Zulu plugin dial prefix

October 1, 2018, 8:08 am

≫ Next: Sangoma phone only sends first DTMF tone

≪ Previous: VEGA400 Call Leg/Transaction Does Not Exist

Hi, I am new to this forum and also quite new to freepbx. We run a freepbx installationwith the two for free zulu client licenses. One issue I have is that in our company we have to dial prefix 0 if we want to dial out, like if I need to dial 0895424330 I need to dial 0 0895423330. Due to this, none of my Zulu browser plugin dials works instantly. is there a way how to set such a dial prefix for zulu. Or I am all wrong, and this needs to be set on freepbx server itself?

kind regards,
Dieter

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Sangoma phone only sends first DTMF tone

October 1, 2018, 10:48 am

≫ Next: OSS replaced Freepbx manager

≪ Previous: Zulu plugin dial prefix

@moonlighting wrote:

I have FreePBX running with a collection of Cisco SPA phones (508 and 525 models) and some Sangoma S500 phones. If we dial out on our phones to a number that requires keypress navigation of some sort (for sales press 1, please enter your account number, &etc.) the Cisco phones will work fine, but the Sangoma phones won’t. Further testing revealed the Sangoma phones will transmit the first key press tone, but no further tones are transmitted.

FreePBX version: 14.0.3.19
Trunk SIP settings (we use FlowRoute as our provider): dtmfmode=rfc2833
Sangoma Phone Firmware: 1.50 (2.0.4.55)
Using SIP, not PJSIP

In the config file for the Sangoma phones, here are all the entries relating to DTMF (we are only using account 1 on the phones)
101
0
0
1
1

Anyone have any suggestions on where to start troubleshooting this? Advice appreciated. Thanks in advance.

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OSS replaced Freepbx manager

October 1, 2018, 12:52 pm

≫ Next: Would it be possible to change PABX settings via CLI or changing files manually?

≪ Previous: Sangoma phone only sends first DTMF tone

@dataspeed wrote:

I ran online updates today that somehow removed my commercial version of endpoint manager and added the free OSS endpoint manager. When I putty into my system and run fwconsole ma install endpoint I get error:
Failed to install EndPoint Manager due to the following conflicting module(s): OSS PBX End Point Manager

How do I get my commercial EPM back and not lose my configs?
Please help

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Would it be possible to change PABX settings via CLI or changing files manually?

October 1, 2018, 1:13 pm

≫ Next: Fax pro email

≪ Previous: OSS replaced Freepbx manager

@viniciusbarreto wrote:

I asked, because I opened some files to make changes manually as if it were a Pure Asterisk and I came across a message from Freepbx when editing the configuration file, stating that we can not make changes directly in it …

But it turns out we’d like to automate some of the freepbx settings like queuing, creating extensions, creating routing, URA, and so on.
And it would be very formidable if it were possible to do this through a bash script by changing files with regular expressions or even through CLI, or even API.
But in order to see the changes in the freepbx web interface

Could anyone help with this?

Thank you!

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Fax pro email

October 1, 2018, 1:53 pm

≫ Next: Modify time out for a extension

≪ Previous: Would it be possible to change PABX settings via CLI or changing files manually?

@bajramia wrote:

Can i send an email to the user when they receive a fax so the can login to UCP and get it thank you.

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Modify time out for a extension

October 2, 2018, 12:41 am

≫ Next: What causing so much security events that fill fail2ban.log

≪ Previous: Fax pro email

@anscomputer wrote:

Hello,

I would like to make that :
When my extension 2080 call the extension 2081 the ring time is set to 10sec.
But when others extensions call the extension 2081 the default ring time is 40sec.

Is it possible to make that ?

Best regards,
Luca

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What causing so much security events that fill fail2ban.log

October 2, 2018, 2:43 am

≫ Next: V14 Automatic Updates Not Occurring - Modules or System

≪ Previous: Modify time out for a extension

@serinfbco wrote:

Fresh install of FreePBX 14 distro with around 300 endpoints. All is working fine but I noticed an huge amount of security events that seems coming from the Asterisk Manager. There is around 20 security events per seconds that fill the fail2ban log file. Around 1Gb at the end of the day.

FreePBX: 14.0.3.13
Asterisk: 13.22.0

Here what we see in the /var/log/asterisk/fail2ban

**[2018-10-02 05:02:02] SECURITY[2177] res_security_log.c: SecurityEvent="SuccessfulAuth",EventTV="2018-10-02T05:02:02.312-0400",Severity="Informational",Service="PJSIP",EventVersion="1",AccountID="386",SessionID="613daa02-19bf779d-d85ef138@192.18.100.174",LocalAddress="IPV4/UDP/192.18.100.2/5060",RemoteAddress="IPV4/UDP/192.18.100.174/5060",UsingPassword="1"**
**[2018-10-02 05:02:02] SECURITY[2177] res_security_log.c: SecurityEvent="ChallengeSent",EventTV="2018-10-02T05:02:02.329-0400",Severity="Informational",Service="PJSIP",EventVersion="1",AccountID="106",SessionID="17db7bdf-e56725ea-53229fc9@192.18.100.194",LocalAddress="IPV4/UDP/192.18.100.2/5060",RemoteAddress="IPV4/UDP/192.18.100.194/5060",Challenge=""**
**[2018-10-02 05:02:02] SECURITY[2177] res_security_log.c: SecurityEvent="SuccessfulAuth",EventTV="2018-10-02T05:02:02.348-0400",Severity="Informational",Service="PJSIP",EventVersion="1",AccountID="106",SessionID="17db7bdf-e56725ea-53229fc9@192.18.100.194",LocalAddress="IPV4/UDP/192.18.100.2/5060",RemoteAddress="IPV4/UDP/192.18.100.194/5060",UsingPassword="1"**
**[2018-10-02 05:02:02] SECURITY[2177] res_security_log.c: SecurityEvent="ChallengeSent",EventTV="2018-10-02T05:02:02.370-0400",Severity="Informational",Service="PJSIP",EventVersion="1",AccountID="309",SessionID="a46c9148-5e10bd4b-4f9b4492@192.18.100.133",LocalAddress="IPV4/UDP/192.18.100.2/5060",RemoteAddress="IPV4/UDP/192.18.100.133/5060",Challenge=""**

And here a sample of the traffic that is going on (tcpdump -i lo -A -s0 port 5038)

