-16 11:37:26] NOTICE[112985]: res_pjsip_exten_state.c:358 new_subscribe: Endpoint '7496' state subscription failed: Extension '2396562300' does not exist in context 'from-internal' or has no associated hint
[2017-08-16 11:37:26] NOTICE[108333]: res_pjsip_exten_state.c:358 new_subscribe: Endpoint '7496' state subscription failed: Extension '8007082848' does not exist in context 'from-internal' or has no associated hint

Note when we delete some extensions (imported from previous server) we get a similar error


[2017-08-16 11:34:01] NOTICE[108333]: res_pjsip_exten_state.c:358 new_subscribe: Endpoint '7895' state subscription failed: Extension '71050' does not exist in context 'from-internal' or has no associated hint

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Is CNAM Lookup/Passing for SIP to IAX2 Forwarded External Calls Possible?

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@jgiebler wrote:

We use IAX2 Trunks between all of our sites. Internal calls work great. The CNAM information for an extension passes without issue. (Figure 1)

We are now starting to forward external calls (from SIP trunks) to other PBXs in the network. Reasons may be: overflow, night answering etc. When the call is received by the other PBX, the CallerID Number appears but no CNAM information. The CNAM information is populated initially by OpenCNAM lookups on the initial server but does not pass along. (Figure 2)

Is there some way to either pass the CNAM information along from the initial server or do another lookup on the destination server?

Example: PBX A and PBX B

Figure 1 (Internal... Working)
1. Extension 100101 on PBX A calls Extension 101101 on PBX B.
2. Name and Number appear on extension 101101.

*This call routes over an IAX2 trunk to PBX B

Figure 2 (External to Internal then Forwarded... Not Working)
1. External Caller +1-555-212-0505 calls PBX A
2. CNAM Resolves to "Fake Test Number" on PBX A
3. Destination for inbound SIP route is set to ring Extension 101101 on PBX B. ( "After Hours" (Time Conditions))
4. External Call is ringing on 101101 on PBX B
5. Extension 101101 Phone shows +1-555-212-0505 as CallerID but CNAM is no longer on the record

*This call routes from external SIP trunk to internal IAX2 trunk

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Directory /daily not found when backup is run

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@Hawkeye wrote:

Hi,
After latest module admin upgrade in FreePBX 13 (~20 modules including backup/restore) backup is failing on remote ftp server:

Path: daily/20170814-221801-1502763481-13.0.192.16-1984672456.tgz
Could not find 20170814-221801-1502763481-13.0.192.16-1984672456.tgz on the remote system
Directory /daily not found

Fix seems to be to remove the 'daily' DIR.

mv daily daily.old

Run backup again:

Initializing Backup 3
Backup Lock acquired!
Running pre-backup hooks...
Adding items...
Building manifest...
Creating backup...
Storing backup...
Creating directory '/client-name/daily'
Saving file to remote ftp
Backup file uploaded to the remote server
Running maintenance..
Running post-backup hooks...
Backup successfully completed!

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Hard time hearing the caller on speaker phone (Sangoma S500)

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@gwntc wrote:

We had a Sangoma S500 telephone in a small conference room. We always had a hard time hearing the caller when we were on speaker phone. We purchased a yealink conference telephone. We still find that the call volume is low (even on max volume). The system is a FreePBX 60 with a 4 port FXO card (a200 I believe) with echo cancellation built in. I know that I can change the settings for the receive gain under dahdi config however I am not sure what I should change it to.

any help would be great

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Module upgrades failed

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@D4RR3N wrote:

Hi i received the usual email telling me there were module upgrades available, I have done these and this has now resulted in both the web interface and the terminal both giving me the error of "No direct script access allowed".
If I try to run
cd /var/lib/asterisk/bin
./module_admin --help

this also gives me the error, I'm stuck please help.

Darren

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911 dial pattern requirement [solved]

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@bob_dt wrote:

Having configured FreePBX on various systems over the past years, I still have little experience setting up E911.

Past systems had a "landline" connected that we "sent" our 911 calls out through. (Never had the need to dial 911 but, that is the way it was setup.)

My memory could be wrong so, I think I remember there being a dial pattern for "911" within the dial pattern listings for the outbound route. On my current system, until now, there has not been a 911 dial pattern. Now my provider offers a "911 test number" (922 in my case) to call, I had to add that number to my dial patterns to allow the test. Dialing this test number now reports the correct DID number associated for 911 emergency services location information.

So, this may seem like a really dumb question but, I now need to add "911" to my dial patterns for the associated "Emergency outbound route" or is this something that is part of some default dial pattern anywhere else within FreePBX? (All the legal requirements for E911 and the use of 911 for testing make me VERY uncomfortable, hence the question.)

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Echo on inter-extension calls

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@jhelou wrote:

Hello community, we have had many happy stories to share about implementations of Asterisk. There is one specific deployment though that has been a running nightmare. Echo and sound quality issues that refuse to go!
the latest is echo on internal calls (SIP to SIP) this is the first time that we see this! the environment is made up of FreePBX 13.0.192.9 running on a VM with cisco catalyst 3750 PoE switch and a mix of Cisco SPA5xx and Aastra 6731i phones.

We have a sangoma vega 50 TDM gateway to interface with PSTN. there is intermittent echo across the board, external call (often), internal calls from time to time. i've replaced the server, the gateway and the phones. Still no solution in sight, anyone has any ideas?
Thank you

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Any way to annex a pin number when setting findme/follow?