05:14:28.039964 IP localhost.53628 > localhost.5038: Flags [.], ack 74494, win 6146, options [nop,nop,TS val 1557381226 ecr 1557381226], length 0
E..4t.@.@............|..@y.k...<.....(.....
\..j\..j
05:14:28.118549 IP localhost.5038 > localhost.53628: Flags [P.], seq 74494:74808, ack 1, win 342, options [nop,nop,TS val 1557381304 ecr 1557381226], length 314
E..nkf@.@..!...........|...<@y.k...V.b.....
\...\..jEvent: ChallengeSent
Privilege: security,all
EventTV: 2018-10-02T05:14:28.118-0400
Severity: Informational
Service: PJSIP
EventVersion: 1
AccountID: 248
SessionID: 654440dd-d829c004-35872147@192.18.100.81
LocalAddress: IPV4/UDP/192.18.100.2/5060
RemoteAddress: IPV4/UDP/192.18.100.81/5060
Challenge:


05:14:28.118574 IP localhost.53628 > localhost.5038: Flags [.], ack 74808, win 6146, options [nop,nop,TS val 1557381304 ecr 1557381304], length 0
E..4t/@.@............|..@y.k...v.....(.....
\...\...
05:14:28.135823 IP localhost.5038 > localhost.53628: Flags [P.], seq 74808:75128, ack 1, win 342, options [nop,nop,TS val 1557381322 ecr 1557381304], length 320
E..tkg@.@..............|...v@y.k...V.h.....
\...\...Event: SuccessfulAuth
Privilege: security,all
EventTV: 2018-10-02T05:14:28.135-0400
Severity: Informational
Service: PJSIP
EventVersion: 1
AccountID: 248
SessionID: 654440dd-d829c004-35872147@192.18.100.81
LocalAddress: IPV4/UDP/192.18.100.2/5060
RemoteAddress: IPV4/UDP/192.18.100.81/5060
UsingPassword: 1

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V14 Automatic Updates Not Occurring - Modules or System

October 2, 2018, 5:51 am

≫ Next: Transferring calls with the BLF keys proper method?

≪ Previous: What causing so much security events that fill fail2ban.log

@mvogel4949 wrote:

I have multiple systems that have system and module updates sent to automatic and none of them are happening. Are there any logfiles I can check to see why the updates aren’t running?

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Transferring calls with the BLF keys proper method?

October 2, 2018, 6:04 am

≫ Next: Matching incoming calls from goip and freepbx14

≪ Previous: V14 Automatic Updates Not Occurring - Modules or System

@sentinelace wrote:

What is the best method for a receptionist to transfer calls using the BLF keys? They want to let the end user know they have a call. When they put the caller on hold, then push their BLF key, it transfers the calls. We currently have them pick up a line and then hit the BLF to call. Is there an easier way? We want the BLF keys so we can see who is on the phone

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Matching incoming calls from goip and freepbx14

October 2, 2018, 9:20 am

≫ Next: CDR and CEL does not appear to be inserting into database

≪ Previous: Transferring calls with the BLF keys proper method?

@uncleanon wrote:

Hello,

I’ve setup goip as a trunk to my freepbx installation everything seems to be fine. I’ve also setup an IVR and I want calls to the GSM number of goip to be answered by the IVR. It seems that I’m not able to do that. Basically I’ve configured goip to forward incoming calls to a non-existent extension on my pbx - 5000, then I have created an inbound route which matches for DID 5000 and invokes the ivr. THe sip output of an unsuccessful call is:

   > 0x7fd4041db290 -- Strict RTP learning after remote address set to: 10.25.0.47:10000
    -- Executing [5000@from-internal:1] ResetCDR("SIP/goip-trunk-1-00000025", "") in new stack
    -- Executing [5000@from-internal:2] NoCDR("SIP/goip-trunk-1-00000025", "") in new stack
    -- Executing [5000@from-internal:3] Progress("SIP/goip-trunk-1-00000025", "") in new stack
    -- Executing [5000@from-internal:4] Wait("SIP/goip-trunk-1-00000025", "1") in new stack
       > 0x7fd4041db290 -- Strict RTP switching to RTP target address 10.25.0.47:10000 as source
    -- Executing [5000@from-internal:5] Playback("SIP/goip-trunk-1-00000025", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
    -- <SIP/goip-trunk-1-00000025> Playing 'silence/1.slin16' (language 'en')
    -- <SIP/goip-trunk-1-00000025> Playing 'cannot-complete-as-dialed.slin16' (language 'en')
    -- <SIP/goip-trunk-1-00000025> Playing 'check-number-dial-again.slin16' (language 'en')
       > 0x7fd4041db290 -- Strict RTP learning complete - Locking on source address 10.25.0.47:10000
    -- Executing [5000@from-internal:6] Wait("SIP/goip-trunk-1-00000025", "1") in new stack
    -- Executing [5000@from-internal:7] Congestion("SIP/goip-trunk-1-00000025", "20") in new stack
[2018-10-02 19:09:22] WARNING[3366][C-00000023]: channel.c:5005 ast_prod: Prodding channel 'SIP/goip-trunk-1-00000025' failed
  == Spawn extension (from-internal, 5000, 7) exited non-zero on 'SIP/goip-trunk-1-00000025'
    -- Executing [h@from-internal:1] Macro("SIP/goip-trunk-1-00000025", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/goip-trunk-1-00000025", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] ExecIf("SIP/goip-trunk-1-00000025", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [s@macro-hangupcall:4] NoOp("SIP/goip-trunk-1-00000025", " monior file= ") in new stack
    -- Executing [s@macro-hangupcall:5] AGI("SIP/goip-trunk-1-00000025", "attendedtransfer-rec-restart.php,,") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
    -- <SIP/goip-trunk-1-00000025>AGI Script attendedtransfer-rec-restart.php completed, returning 0
    -- Executing [s@macro-hangupcall:6] Hangup("SIP/goip-trunk-1-00000025", "") in new stack
  == Spawn extension (macro-hangupcall, s, 6) exited non-zero on 'SIP/goip-trunk-1-00000025' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/goip-trunk-1-00000025'

However, if I change the inbound rule to match any did/cid then it works as expected:

 > 0x7fd4041e0980 -- Strict RTP learning after remote address set to: 10.25.0.47:10000
    -- Executing [5000@from-trunk:1] NoOp("SIP/goip-1-00000026", "Catch-All DID Match - Found 5000 - You probably want a DID for this.") in new stack
    -- Executing [5000@from-trunk:2] Log("SIP/goip-1-00000026", "WARNING,Friendly Scanner from 10.25.0.47") in new stack
[2018-10-02 19:10:46] WARNING[3766][C-00000024]: Ext. 5000:2 @ from-trunk: Friendly Scanner from 10.25.0.47
    -- Executing [5000@from-trunk:3] Set("SIP/goip-1-00000026", "__FROM_DID=5000") in new stack
    -- Executing [5000@from-trunk:4] Goto("SIP/goip-1-00000026", "ext-did,s,1") in new stack
    -- Goto (ext-did,s,1)
    -- Executing [s@ext-did:1] Set("SIP/goip-1-00000026", "__DIRECTION=INBOUND") in new stack
    -- Executing [s@ext-did:2] Gosub("SIP/goip-1-00000026", "app-blacklist-check,s,1()") in new stack
    -- Executing [s@app-blacklist-check:1] GotoIf("SIP/goip-1-00000026", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:2] Set("SIP/goip-1-00000026", "CALLED_BLACKLIST=1") in new stack
    -- Executing [s@app-blacklist-check:3] Return("SIP/goip-1-00000026", "") in new stack
    -- Executing [s@ext-did:3] ExecIf("SIP/goip-1-00000026", "0?Set(__FROM_DID=s)") in new stack
    -- Executing [s@ext-did:4] Set("SIP/goip-1-00000026", "CDR(did)=5000") in new stack
    -- Executing [s@ext-did:5] ExecIf("SIP/goip-1-00000026", "0 ?Set(CALLERID(name)=+359887xxxxx)") in new stack
    -- Executing [s@ext-did:6] Set("SIP/goip-1-00000026", "__MOHCLASS=") in new stack
    -- Executing [s@ext-did:7] Set("SIP/goip-1-00000026", "__REVERSAL_REJECT=FALSE") in new stack
    -- Executing [s@ext-did:8] GotoIf("SIP/goip-1-00000026", "1?post-reverse-charge") in new stack

ALso during my testing I also came accross a situation where even with the catch-all rule the call is still not handled by freepbx. My goip sip trunk is as follows:

Outgoing:
host=10.25.0.47
port=5060
context=from-internal
insecure=port,invite
qualify=yes
secret=xxxxxxxx
type=peer
nat=no
canreinvite=no
dtmfmode=rfc2833

Inbound:
Usercontext: goip-1
host=dynamic
dtmfmode=rfc2833
canreinvite=no
qualify=yes
secret=xxxxxxx
type=friend
context=from-trunk

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CDR and CEL does not appear to be inserting into database

October 2, 2018, 10:52 am

≫ Next: GV Connected but Incoming and Outgoing calls don't work

≪ Previous: Matching incoming calls from goip and freepbx14

@zfleck wrote:

Tried every solution I could find already. Sounds like this is a common issue. Let me start out by saying, I compiled the system myself on Ubuntu 18.04. Latest, up to date everything. Functionally, everything pretty much works. There are some minor kinks, as to be expected, but the system is running reliably nonetheless.

We need the system to do call recording, so CEL and CDR is important. It seems no data is being inserted into either DB. I can see in MySQL that the databases are there, and contain all fields they should, but they are just empty. When trying to pull CEL reports in FreePBX, I get the following error:

SQLSTATE[HY000]: General error: 3065 Expression #1 of ORDER BY clause is not in SELECT list, references column ‘asteriskcdrdb.cel.eventtime’ which is not in SELECT list; this is incompatible with DISTINCT
File:/var/www/html/admin/modules/cel/Cel.class.php:324

I assume this is because those databases are empty.

Like I said, I’ve tried so many solutions, checked so many things, and really can’t figure out what’s up. Everything else is working pretty okay with the database. Unsure if maybe there could be permission issues with “freepbxuser” and “asteriskcdrdb” but I’m no SQL expert and am having troubles finding ways to check that.

Any help is appreciated.

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GV Connected but Incoming and Outgoing calls don't work

October 2, 2018, 10:52 am

≫ Next: Error: Line unregistered, Polycom IP 650

≪ Previous: CDR and CEL does not appear to be inserting into database

@SPSDevon wrote:

Hey everyone, I’m using the latest addition of Incredible PBX and I I’ve got GV connected and all set up but for some reason when I place a call it rings endlessly (without actually calling anyone) and incoming calls just never come in. If anyone can help with this I would appreciate it.

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Error: Line unregistered, Polycom IP 650

October 2, 2018, 12:38 pm

≫ Next: Call Center Solution

≪ Previous: GV Connected but Incoming and Outgoing calls don't work

@travis_farmer wrote:

i finally got provisioning to work, so my phone loads. now i can’t get the phone to register with the Asterisk/FreePBX system.
I suspect i am not imputing the password correctly, or not grabbing the correct password. i have tried the password given when i created the extension, and i have tried grabbing Secret from Applications -> Extensions, from the one (so far) extension i have. i have also tried the “password for new user”. where do i grab the password from?

~Travis

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Call Center Solution

October 2, 2018, 5:54 pm

≫ Next: Show the CF status on phone with BLF

≪ Previous: Error: Line unregistered, Polycom IP 650

@bajramia wrote:

Hi All,
Im looking for call center solution like CRM we are hospital we have our scheduling department receiving calls every day i want to have a software to pop up in screen when reciving calls so agent can take notes. Next time when calles notes to pop up from previous call.

Thank you.

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Show the CF status on phone with BLF

October 3, 2018, 1:51 am

≫ Next: Outbound CID dahdi -> sip

≪ Previous: Call Center Solution

@anscomputer wrote:

Hello,

Is it possible to show the call forward status (*72) on a phone with BLF (to know if the CF is activate or no)
I use a grandstream phone (GXV3240).
If it’s possible how to make that ?

Best regards,
Luca

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Outbound CID dahdi -> sip

October 3, 2018, 2:26 am

≫ Next: Global Address Book entry with Caller ID on All Phones

≪ Previous: Show the CF status on phone with BLF

@twojtczak wrote:

Hello Everyone!

Config in this case:
Legacy('99)PBX_E1------E1(dahdi)FREEPBX_pjsip-trunk---------WORLD
sip_extension(no.55)_______________|

PBX not send CID via E1 and we want use option like a “Force trunk CID”

When we call from ext55 - all CID options working fine, I mean:
outbound route CID with “override ext.”(%202),
trunk outbound CID with “force trunk CID”(%200),
App/Set CID option (%201)

CID from WORLD to PBX work ok.

When we call from PBX to WORLD - CID is unavailable.