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@jhelou wrote:

Hello community, we have an installation that uses pin sets to control long distance access. each authorized user has a pin number they can use. When these users want to set their call forwarding or findme/follow the calls don't go through when they are to a long distance number. Any way the user can annex their pin to the destination?
Thank you

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"Whoops" errors for every minor PHP problem

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@miken32 wrote:

Just did the upgrade from 13 to 14, via the PBX Upgrader module. I'm now getting show-stopping errors for silly things like undefined variables. I'm assuming this is down to my PHP settings, or more people would be complaining about it. php.ini has error_reporting=E_ERROR (i.e. only report fatal errors.) If I disable the standard FreePBX error handler by adding $bootstrap_settings["freepbx_error_handler"] = false to /etc/freepbx.conf, everything works fine. Anything else I can try?

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Dialplan issues for prepending the extension in parking timeouts

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@hansentech wrote:

I wanted to have the extension prepended to the caller ID when a parked call times out and sends it back to the original extension that parked it or the alternate destination. I set the Auto CallerID Prepend selection accordingly but noticed that it always prepended UNKNOWN instead. I couldn't figure out why it wouldn't work so I dug into the dialplan within extensions_additional.conf to see what was going on. I added some Verbose commands so that I could debug it as shown below:

[sub-park-user]
exten => s,1,Set(UEXTEN=UNKNOWN)
exten => s,n,Set(UNAME=UNKNOWN)
exten => s,n,Set(DEVS=${DB_KEYS(DEVICE)})
exten => s,n,While($["${SET(DEV=${POP(DEVS)})}" != ""])
exten => s,n,Verbose(2, ${DB(DEVICE/${DEV}/dial)} compares to {CUT(CHANNEL(name),-,1)})
exten => s,n,GotoIf($["${DB(DEVICE/${DEV}/dial)}" = "${CUT(CHANNEL(name),-,1)}"]?found)

Once found, it updates UEXTEN with the matching extension. DEVS is a list of extensions (eg. - 301,302,303,304) and gets the associated entry for the extension's channel. However, CHANNEL(name) is always returning a channel for the inbound call's trunk meaning this will never match. I don't see how this feature would ever work yet I don't find anything in my searches that even mention people using the feature. Is there a bug in the default parking lot dialplan generation or am I missing something?

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Upgrade to 10.13.66-21

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@defres wrote:

I was upgrading from 10.13.66-20 to 10.13.66-21

After the upgraded finished I now get Unsigned Module(s) in yellow on the top of the dashboard.

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Odoo CRM Freepbx 13 integration

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@evanf wrote:

Hi Guys,

I've recently been tasked to integrate Odoo CRM with asterisk. I'm running Freepbx 13 at the moment and I was wondering if anyone has successfuly managed to integrate Odoo with Freepbx 13?

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Asterisk won't work as diaplan says

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@sam94 wrote:

I recently started a asterisk project and I wanted to play a sound file when I receives a call to my PSTN line. Actually using Obi110 I convert it to VoIP and transfer to raspberry pi which runs FreePBX and asterisk. However I registered Obi110 device on FreePBX and when I opened sip_addictional.conf , it shows Obi110 context is from-trunk. So I added;

[from-trunk]
exten => s,1,Answer()
exten => s,n,Playback(hello-world)
exten => s,n,Hangup

this to extension_custom.conf. However after that reloading the dialplan in asterisk, I made a call to PSTN line, but it doesn't work as this says. Just rings. Am I doing something wrong?

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Inbound call issues - The person at extension xxx is unavailable

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@cjj5167 wrote:

I have been assisting configuring a FreePBX setup. Previous to me, somebody in IT has used SIPStation to setup a trial to prove out creating extensions/outbound/inbound calls. Now that the trial has expired, I was tasked with getting one DID up and setting up the trunk. We have the DID through VoipInnovations. I was able to quickly get outgoing calls working, but I have been struggling with Incoming calls.

Long story short, after many configuration changes, I can now call my DID and I get the message "The person at extension xxx is unavailable." I just figured out how to to the Asterisk CLI, but the log is rather complicated to understand without knowing exactly what to look for.

In addition, I can't call from one extension to another.

At Asterisk CLI I typed 'core set verbose 5' and then I dialed ext 272. Here is the log from that....

Console verbose was OFF and is now 5.
== Setting global variable 'SIPDOMAIN' to 'MYSERVER'
-- Executing [272@from-internal:1] GotoIf("PJSIP/787-00000014", "1?ext-local,272,1:followme-check,272,1") in new stack
-- Goto (ext-local,272,1)
-- Executing [272@ext-local:1] Set("PJSIP/787-00000014", "__RINGTIMER=15") in new stack
-- Executing [272@ext-local:2] Macro("PJSIP/787-00000014", "exten-vm,novm,272,0,0,1") in new stack
-- Executing [s@macro-exten-vm:1] Macro("PJSIP/787-00000014", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("PJSIP/787-00000014", "TOUCH_MONITOR=1503086071.21") in new stack
-- Executing [s@macro-user-callerid:2] Set("PJSIP/787-00000014", "AMPUSER=787") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("PJSIP/787-00000014", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("PJSIP/787-00000014", "1?Set(__REALCALLERIDNUM=787)") in new stack
-- Executing [s@macro-user-callerid:5] Set("PJSIP/787-00000014", "AMPUSER=787") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("PJSIP/787-00000014", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("PJSIP/787-00000014", "AMPUSERCIDNAME=Extension 787") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("PJSIP/787-00000014", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("PJSIP/787-00000014", "AMPUSERCID=787") in new stack
-- Executing [s@macro-user-callerid:10] Set("PJSIP/787-00000014", "_DIALOPTIONS=Ttr") in new stack
-- Executing [s@macro-user-callerid:11] Set("PJSIP/787-00000014", "CALLERID(all)="Extension 787" <787>") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("PJSIP/787-00000014", "0?limit") in new stack
-- Executing [s@macro-user-callerid:13] ExecIf("PJSIP/787-00000014", "0?Set(GROUP(concurrency_limit)=787)") in new stack
-- Executing [s@macro-user-callerid:14] ExecIf("PJSIP/787-00000014", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:15] GotoIf("PJSIP/787-00000014", "0?continue") in new stack
-- Executing [s@macro-user-callerid:16] ExecIf("PJSIP/787-00000014", "1?Set(_CALLEEACCOUNCODE=)") in new stack
-- Executing [s@macro-user-callerid:17] Set("PJSIP/787-00000014", "__TTL=6") in new stack
-- Executing [s@macro-user-callerid:18] GotoIf("PJSIP/787-00000014", "1?continue") in new stack
-- Goto (macro-user-callerid,s,29)
-- Executing [s@macro-user-callerid:29] Set("PJSIP/787-00000014", "CALLERID(number)=787") in new stack
-- Executing [s@macro-user-callerid:30] Set("PJSIP/787-00000014", "CALLERID(name)=Extension 787") in new stack
-- Executing [s@macro-user-callerid:31] GotoIf("PJSIP/787-00000014", "0?cnum") in new stack
-- Executing [s@macro-user-callerid:32] Set("PJSIP/787-00000014", "CDR(cnam)=Extension 787") in new stack
-- Executing [s@macro-user-callerid:33] Set("PJSIP/787-00000014", "CDR(cnum)=787") in new stack
-- Executing [s@macro-user-callerid:34] Set("PJSIP/787-00000014", "CHANNEL(language)=en") in new stack
-- Executing [s@macro-exten-vm:2] Set("PJSIP/787-00000014", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("PJSIP/787-00000014", "__EXTTOCALL=272") in new stack
-- Executing [s@macro-exten-vm:4] Set("PJSIP/787-00000014", "__PICKUPMARK=272") in new stack
-- Executing [s@macro-exten-vm:5] Set("PJSIP/787-00000014", "RT=15") in new stack
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: chan_sip.c:22955 func_header_read: This function can only be used on SIP channels.
-- Executing [s@macro-exten-vm:6] ExecIf("PJSIP/787-00000014", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: chan_sip.c:22955 func_header_read: This function can only be used on SIP channels.
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: chan_sip.c:22955 func_header_read: This function can only be used on SIP channels.
-- Executing [s@macro-exten-vm:7] ExecIf("PJSIP/787-00000014", "0?MacroExit()") in new stack
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: chan_sip.c:22955 func_header_read: This function can only be used on SIP channels.
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: chan_sip.c:22955 func_header_read: This function can only be used on SIP channels.
-- Executing [s@macro-exten-vm:8] ExecIf("PJSIP/787-00000014", "0?Gosub(ext-intercom,*80272,1())") in new stack
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: chan_sip.c:22955 func_header_read: This function can only be used on SIP channels.
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: chan_sip.c:22955 func_header_read: This function can only be used on SIP channels.
-- Executing [s@macro-exten-vm:9] ExecIf("PJSIP/787-00000014", "0?MacroExit()") in new stack
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: chan_sip.c:22955 func_header_read: This function can only be used on SIP channels.
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: chan_sip.c:22955 func_header_read: This function can only be used on SIP channels.
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: pbx_functions.c:460 func_args: Can't find trailing parenthesis for function 'DB(DEVICE/272/dial'?
-- Executing [s@macro-exten-vm:10] ExecIf("PJSIP/787-00000014", "0?ChanSpy(PJSIP/272,q)") in new stack
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: chan_sip.c:22955 func_header_read: This function can only be used on SIP channels.
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: pbx_functions.c:460 func_args: Can't find trailing parenthesis for function 'DB(DEVICE/272/dial'?
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: chan_sip.c:22955 func_header_read: This function can only be used on SIP channels.
-- Executing [s@macro-exten-vm:11] ExecIf("PJSIP/787-00000014", "0?MacroExit()") in new stack
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: chan_sip.c:22955 func_header_read: This function can only be used on SIP channels.
-- Executing [s@macro-exten-vm:12] Gosub("PJSIP/787-00000014", "sub-record-check,s,1(exten,272,dontcare)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("PJSIP/787-00000014", "0?initialized") in new stack
-- Executing [s@sub-record-check:2] Set("PJSIP/787-00000014", "_RECSTATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:3] Set("PJSIP/787-00000014", "NOW=1503086071") in new stack
-- Executing [s@sub-record-check:4] Set("PJSIP/787-00000014", "__DAY=18") in new stack