In extension.conf, I cant find variable CALLERID in [globals], App/Set CID option use this variable.

exten => s,n,Set(__CALLINGNAMEPRES_SV=${CALLERID(name-pres)})
exten => s,n,Set(__CALLINGNUMPRES_SV=${CALLERID(num-pres)})

This is log - call from extension 55 and from dahdi trunk.

pbx.c: Executing [0%%%%%%%26@from-internal:1] Macro(“PJSIP/55-00000c4a”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
pbx.c: Executing [s@macro-user-callerid:1] Set(“PJSIP/55-00000c4a”, “TOUCH_MONITOR=1538514526.6271”) in new stack
pbx.c: Executing [s@macro-user-callerid:2] Set(“PJSIP/55-00000c4a”, “AMPUSER=55”) in new stack
pbx.c: Executing [s@macro-user-callerid:3] GotoIf(“PJSIP/55-00000c4a”, “0?report”) in new stack
pbx.c: Executing [s@macro-user-callerid:4] ExecIf(“PJSIP/55-00000c4a”, “1?Set(REALCALLERIDNUM=55)”) in new stack
pbx.c: Executing [s@macro-user-callerid:5] Set(“PJSIP/55-00000c4a”, “AMPUSER=55”) in new stack
pbx.c: Executing [s@macro-user-callerid:6] GotoIf(“PJSIP/55-00000c4a”, “0?limit”) in new stack
pbx.c: Executing [s@macro-user-callerid:7] Set(“PJSIP/55-00000c4a”, “AMPUSERCIDNAME=piec-piec”) in new stack
pbx.c: Executing [s@macro-user-callerid:8] ExecIf(“PJSIP/55-00000c4a”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
pbx.c: Executing [s@macro-user-callerid:9] GotoIf(“PJSIP/55-00000c4a”, “0?report”) in new stack
pbx.c: Executing [s@macro-user-callerid:10] Set(“PJSIP/55-00000c4a”, “AMPUSERCID=55”) in new stack
pbx.c: Executing [s@macro-user-callerid:11] Set(“PJSIP/55-00000c4a”, “__DIAL_OPTIONS=HhTtr”) in new stack
pbx.c: Executing [s@macro-user-callerid:12] Set(“PJSIP/55-00000c4a”, “CALLERID(all)=“piec-piec” <55>”) in new stack
pbx.c: Executing [s@macro-user-callerid:13] GotoIf(“PJSIP/55-00000c4a”, “0?limit”) in new stack
pbx.c: Executing [s@macro-user-callerid:14] ExecIf(“PJSIP/55-00000c4a”, “1?Set(GROUP(concurrency_limit)=55)”) in new stack
pbx.c: Executing [s@macro-user-callerid:15] ExecIf(“PJSIP/55-00000c4a”, “0?Set(CHANNEL(language)=)”) in new stack
pbx.c: Executing [s@macro-user-callerid:16] NoOp(“PJSIP/55-00000c4a”, “Macro Depth is 1”) in new stack
pbx.c: Executing [s@macro-user-callerid:17] GotoIf(“PJSIP/55-00000c4a”, “1?report2:macroerror”) in new stack
pbx_builtins.c: Goto (macro-user-callerid,s,18)
pbx.c: Executing [s@macro-user-callerid:18] GotoIf(“PJSIP/55-00000c4a”, “1?continue”) in new stack
pbx_builtins.c: Goto (macro-user-callerid,s,37)
pbx.c: Executing [s@macro-user-callerid:37] Set(“PJSIP/55-00000c4a”, “CALLERID(number)=55”) in new stack
pbx.c: Executing [s@macro-user-callerid:38] Set(“PJSIP/55-00000c4a”, “CALLERID(name)=piec-piec”) in new stack
pbx.c: Executing [s@macro-user-callerid:39] GotoIf(“PJSIP/55-00000c4a”, “0?cnum”) in new stack
pbx.c: Executing [s@macro-user-callerid:40] Set(“PJSIP/55-00000c4a”, “CDR(cnam)=piec-piec”) in new stack
pbx.c: Executing [s@macro-user-callerid:41] Set(“PJSIP/55-00000c4a”, “CDR(cnum)=55”) in new stack
pbx.c: Executing [s@macro-user-callerid:42] Set(“PJSIP/55-00000c4a”, “CHANNEL(language)=pl”) in new stack
pbx.c: Executing [0%%%%%%%26@from-internal:2] Gosub(“PJSIP/55-00000c4a”, “sub-record-check,s,1(out,0%%%%%%%26,never)”) in new stack
pbx.c: Executing [s@sub-record-check:1] GotoIf(“PJSIP/55-00000c4a”, “0?initialized”) in new stack
pbx.c: Executing [s@sub-record-check:2] Set(“PJSIP/55-00000c4a”, “__REC_STATUS=INITIALIZED”) in new stack
pbx.c: Executing [s@sub-record-check:3] Set(“PJSIP/55-00000c4a”, “NOW=1538514526”) in new stack
pbx.c: Executing [s@sub-record-check:4] Set(“PJSIP/55-00000c4a”, “__DAY=02”) in new stack
pbx.c: Executing [s@sub-record-check:5] Set(“PJSIP/55-00000c4a”, “__MONTH=10”) in new stack
pbx.c: Executing [s@sub-record-check:6] Set(“PJSIP/55-00000c4a”, “__YEAR=2018”) in new stack
pbx.c: Executing [s@sub-record-check:7] Set(“PJSIP/55-00000c4a”, “__TIMESTR=20181002-230846”) in new stack
pbx.c: Executing [s@sub-record-check:8] Set(“PJSIP/55-00000c4a”, “__FROMEXTEN=55”) in new stack
pbx.c: Executing [s@sub-record-check:9] Set(“PJSIP/55-00000c4a”, “__MON_FMT=wav”) in new stack
pbx.c: Executing [s@sub-record-check:10] NoOp(“PJSIP/55-00000c4a”, “Recordings initialized”) in new stack
pbx.c: Executing [s@sub-record-check:11] ExecIf(“PJSIP/55-00000c4a”, “0?Set(ARG3=dontcare)”) in new stack
pbx.c: Executing [s@sub-record-check:12] Set(“PJSIP/55-00000c4a”, “REC_POLICY_MODE_SAVE=”) in new stack
pbx.c: Executing [s@sub-record-check:13] ExecIf(“PJSIP/55-00000c4a”, “0?Set(REC_STATUS=NO)”) in new stack
pbx.c: Executing [s@sub-record-check:14] GotoIf(“PJSIP/55-00000c4a”, “3?checkaction”) in new stack
pbx_builtins.c: Goto (sub-record-check,s,17)
pbx.c: Executing [s@sub-record-check:17] GotoIf(“PJSIP/55-00000c4a”, “1?sub-record-check,out,1”) in new stack
pbx_builtins.c: Goto (sub-record-check,out,1)
pbx.c: Executing [out@sub-record-check:1] NoOp(“PJSIP/55-00000c4a”, “Outbound Recording Check from 55 to 0%%%%%%%26”) in new stack
pbx.c: Executing [out@sub-record-check:2] Set(“PJSIP/55-00000c4a”, “RECMODE=dontcare”) in new stack
pbx.c: Executing [out@sub-record-check:3] ExecIf(“PJSIP/55-00000c4a”, “1?Goto(routewins)”) in new stack
pbx_builtins.c: Goto (sub-record-check,out,7)
pbx.c: Executing [out@sub-record-check:7] Gosub(“PJSIP/55-00000c4a”, “recordcheck,1(never,out,0%%%%%%%26)”) in new stack
pbx.c: Executing [recordcheck@sub-record-check:1] NoOp(“PJSIP/55-00000c4a”, “Starting recording check against never”) in new stack
pbx.c: Executing [recordcheck@sub-record-check:2] Goto(“PJSIP/55-00000c4a”, “never”) in new stack
pbx_builtins.c: Goto (sub-record-check,recordcheck,14)
pbx.c: Executing [recordcheck@sub-record-check:14] Set(“PJSIP/55-00000c4a”, “__REC_POLICY_MODE=NEVER”) in new stack
pbx.c: Executing [recordcheck@sub-record-check:15] Goto(“PJSIP/55-00000c4a”, “stoprec”) in new stack
pbx_builtins.c: Goto (sub-record-check,recordcheck,25)
pbx.c: Executing [recordcheck@sub-record-check:25] NoOp(“PJSIP/55-00000c4a”, “Stopping recording: out, 0%%%%%%%26”) in new stack
pbx.c: Executing [recordcheck@sub-record-check:26] Set(“PJSIP/55-00000c4a”, “__REC_STATUS=STOPPED”) in new stack
pbx.c: Executing [recordcheck@sub-record-check:27] System(“PJSIP/55-00000c4a”, “/var/lib/asterisk/bin/stoprecording.php “PJSIP/55-00000c4a””) in new stack
pbx.c: Executing [recordcheck@sub-record-check:28] Return(“PJSIP/55-00000c4a”, “”) in new stack
pbx.c: Executing [out@sub-record-check:8] Return(“PJSIP/55-00000c4a”, “”) in new stack
pbx.c: Executing [0%%%%%%%26@from-internal:3] ExecIf(“PJSIP/55-00000c4a”, “0 ?Set(CDR(accountcode)=)”) in new stack
pbx.c: Executing [0%%%%%%%26@from-internal:4] Set(“PJSIP/55-00000c4a”, “MOHCLASS=default”) in new stack
pbx.c: Executing [0%%%%%%%26@from-internal:5] ExecIf(“PJSIP/55-00000c4a”, “0?Set(TRUNKCIDOVERRIDE=<48%%%%%%202>)”) in new stack
pbx.c: Executing [0%%%%%%%26@from-internal:6] Set(“PJSIP/55-00000c4a”, “_NODEST=”) in new stack
pbx.c: Executing [0%%%%%%%26@from-internal:7] Macro(“PJSIP/55-00000c4a”, “dialout-trunk,1,%%%%%%%26,off”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:1] Set(“PJSIP/55-00000c4a”, “DIAL_TRUNK=1”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:2] ExecIf(“PJSIP/55-00000c4a”, “0?Set(DIAL_OPTIONS=Hhtr)”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:3] GosubIf(“PJSIP/55-00000c4a”, “0?sub-pincheck,s,1()”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:4] ExecIf(“PJSIP/55-00000c4a”, “0?Set(CALLERID(num)=55)”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:5] GotoIf(“PJSIP/55-00000c4a”, “0?disabletrunk,1”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:6] Set(“PJSIP/55-00000c4a”, “DIAL_NUMBER=%%%%%%%26”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:7] Set(“PJSIP/55-00000c4a”, “DIAL_TRUNK_OPTIONS=HhTtr”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:8] Set(“PJSIP/55-00000c4a”, “OUTBOUND_GROUP=OUT_1”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:9] Set(“PJSIP/55-00000c4a”, “DIAL_TRUNK_OPTIONS=T”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:10] GotoIf(“PJSIP/55-00000c4a”, “0?nomax”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:11] GotoIf(“PJSIP/55-00000c4a”, “0?chanfull”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:12] GotoIf(“PJSIP/55-00000c4a”, “0?skipoutcid”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:13] Macro(“PJSIP/55-00000c4a”, “outbound-callerid,1”) in new stack
pbx.c: Executing [s@macro-outbound-callerid:1] NoOp(“PJSIP/55-00000c4a”, “55”) in new stack
pbx.c: Executing [s@macro-outbound-callerid:2] NoOp(“PJSIP/55-00000c4a”, “”) in new stack
pbx.c: Executing [s@macro-outbound-callerid:3] NoOp(“PJSIP/55-00000c4a”, “all”) in new stack
pbx.c: Executing [s@macro-outbound-callerid:4] ExecIf(“PJSIP/55-00000c4a”, “0?Set(CALLERPRES(name-pres)=)”) in new stack
pbx.c: Executing [s@macro-outbound-callerid:5] ExecIf(“PJSIP/55-00000c4a”, “0?Set(CALLERPRES(num-pres)=)”) in new stack
pbx.c: Executing [s@macro-outbound-callerid:6] ExecIf(“PJSIP/55-00000c4a”, “0?Set(REALCALLERIDNUM=55)”) in new stack
pbx.c: Executing [s@macro-outbound-callerid:7] ExecIf(“PJSIP/55-00000c4a”, “0?Set(AMPUSER=55)”) in new stack
pbx.c: Executing [s@macro-outbound-callerid:8] GotoIf(“PJSIP/55-00000c4a”, “1?normcid”) in new stack
pbx_builtins.c: Goto (macro-outbound-callerid,s,12)
pbx.c: Executing [s@macro-outbound-callerid:12] Set(“PJSIP/55-00000c4a”, “USEROUTCID=“test”<48%%%%%%201>”) in new stack
pbx.c: Executing [s@macro-outbound-callerid:13] Set(“PJSIP/55-00000c4a”, “EMERGENCYCID=”) in new stack
pbx.c: Executing [s@macro-outbound-callerid:14] Set(“PJSIP/55-00000c4a”, “TRUNKOUTCID=<48%%%%%%200>”) in new stack
pbx.c: Executing [s@macro-outbound-callerid:15] GotoIf(“PJSIP/55-00000c4a”, “1?trunkcid”) in new stack