-- Executing [s@sub-record-check:5] Set("PJSIP/787-00000014", "__MONTH=08") in new stack
-- Executing [s@sub-record-check:6] Set("PJSIP/787-00000014", "__YEAR=2017") in new stack
-- Executing [s@sub-record-check:7] Set("PJSIP/787-00000014", "__TIMESTR=20170818-155431") in new stack
-- Executing [s@sub-record-check:8] Set("PJSIP/787-00000014", "__FROMEXTEN=787") in new stack
-- Executing [s@sub-record-check:9] Set("PJSIP/787-00000014", "_MONFMT=wav") in new stack
-- Executing [s@sub-record-check:10] NoOp("PJSIP/787-00000014", "Recordings initialized") in new stack
-- Executing [s@sub-record-check:11] ExecIf("PJSIP/787-00000014", "0?Set(ARG3=dontcare)") in new stack
-- Executing [s@sub-record-check:12] Set("PJSIP/787-00000014", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:13] ExecIf("PJSIP/787-00000014", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [s@sub-record-check:14] GotoIf("PJSIP/787-00000014", "5?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [s@sub-record-check:17] GotoIf("PJSIP/787-00000014", "1?sub-record-check,exten,1") in new stack
-- Goto (sub-record-check,exten,1)
-- Executing [exten@sub-record-check:1] NoOp("PJSIP/787-00000014", "Exten Recording Check between 787 and 272") in new stack
-- Executing [exten@sub-record-check:2] Set("PJSIP/787-00000014", "CALLTYPE=internal") in new stack
-- Executing [exten@sub-record-check:3] ExecIf("PJSIP/787-00000014", "0?Set(CALLTYPE=)") in new stack
-- Executing [exten@sub-record-check:4] Set("PJSIP/787-00000014", "CALLEE=dontcare") in new stack
-- Executing [exten@sub-record-check:5] ExecIf("PJSIP/787-00000014", "0?Set(CALLEE=dontcare)") in new stack
-- Executing [exten@sub-record-check:6] GotoIf("PJSIP/787-00000014", "0?callee") in new stack
-- Executing [exten@sub-record-check:7] GotoIf("PJSIP/787-00000014", "1?caller") in new stack
-- Goto (sub-record-check,exten,13)
-- Executing [exten@sub-record-check:13] Set("PJSIP/787-00000014", "RECMODE=dontcare") in new stack
-- Executing [exten@sub-record-check:14] ExecIf("PJSIP/787-00000014", "0?Set(RECMODE=dontcare)") in new stack
-- Executing [exten@sub-record-check:15] ExecIf("PJSIP/787-00000014", "1?Set(RECMODE=dontcare)") in new stack
-- Executing [exten@sub-record-check:16] Gosub("PJSIP/787-00000014", "recordcheck,1(dontcare,internal,272)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/787-00000014", "Starting recording check against dontcare") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("PJSIP/787-00000014", "dontcare") in new stack
-- Goto (sub-record-check,recordcheck,3)
-- Executing [recordcheck@sub-record-check:3] Return("PJSIP/787-00000014", "") in new stack
-- Executing [exten@sub-record-check:17] Return("PJSIP/787-00000014", "") in new stack
-- Executing [s@macro-exten-vm:13] GotoIf("PJSIP/787-00000014", "1?macrodial") in new stack
-- Goto (macro-exten-vm,s,19)
-- Executing [s@macro-exten-vm:19] GosubIf("PJSIP/787-00000014", "0?clrheader,1()") in new stack
-- Executing [s@macro-exten-vm:20] Macro("PJSIP/787-00000014", "dial-one,15,Ttr,272") in new stack
-- Executing [s@macro-dial-one:1] Set("PJSIP/787-00000014", "DEXTEN=272") in new stack
-- Executing [s@macro-dial-one:2] Set("PJSIP/787-00000014", "_CRMSOURCE=787") in new stack
-- Executing [s@macro-dial-one:3] ExecIf("PJSIP/787-00000014", "0?Set(EXTTOCALL=272)") in new stack
-- Executing [s@macro-dial-one:4] Set("PJSIP/787-00000014", "DIALSTATUS_CW=") in new stack
-- Executing [s@macro-dial-one:5] GosubIf("PJSIP/787-00000014", "0?screen,1()") in new stack
-- Executing [s@macro-dial-one:6] GosubIf("PJSIP/787-00000014", "0?cf,1()") in new stack
-- Executing [s@macro-dial-one:7] GotoIf("PJSIP/787-00000014", "1?skip1") in new stack
-- Goto (macro-dial-one,s,10)
-- Executing [s@macro-dial-one:10] GotoIf("PJSIP/787-00000014", "0?nodial") in new stack
-- Executing [s@macro-dial-one:11] GotoIf("PJSIP/787-00000014", "0?continue") in new stack
-- Executing [s@macro-dial-one:12] Set("PJSIP/787-00000014", "EXTHASCW=ENABLED") in new stack
-- Executing [s@macro-dial-one:13] GotoIf("PJSIP/787-00000014", "0?next1:cwinusebusy") in new stack
-- Goto (macro-dial-one,s,25)
-- Executing [s@macro-dial-one:25] GotoIf("PJSIP/787-00000014", "0?next3:continue") in new stack
-- Goto (macro-dial-one,s,27)
-- Executing [s@macro-dial-one:27] GotoIf("PJSIP/787-00000014", "0?nodial") in new stack
-- Executing [s@macro-dial-one:28] GosubIf("PJSIP/787-00000014", "1?dstring,1():dlocal,1()") in new stack
-- Executing [dstring@macro-dial-one:1] Set("PJSIP/787-00000014", "DSTRING=") in new stack
-- Executing [dstring@macro-dial-one:2] Set("PJSIP/787-00000014", "DEVICES=99272&272") in new stack
-- Executing [dstring@macro-dial-one:3] ExecIf("PJSIP/787-00000014", "0?Return()") in new stack
-- Executing [dstring@macro-dial-one:4] ExecIf("PJSIP/787-00000014", "0?Set(DEVICES=9272&272)") in new stack
-- Executing [dstring@macro-dial-one:5] Set("PJSIP/787-00000014", "LOOPCNT=2") in new stack
-- Executing [dstring@macro-dial-one:6] Set("PJSIP/787-00000014", "ITER=1") in new stack
-- Executing [dstring@macro-dial-one:7] Set("PJSIP/787-00000014", "THISDIAL=SIP/99272") in new stack
-- Executing [dstring@macro-dial-one:8] GosubIf("PJSIP/787-00000014", "1?zap2dahdi,1()") in new stack
-- Executing [zap2dahdi@macro-dial-one:1] ExecIf("PJSIP/787-00000014", "0?Return()") in new stack
-- Executing [zap2dahdi@macro-dial-one:2] Set("PJSIP/787-00000014", "NEWDIAL=") in new stack
-- Executing [zap2dahdi@macro-dial-one:3] Set("PJSIP/787-00000014", "LOOPCNT2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:4] Set("PJSIP/787-00000014", "ITER2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:5] Set("PJSIP/787-00000014", "THISPART2=SIP/99272") in new stack
-- Executing [zap2dahdi@macro-dial-one:6] ExecIf("PJSIP/787-00000014", "0?Set(THISPART2=DAHDI/99272)") in new stack
-- Executing [zap2dahdi@macro-dial-one:7] Set("PJSIP/787-00000014", "NEWDIAL=SIP/99272&") in new stack
-- Executing [zap2dahdi@macro-dial-one:8] Set("PJSIP/787-00000014", "ITER2=2") in new stack
-- Executing [zap2dahdi@macro-dial-one:9] GotoIf("PJSIP/787-00000014", "0?begin2") in new stack
-- Executing [zap2dahdi@macro-dial-one:10] Set("PJSIP/787-00000014", "THISDIAL=SIP/99272") in new stack
-- Executing [zap2dahdi@macro-dial-one:11] Return("PJSIP/787-00000014", "") in new stack