pbx_builtins.c: Goto (macro-outbound-callerid,s,20)
pbx.c: Executing [s@macro-outbound-callerid:20] ExecIf(“PJSIP/55-00000c4a”, “1?Set(CALLERID(all)=<48%%%%%%200>)”) in new stack
pbx.c: Executing [s@macro-outbound-callerid:21] ExecIf(“PJSIP/55-00000c4a”, “1?Set(CALLERID(all)=“test”<48%%%%%%201>)”) in new stack
pbx.c: Executing [s@macro-outbound-callerid:22] ExecIf(“PJSIP/55-00000c4a”, “1?Set(CALLERID(all)=<48%%%%%%200>)”) in new stack
pbx.c: Executing [s@macro-outbound-callerid:23] ExecIf(“PJSIP/55-00000c4a”, “0?Set(CALLERPRES(name-pres)=prohib_passed_screen)”) in new stack
pbx.c: Executing [s@macro-outbound-callerid:24] ExecIf(“PJSIP/55-00000c4a”, “0?Set(CALLERPRES(num-pres)=prohib_passed_screen)”) in new stack
pbx.c: Executing [s@macro-outbound-callerid:25] Set(“PJSIP/55-00000c4a”, “CDR(outbound_cnum)=48%%%%%%200”) in new stack
pbx.c: Executing [s@macro-outbound-callerid:26] Set(“PJSIP/55-00000c4a”, “CDR(outbound_cnam)=”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:14] GosubIf(“PJSIP/55-00000c4a”, “0?sub-flp-1,s,1()”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:15] Set(“PJSIP/55-00000c4a”, “OUTNUM=%%%%%%%26”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:16] Set(“PJSIP/55-00000c4a”, “custom=PJSIP”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:17] ExecIf(“PJSIP/55-00000c4a”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:18] ExecIf(“PJSIP/55-00000c4a”, “0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:19] Macro(“PJSIP/55-00000c4a”, “dialout-trunk-predial-hook,”) in new stack
pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“PJSIP/55-00000c4a”, “”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:20] GotoIf(“PJSIP/55-00000c4a”, “0?bypass,1”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:21] ExecIf(“PJSIP/55-00000c4a”, “1?Set(CONNECTEDLINE(num,i)=%%%%%%%26)”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:22] ExecIf(“PJSIP/55-00000c4a”, “1?Set(CONNECTEDLINE(name,i)=CID:48%%%%%%200)”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:23] ExecIf(“PJSIP/55-00000c4a”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)48%%%%%%200)”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:24] GotoIf(“PJSIP/55-00000c4a”, “0?customtrunk”) in new stack
pbx.c: Executing [s@macro-dialout-trunk:25] Dial(“PJSIP/55-00000c4a”, “PJSIP/%%%%%%%26@trunk.name,300,Tb(func-apply-sipheaders^s^1)”) in new stack
app_stack.c: PJSIP/trunk.name-00000c4b Internal Gosub(func-apply-sipheaders,s,1) start
pbx.c: Executing [s@func-apply-sipheaders:1] NoOp(“PJSIP/trunk.name-00000c4b”, “Applying SIP Headers to channel”) in new stack
pbx.c: Executing [s@func-apply-sipheaders:2] Set(“PJSIP/trunk.name-00000c4b”, “SIPHEADERKEYS=”) in new stack
pbx.c: Executing [s@func-apply-sipheaders:3] ExecIf(“PJSIP/trunk.name-00000c4b”, “0?Set(Rheader=1)”) in new stack
pbx.c: Executing [s@func-apply-sipheaders:4] While(“PJSIP/trunk.name-00000c4b”, “0”) in new stack
app_while.c: Jumping to priority 7
pbx.c: Executing [s@func-apply-sipheaders:8] ExecIf(“PJSIP/trunk.name-00000c4b”, “0?Set(PJSIP_HEADER(remove,Alert-Info)=)”) in new stack
pbx.c: Executing [s@func-apply-sipheaders:9] Return(“PJSIP/trunk.name-00000c4b”, “”) in new stack
app_stack.c: Spawn extension (from-pstn, 0%%%%%%%26, 1) exited non-zero on ‘PJSIP/trunk.name-00000c4b’
app_stack.c: PJSIP/trunk.name-00000c4b Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
app_dial.c: Called PJSIP/%%%%%%%26@trunk.name
app_dial.c: PJSIP/trunk.name-00000c4b is making progress passing it to PJSIP/55-00000c4a
app_macro.c: Spawn extension (macro-dialout-trunk, s, 25) exited non-zero on ‘PJSIP/55-00000c4a’ in macro ‘dialout-trunk’
pbx.c: Spawn extension (from-internal, 0%%%%%%%26, 7) exited non-zero on ‘PJSIP/55-00000c4a’
pbx.c: Executing [h@from-internal:1] Macro(“PJSIP/55-00000c4a”, “hangupcall”) in new stack
pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/55-00000c4a”, “1?theend”) in new stack
pbx_builtins.c: Goto (macro-hangupcall,s,3)
pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/55-00000c4a”, “0?Set(CDR(recordingfile)=)”) in new stack
pbx.c: Executing [s@macro-hangupcall:4] NoOp(“PJSIP/55-00000c4a”, "PJSIP/trunk.name-00000c4b monior file= ") in new stack
pbx.c: Executing [s@macro-hangupcall:5] AGI(“PJSIP/55-00000c4a”, “attendedtransfer-rec-restart.php,PJSIP/trunk.name-00000c4b,”) in new stack
res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
res_agi.c: <PJSIP/55-00000c4a>AGI Script attendedtransfer-rec-restart.php completed, returning 0
pbx.c: Executing [s@macro-hangupcall:6] Hangup(“PJSIP/55-00000c4a”, “”) in new stack
app_macro.c: Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘PJSIP/55-00000c4a’ in macro ‘hangupcall’
pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/55-00000c4a’