-- Executing [dstring@macro-dial-one:9] GotoIf("PJSIP/787-00000014", "1?docheck") in new stack
-- Goto (macro-dial-one,dstring,14)
-- Executing [dstring@macro-dial-one:14] GotoIf("PJSIP/787-00000014", "0?skipset") in new stack
-- Executing [dstring@macro-dial-one:15] Set("PJSIP/787-00000014", "DSTRING=SIP/99272&") in new stack
-- Executing [dstring@macro-dial-one:16] Set("PJSIP/787-00000014", "ITER=2") in new stack
-- Executing [dstring@macro-dial-one:17] GotoIf("PJSIP/787-00000014", "1?begin") in new stack
-- Goto (macro-dial-one,dstring,7)
-- Executing [dstring@macro-dial-one:7] Set("PJSIP/787-00000014", "THISDIAL=PJSIP/272") in new stack
-- Executing [dstring@macro-dial-one:8] GosubIf("PJSIP/787-00000014", "1?zap2dahdi,1()") in new stack
-- Executing [zap2dahdi@macro-dial-one:1] ExecIf("PJSIP/787-00000014", "0?Return()") in new stack
-- Executing [zap2dahdi@macro-dial-one:2] Set("PJSIP/787-00000014", "NEWDIAL=") in new stack
-- Executing [zap2dahdi@macro-dial-one:3] Set("PJSIP/787-00000014", "LOOPCNT2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:4] Set("PJSIP/787-00000014", "ITER2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:5] Set("PJSIP/787-00000014", "THISPART2=PJSIP/272") in new stack
-- Executing [zap2dahdi@macro-dial-one:6] ExecIf("PJSIP/787-00000014", "0?Set(THISPART2=DAHDIIP/272)") in new stack
-- Executing [zap2dahdi@macro-dial-one:7] Set("PJSIP/787-00000014", "NEWDIAL=PJSIP/272&") in new stack
-- Executing [zap2dahdi@macro-dial-one:8] Set("PJSIP/787-00000014", "ITER2=2") in new stack
-- Executing [zap2dahdi@macro-dial-one:9] GotoIf("PJSIP/787-00000014", "0?begin2") in new stack
-- Executing [zap2dahdi@macro-dial-one:10] Set("PJSIP/787-00000014", "THISDIAL=PJSIP/272") in new stack
-- Executing [zap2dahdi@macro-dial-one:11] Return("PJSIP/787-00000014", "") in new stack
-- Executing [dstring@macro-dial-one:9] GotoIf("PJSIP/787-00000014", "0?docheck") in new stack
-- Executing [dstring@macro-dial-one:10] NoOp("PJSIP/787-00000014", "Debug: Found PJSIP Destination PJSIP/272") in new stack
-- Executing [dstring@macro-dial-one:11] GotoIf("PJSIP/787-00000014", "0?doset") in new stack
-- Executing [dstring@macro-dial-one:12] NoOp("PJSIP/787-00000014", "Debug: Updating PJSIP Destination with PJSIP_DIAL_CONTACTS") in new stack
-- Executing [dstring@macro-dial-one:13] Set("PJSIP/787-00000014", "THISDIAL=") in new stack
-- Executing [dstring@macro-dial-one:14] GotoIf("PJSIP/787-00000014", "1?skipset") in new stack
-- Goto (macro-dial-one,dstring,16)
-- Executing [dstring@macro-dial-one:16] Set("PJSIP/787-00000014", "ITER=3") in new stack
-- Executing [dstring@macro-dial-one:17] GotoIf("PJSIP/787-00000014", "0?begin") in new stack
-- Executing [dstring@macro-dial-one:18] ExecIf("PJSIP/787-00000014", "0?Return()") in new stack
-- Executing [dstring@macro-dial-one:19] Set("PJSIP/787-00000014", "DSTRING=SIP/99272") in new stack
-- Executing [dstring@macro-dial-one:20] Return("PJSIP/787-00000014", "") in new stack
-- Executing [s@macro-dial-one:29] GotoIf("PJSIP/787-00000014", "0?nodial") in new stack
-- Executing [s@macro-dial-one:30] GotoIf("PJSIP/787-00000014", "0?skiptrace") in new stack
-- Executing [s@macro-dial-one:31] GosubIf("PJSIP/787-00000014", "1?ctset,1():ctclear,1()") in new stack
-- Executing [ctset@macro-dial-one:1] Set("PJSIP/787-00000014", "DB(CALLTRACE/272)=787") in new stack
-- Executing [ctset@macro-dial-one:2] Return("PJSIP/787-00000014", "") in new stack
-- Executing [s@macro-dial-one:32] Set("PJSIP/787-00000014", "D_OPTIONS=Ttr") in new stack
-- Executing [s@macro-dial-one:33] NoOp("PJSIP/787-00000014", "Blind Transfer: , Attended Transfer: , User: 787, Alert Info: ") in new stack
-- Executing [s@macro-dial-one:34] ExecIf("PJSIP/787-00000014", "1?Set(ALERT_INFO=)") in new stack
-- Executing [s@macro-dial-one:35] ExecIf("PJSIP/787-00000014", "0?Set(ALERT_INFO=)") in new stack
-- Executing [s@macro-dial-one:36] ExecIf("PJSIP/787-00000014", "0?Set(ALERT_INFO=)") in new stack
-- Executing [s@macro-dial-one:37] ExecIf("PJSIP/787-00000014", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
-- Executing [s@macro-dial-one:38] ExecIf("PJSIP/787-00000014", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
-- Executing [s@macro-dial-one:39] GosubIf("PJSIP/787-00000014", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
-- Executing [s@macro-dial-one:40] ExecIf("PJSIP/787-00000014", "0?Set(CHANNEL(musicclass)=)") in new stack
-- Executing [s@macro-dial-one:41] GosubIf("PJSIP/787-00000014", "0?qwait,1()") in new stack
-- Executing [s@macro-dial-one:42] Set("PJSIP/787-00000014", "__CWIGNORE=") in new stack
-- Executing [s@macro-dial-one:43] Set("PJSIP/787-00000014", "__KEEPCID=TRUE") in new stack
-- Executing [s@macro-dial-one:44] GotoIf("PJSIP/787-00000014", "0?usegoto,1") in new stack
-- Executing [s@macro-dial-one:45] GotoIf("PJSIP/787-00000014", "0?godial") in new stack
-- Executing [s@macro-dial-one:46] Gosub("PJSIP/787-00000014", "sub-presencestate-display,s,1(272)") in new stack
-- Executing [s@sub-presencestate-display:1] Goto("PJSIP/787-00000014", "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/787-00000014", "PRESENCESTATE_DISPLAY=") in new stack
-- Executing [state-not_set@sub-presencestate-display:2] Return("PJSIP/787-00000014", "") in new stack
-- Executing [s@macro-dial-one:47] Set("PJSIP/787-00000014", "CONNECTEDLINE(name,i)=fname lname") in new stack
-- Executing [s@macro-dial-one:48] Set("PJSIP/787-00000014", "CONNECTEDLINE(num)=272") in new stack
-- Executing [s@macro-dial-one:49] Set("PJSIP/787-00000014", "D_OPTIONS=TtrI") in new stack
-- Executing [s@macro-dial-one:50] Macro("PJSIP/787-00000014", "dialout-one-predial-hook,") in new stack
-- Executing [s@macro-dialout-one-predial-hook:1] MacroExit("PJSIP/787-00000014", "") in new stack
-- Executing [s@macro-dial-one:51] ExecIf("PJSIP/787-00000014", "0?Set(D_OPTIONS=trII)") in new stack
-- Executing [s@macro-dial-one:52] NoOp("PJSIP/787-00000014", "") in new stack
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: taskprocessor.c:888 taskprocessor_push: The 'subm:ast_channel_topic_all-cached-00000064' task processor queue reached 500 scheduled tasks again.
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: taskprocessor.c:888 taskprocessor_push: The 'subm:ast_channel_topic_all-cached-00000065' task processor queue reached 500 scheduled tasks again.
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: taskprocessor.c:888 taskprocessor_push: The 'subm:ast_channel_topic_all-cached-00000067' task processor queue reached 500 scheduled tasks again.
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: taskprocessor.c:888 taskprocessor_push: The 'subm:endpoint_topic_all-cached-00000008' task processor queue reached 500 scheduled tasks again.
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: taskprocessor.c:888 taskprocessor_push: The 'subp:PJSIP/787-0000001e' task processor queue reached 500 scheduled tasks.
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: taskprocessor.c:888 taskprocessor_push: The 'subm:endpoint_topic_all-cached-0000006b' task processor queue reached 500 scheduled tasks.
-- Executing [s@macro-dial-one:53] Dial("PJSIP/787-00000014", "SIP/99272,15,TtrIb(func-apply-sipheaders^s^1)") in new stack
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: app_dial.c:2525 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dial-one:54] ExecIf("PJSIP/787-00000014", "0?MacroExit()") in new stack
[2017-08-18 15:54:31] WARNING[27938]: db.c:288 db_execute_sql: Error executing SQL (COMMIT): database is locked
-- Executing [s@macro-dial-one:55] ExecIf("PJSIP/787-00000014", "0?Set(DIALSTATUS=)") in new stack
-- Executing [s@macro-dial-one:56] GosubIf("PJSIP/787-00000014", "0?s-CHANUNAVAIL,1()") in new stack
-- Executing [s@macro-dial-one:57] MacroExit("PJSIP/787-00000014", "") in new stack
-- Executing [s@macro-exten-vm:21] Set("PJSIP/787-00000014", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:22] GosubIf("PJSIP/787-00000014", "0?docfu,1()") in new stack
-- Executing [s@macro-exten-vm:23] GosubIf("PJSIP/787-00000014", "0?docfb,1()") in new stack
-- Executing [s@macro-exten-vm:24] Set("PJSIP/787-00000014", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:25] ExecIf("PJSIP/787-00000014", "1?MacroExit()") in new stack
-- Executing [272@ext-local:3] Set("PJSIP/787-00000014", "__PICKUPMARK=") in new stack
-- Executing [272@ext-local:4] GotoIf("PJSIP/787-00000014", "1?ext-local,vmu889,1") in new stack
-- Goto (ext-local,vmu889,1)
-- Executing [vmu889@ext-local:1] Macro("PJSIP/787-00000014", "vm,889,NOANSWER,") in new stack
-- Executing [s@macro-vm:1] Macro("PJSIP/787-00000014", "user-callerid,SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:1] Set("PJSIP/787-00000014", "TOUCH_MONITOR=1503086071.21") in new stack
-- Executing [s@macro-user-callerid:2] Set("PJSIP/787-00000014", "AMPUSER=787") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("PJSIP/787-00000014", "13?report") in new stack
-- Goto (macro-user-callerid,s,15)
-- Executing [s@macro-user-callerid:15] GotoIf("PJSIP/787-00000014", "1?continue") in new stack
-- Goto (macro-user-callerid,s,29)
-- Executing [s@macro-user-callerid:29] Set("PJSIP/787-00000014", "CALLERID(number)=787") in new stack
-- Executing [s@macro-user-callerid:30] Set("PJSIP/787-00000014", "CALLERID(name)=Extension 787") in new stack
-- Executing [s@macro-user-callerid:31] GotoIf("PJSIP/787-00000014", "0?cnum") in new stack
-- Executing [s@macro-user-callerid:32] Set("PJSIP/787-00000014", "CDR(cnam)=Extension 787") in new stack
-- Executing [s@macro-user-callerid:33] Set("PJSIP/787-00000014", "CDR(cnum)=787") in new stack
-- Executing [s@macro-user-callerid:34] Set("PJSIP/787-00000014", "CHANNEL(language)=en") in new stack
-- Executing [s@macro-vm:2] Set("PJSIP/787-00000014", "VMGAIN=") in new stack
-- Executing [s@macro-vm:3] Macro("PJSIP/787-00000014", "blkvm-check,") in new stack
-- Executing [s@macro-blkvm-check:1] Set("PJSIP/787-00000014", "GOSUB_RETVAL=") in new stack
-- Executing [s@macro-blkvm-check:2] ExecIf("PJSIP/787-00000014", "0?Set(GOSUB_RETVAL=TRUE)") in new stack
-- Executing [s@macro-blkvm-check:3] MacroExit("PJSIP/787-00000014", "") in new stack
-- Executing [s@macro-vm:4] GotoIf("PJSIP/787-00000014", "1?vmx,1") in new stack
-- Goto (macro-vm,vmx,1)
-- Executing [vmx@macro-vm:1] Set("PJSIP/787-00000014", "__EXTTOCALL=889") in new stack
-- Executing [vmx@macro-vm:2] Set("PJSIP/787-00000014", "_CRMVOICEMAIL=889") in new stack
-- Executing [vmx@macro-vm:3] Set("PJSIP/787-00000014", "MEXTEN=889") in new stack
-- Executing [vmx@macro-vm:4] Set("PJSIP/787-00000014", "MMODE=NOANSWER") in new stack
-- Executing [vmx@macro-vm:5] Set("PJSIP/787-00000014", "RETVM=") in new stack
-- Executing [vmx@macro-vm:6] Set("PJSIP/787-00000014", "MODE=unavail") in new stack
-- Executing [vmx@macro-vm:7] Macro("PJSIP/787-00000014", "get-vmcontext,889") in new stack
-- Executing [s@macro-get-vmcontext:1] Set("PJSIP/787-00000014", "VMCONTEXT=default") in new stack
-- Executing [s@macro-get-vmcontext:2] GotoIf("PJSIP/787-00000014", "0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing [s@macro-get-vmcontext:300] NoOp("PJSIP/787-00000014", "") in new stack
-- Executing [vmx@macro-vm:8] Set("PJSIP/787-00000014", "MODE=unavail") in new stack
-- Executing [vmx@macro-vm:9] NoOp("PJSIP/787-00000014", "MODE IS: unavail") in new stack
-- Executing [vmx@macro-vm:10] GotoIf("PJSIP/787-00000014", "1?chknomsg") in new stack
-- Goto (macro-vm,vmx,12)
-- Executing [vmx@macro-vm:12] GotoIf("PJSIP/787-00000014", "0?s-NOANSWER,1") in new stack
-- Executing [vmx@macro-vm:13] GotoIf("PJSIP/787-00000014", "1?notdirect") in new stack
-- Goto (macro-vm,vmx,15)
-- Executing [vmx@macro-vm:15] NoOp("PJSIP/787-00000014", "Checking if ext 889 is enabled: ") in new stack
-- Executing [vmx@macro-vm:16] GotoIf("PJSIP/787-00000014", "1?s-NOANSWER,1") in new stack
-- Goto (macro-vm,s-NOANSWER,1)
-- Executing [s-NOANSWER@macro-vm:1] Macro("PJSIP/787-00000014", "get-vmcontext,889") in new stack
-- Executing [s@macro-get-vmcontext:1] Set("PJSIP/787-00000014", "VMCONTEXT=default") in new stack
-- Executing [s@macro-get-vmcontext:2] GotoIf("PJSIP/787-00000014", "0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing [s@macro-get-vmcontext:300] NoOp("PJSIP/787-00000014", "") in new stack
-- Executing [s-NOANSWER@macro-vm:2] VoiceMail("PJSIP/787-00000014", "889@default,u") in new stack