EDIT: log from dahdi with options in dahdi config: “Use Caller ID:” and “Use Caller ID Presentation” set to Yes

pbx.c: Executing [%%%%%326@from-digital:2] Log(“DAHDI/i1/-7”, "WARNING,Friendly Scanner from ") in new stack
Ext. %%%%%326: Friendly Scanner from
pbx.c: Executing [%%%%%326@from-digital:3] Set(“DAHDI/i1/-7”, “__FROM_DID=%%%%%326”) in new stack
pbx.c: Executing [%%%%%326@from-digital:4] Goto(“DAHDI/i1/-7”, “ext-did,s,1”) in new stack
pbx_builtins.c: Goto (ext-did,s,1)
pbx.c: Executing [s@ext-did:1] Set(“DAHDI/i1/-7”, “__DIRECTION=INBOUND”) in new stack
pbx.c: Executing [s@ext-did:2] Set(“DAHDI/i1/-7”, “CHANNEL(language)=pl”) in new stack
pbx.c: Executing [s@ext-did:3] Gosub(“DAHDI/i1/-7”, “sub-record-check,s,1(in,s,never)”) in new stack
pbx.c: Executing [s@sub-record-check:1] GotoIf(“DAHDI/i1/-7”, “0?initialized”) in new stack
pbx.c: Executing [s@sub-record-check:2] Set(“DAHDI/i1/-7”, “__REC_STATUS=INITIALIZED”) in new stack
pbx.c: Executing [s@sub-record-check:3] Set(“DAHDI/i1/-7”, “NOW=1538561299”) in new stack
pbx.c: Executing [s@sub-record-check:4] Set(“DAHDI/i1/-7”, “__DAY=03”) in new stack
pbx.c: Executing [s@sub-record-check:5] Set(“DAHDI/i1/-7”, “__MONTH=10”) in new stack
pbx.c: Executing [s@sub-record-check:6] Set(“DAHDI/i1/-7”, “__YEAR=2018”) in new stack
pbx.c: Executing [s@sub-record-check:7] Set(“DAHDI/i1/-7”, “__TIMESTR=20181003-120819”) in new stack
pbx.c: Executing [s@sub-record-check:8] Set(“DAHDI/i1/-7”, “__FROMEXTEN=unknown”) in new stack
pbx.c: Executing [s@sub-record-check:9] Set(“DAHDI/i1/-7”, “__MON_FMT=wav”) in new stack
pbx.c: Executing [s@sub-record-check:10] NoOp(“DAHDI/i1/-7”, “Recordings initialized”) in new stack
pbx.c: Executing [s@sub-record-check:11] ExecIf(“DAHDI/i1/-7”, “0?Set(ARG3=dontcare)”) in new stack
pbx.c: Executing [s@sub-record-check:12] Set(“DAHDI/i1/-7”, “REC_POLICY_MODE_SAVE=”) in new stack
pbx.c: Executing [s@sub-record-check:13] ExecIf(“DAHDI/i1/-7”, “0?Set(REC_STATUS=NO)”) in new stack
pbx.c: Executing [s@sub-record-check:14] GotoIf(“DAHDI/i1/-7”, “2?checkaction”) in new stack
pbx_builtins.c: Goto (sub-record-check,s,17)
pbx.c: Executing [s@sub-record-check:17] GotoIf(“DAHDI/i1/-7”, “1?sub-record-check,in,1”) in new stack
pbx_builtins.c: Goto (sub-record-check,in,1)
pbx.c: Executing [in@sub-record-check:1] NoOp(“DAHDI/i1/-7”, “Inbound Recording Check to s”) in new stack
pbx.c: Executing [in@sub-record-check:2] Set(“DAHDI/i1/-7”, “FROMEXTEN=unknown”) in new stack
pbx.c: Executing [in@sub-record-check:3] ExecIf(“DAHDI/i1/-7”, “0?Set(FROMEXTEN=)”) in new stack
pbx.c: Executing [in@sub-record-check:4] Gosub(“DAHDI/i1/-7”, “recordcheck,1(never,in,s)”) in new stack
pbx.c: Executing [recordcheck@sub-record-check:1] NoOp(“DAHDI/i1/-7”, “Starting recording check against never”) in new stack
pbx.c: Executing [recordcheck@sub-record-check:2] Goto(“DAHDI/i1/-7”, “never”) in new stack
pbx_builtins.c: Goto (sub-record-check,recordcheck,14)
pbx.c: Executing [recordcheck@sub-record-check:14] Set(“DAHDI/i1/-7”, “__REC_POLICY_MODE=NEVER”) in new stack
pbx.c: Executing [recordcheck@sub-record-check:15] Goto(“DAHDI/i1/-7”, “stoprec”) in new stack
pbx_builtins.c: Goto (sub-record-check,recordcheck,25)
pbx.c: Executing [recordcheck@sub-record-check:25] NoOp(“DAHDI/i1/-7”, “Stopping recording: in, s”) in new stack
pbx.c: Executing [recordcheck@sub-record-check:26] Set(“DAHDI/i1/-7”, “__REC_STATUS=STOPPED”) in new stack
pbx.c: Executing [recordcheck@sub-record-check:27] System(“DAHDI/i1/-7”, "/var/lib/asterisk/bin/stoprecording.php “DAHDI/i1/-7"”) in new stack
pbx.c: Executing [recordcheck@sub-record-check:28] Return(“DAHDI/i1/-7”, “”) in new stack
pbx.c: Executing [in@sub-record-check:5] Return(“DAHDI/i1/-7”, “”) in new stack
pbx.c: Executing [s@ext-did:4] Set(“DAHDI/i1/-7”, “CHANNEL(tonezone)=pl”) in new stack
pbx.c: Executing [s@ext-did:5] ExecIf(“DAHDI/i1/-7”, “0?Set(__FROM_DID=s)”) in new stack
pbx.c: Executing [s@ext-did:6] Set(“DAHDI/i1/-7”, “returnhere=1”) in new stack
pbx.c: Executing [s@ext-did:7] Gosub(“DAHDI/i1/-7”, “app-blacklist-check,s,1()”) in new stack
pbx.c: Executing [s@app-blacklist-check:1] GotoIf(“DAHDI/i1/-7”, “0?blacklisted”) in new stack
pbx.c: Executing [s@app-blacklist-check:2] Set(“DAHDI/i1/-7”, “CALLED_BLACKLIST=1”) in new stack
pbx.c: Executing [s@app-blacklist-check:3] Return(“DAHDI/i1/-7”, “”) in new stack
pbx.c: Executing [s@ext-did:8] Set(“DAHDI/i1/-7”, “CDR(did)=%%%%%326”) in new stack
pbx.c: Executing [s@ext-did:9] GotoIf(“DAHDI/i1/-7”, “0?”) in new stack
pbx.c: Executing [s@ext-did:10] ExecIf(“DAHDI/i1/-7”, “1 ?Set(CALLERID(name)=)”) in new stack
pbx.c: Executing [s@ext-did:11] Set(“DAHDI/i1/-7”, “__MOHCLASS=”) in new stack
pbx.c: Executing [s@ext-did:12] Set(“DAHDI/i1/-7”, “__REVERSAL_REJECT=FALSE”) in new stack
pbx.c: Executing [s@ext-did:13] GotoIf(“DAHDI/i1/-7”, “1?post-reverse-charge”) in new stack
pbx_builtins.c: Goto (ext-did,s,15)
pbx.c: Executing [s@ext-did:15] NoOp(“DAHDI/i1/-7”, “”) in new stack
pbx.c: Executing [s@ext-did:16] Set(“DAHDI/i1/-7”, “__CALLINGNAMEPRES_SV=unavailable”) in new stack
pbx.c: Executing [s@ext-did:17] Set(“DAHDI/i1/-7”, “__CALLINGNUMPRES_SV=unavailable”) in new stack
pbx.c: Executing [s@ext-did:18] Set(“DAHDI/i1/-7”, “CALLERID(name-pres)=allowed_not_screened”) in new stack
pbx.c: Executing [s@ext-did:19] Set(“DAHDI/i1/-7”, “CALLERID(num-pres)=allowed_not_screened”) in new stack
pbx.c: Executing [s@ext-did:20] NoOp(“DAHDI/i1/-7”, “CallerID Entry Point”) in new stack
pbx.c: Executing [s@ext-did:21] Goto(“DAHDI/i1/-7”, “ext-trunk,1,1”) in new stack
pbx_builtins.c: Goto (ext-trunk,1,1)
pbx.c: Executing [1@ext-trunk:1] Set(“DAHDI/i1/-7”, “TDIAL_SUFFIX=@TRUNK.NAME”) in new stack
pbx.c: Executing [1@ext-trunk:2] Set(“DAHDI/i1/-7”, “TDIAL_STRING=PJSIP”) in new stack
pbx.c: Executing [1@ext-trunk:3] Set(“DAHDI/i1/-7”, “DIAL_TRUNK=1”) in new stack
pbx.c: Executing [1@ext-trunk:4] Goto(“DAHDI/i1/-7”, “ext-trunk,tdial,1”) in new stack
pbx_builtins.c: Goto (ext-trunk,tdial,1)
pbx.c: Executing [tdial@ext-trunk:1] Set(“DAHDI/i1/-7”, “OUTBOUND_GROUP=OUT_1”) in new stack
pbx.c: Executing [tdial@ext-trunk:2] GotoIf(“DAHDI/i1/-7”, “0?nomax”) in new stack
pbx.c: Executing [tdial@ext-trunk:3] GotoIf(“DAHDI/i1/-7”, “0?hangit”) in new stack
pbx.c: Executing [tdial@ext-trunk:4] ExecIf(“DAHDI/i1/-7”, “1?Set(CALLERPRES(name-pres)=unavailable)”) in new stack
pbx.c: Executing [tdial@ext-trunk:5] ExecIf(“DAHDI/i1/-7”, “1?Set(CALLERPRES(num-pres)=unavailable)”) in new stack
pbx.c: Executing [tdial@ext-trunk:6] Set(“DAHDI/i1/-7”, “DIAL_NUMBER=%%%%%326”) in new stack
pbx.c: Executing [tdial@ext-trunk:7] GosubIf(“DAHDI/i1/-7”, “0?sub-flp-1,s,1()”) in new stack
pbx.c: Executing [tdial@ext-trunk:8] Set(“DAHDI/i1/-7”, “OUTNUM=%%%%%326”) in new stack
pbx.c: Executing [tdial@ext-trunk:9] Set(“DAHDI/i1/-7”, “DIAL_TRUNK_OPTIONS=T”) in new stack
pbx.c: Executing [tdial@ext-trunk:10] Dial(“DAHDI/i1/-7”, “PJSIP/%%%%%326@TRUNK.NAME,300,T”) in new stack
app_dial.c: Called PJSIP/%%%%%326@TRUNK.NAME
app_dial.c: PJSIP/TRUNK.NAME-00000006 is making progress passing it to DAHDI/i1/-7
sig_pri.c: Span 1: Channel 0/4 got hangup request, cause 31
bridge_channel.c: Channel DAHDI/i1/%%%%%%900-6 left ‘simple_bridge’ basic-bridge
chan_dahdi.c: Hungup ‘DAHDI/i1/%%%%%%900-6’
bridge_channel.c: Channel PJSIP/TRUNK.NAME-00000005 left ‘simple_bridge’ basic-bridge
pbx.c: Spawn extension (ext-trunk, tdial, 10) exited non-zero on ‘PJSIP/TRUNK.NAME-00000005’