0x7f50c0e10390 -- Probation passed - setting RTP source address to 146.186.32.137:16388
[2017-08-18 15:54:31] WARNING[30366][C-00000014]: translate.c:487 ast_translator_build_path: No translator path: (ending codec is not valid)
-- Playing 'vm-theperson.ulaw' (language 'en')
[2017-08-18 15:54:33] WARNING[30366][C-00000014]: translate.c:487 ast_translator_build_path: No translator path: (ending codec is not valid)
-- Playing 'digits/8.ulaw' (language 'en')
[2017-08-18 15:54:33] WARNING[30366][C-00000014]: translate.c:487 ast_translator_build_path: No translator path: (ending codec is not valid)
[2017-08-18 15:54:33] WARNING[30366][C-00000014]: translate.c:487 ast_translator_build_path: No translator path: (ending codec is not valid)
-- Playing 'digits/8.ulaw' (language 'en')
[2017-08-18 15:54:33] WARNING[30366][C-00000014]: translate.c:487 ast_translator_build_path: No translator path: (ending codec is not valid)

-- <PJSIP/787-00000014> Playing 'digits/9.ulaw' (language 'en')

[2017-08-18 15:54:34] WARNING[30366][C-00000014]: translate.c:487 ast_translator_build_path: No translator path: (ending codec is not valid)

-- <PJSIP/787-00000014> Playing 'vm-isunavail.ulaw' (language 'en')

[2017-08-18 15:54:35] WARNING[30366][C-00000014]: translate.c:487 ast_translator_build_path: No translator path: (ending codec is not valid)
== Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on 'PJSIP/787-00000014' in macro 'vm'
== Spawn extension (ext-local, vmu889, 1) exited non-zero on 'PJSIP/787-00000014'
-- Executing [h@ext-local:1] Macro("PJSIP/787-00000014", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/787-00000014", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("PJSIP/787-00000014", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("PJSIP/787-00000014", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'PJSIP/787-00000014' in macro 'hangupcall'
== Spawn extension (ext-local, h, 1) exited non-zero on 'PJSIP/787-00000014'
freepbx*CLI>

Any ideas? Any thing to check for inbound calls from outside?

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Wich extension has ansewered the call?


Radio recording

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@psdk wrote:

Hi,

I'm going to connect FreePBX to a radio system and just record all radio stream voices.
Is there any solution?
I read about app_rpt. Is it useful?

Thanks for helps.

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Participants: 1

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Apply Config gives Unable to connect to Asterisk through the CLI

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@falloutphil wrote:

Hi,

Exception: Unable to connect to Asterisk through the CLI

Recently upgraded from 12 from 13 which was a bit bumpy - currently it reports:
FreePBX 13.0.192.16 'VoIP Server'

Asterix is working fine (calls in and out work), and FreePBX front end is working apart from applying changes.

I've added a bit more detail to the php exception string to try to get some more detail.

If I run the command from the console as root it works fine:

/usr/sbin/asterisk -rx 'core show version'

Asterisk 11.12.0 built by root @ foobar on a i686 running Linux on 2014-09-14 19:43:36 UTC

echo $?

0

$loc = fpbx_which("asterisk");
if(empty($loc)) {
throw new \Exception(_("Unable to find the Asterisk binary"));
} else {
exec($loc . " -rx 'core show version'",$out,$ret);
if($ret != 0) {
$foo=implode("|",$out);
throw new \Exception(_("Unable to connect to Asterisk through the CLI $loc $foo $ret"));
}
}

exit: 1
Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
Exception: Unable to connect to Asterisk through the CLI /usr/sbin/asterisk 1 in file /var/lib/asterisk/bin/retrieve_conf on line 42
Stack trace:
1. Exception->() /var/lib/asterisk/bin/retrieve_conf:42

I'm running Ubuntu 16.04 with php5.6 backport.

All modules up to date apart from Digium Addons, iSymphonyV3 and Parking Lot.

Any ideas?

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Participants: 2

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Core module upgrade breaks trunks

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@MiMar wrote:

Hi. After I update FreePBX from Core version 13.0.113 to version 13.0.120.10 my trunks no longer register. I receive the following msg in the log files.

res_pjsip.c: Unable to apply outbound proxy on request OPTIONS to endpoint MyTrunkName as outbound proxy URI 'SipServer.net' is not valid

res_pjsip/pjsip_options.c: Unable to create request to qualify contact sip:myaccount@sipserver.net:5060

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Cant install asterisk on fedora 26

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@Trezeguet17 wrote:

Hi people, when i try to install Dahdi on fedora 26 , after i made the structure of the directories, and i go to dahdi uncompressede folder and type make i get the following

/dahdi/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/dahdi-base.c: In function ‘dahdi_ioctl_iomux’:
/dahdi/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/dahdi-base.c:5954:7: error: implicit declaration of function ‘signal_pending’; did you mean ‘timer_pending’? [-Werror=implicit-function-declaration]
if (signal_pending(current)) {
^~~~~~~~~~~~~~
timer_pending

please help!!

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