What should i do?
Please help.

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Global Address Book entry with Caller ID on All Phones

October 3, 2018, 6:10 am

≫ Next: User Manager not sending Email - V14

≪ Previous: Outbound CID dahdi -> sip

@sentinelace wrote:

I have a customer that needs a global address book on all phones. How do we import this? The asterisk addressbook appears to be just for speed dials. They don’t need speed dial. Users need to be able to access this list from there phones like a cell phone and click and dial. Also when a user calls in thats in that list, that caller ID name shows up on the phone. Any thoughts?

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User Manager not sending Email - V14

October 3, 2018, 6:33 am

≫ Next: Asterisk 15 Security Fix Only Mode

≪ Previous: Global Address Book entry with Caller ID on All Phones

@mvogel4949 wrote:

I’ve noticed that when I use the Quick Create Extension and select Yes for User Manager the user does not receive the email with the initial password as it has in the past. If I go to User Manager and email users it will send them an email with an obfuscation as the password. Any idea why the first email isn’t going out?

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Asterisk 15 Security Fix Only Mode

October 3, 2018, 7:09 am

≫ Next: Can not transfer inbound call

≪ Previous: User Manager not sending Email - V14

@jfinstrom wrote:

From Matt Fredrickson via the Asterisk Developer mailing list…

Hey All,

Just a quick reminder to all of you. As of today, the Asterisk 15
branch transitions into security fix only mode. What this means is
that that 15 branch will stop receiving normal bug-fixes/contributions
and only receive security patches. For those of you that are on this
release, it’s a good time to start planning and evaluating a path
forward to the up-and-coming Asterisk 16.0.0 release.

If you have any questions about the schedule for these types of events
and when this may happen to any other branches/versions, you can read
more about it on the wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

Best wishes to you all, and hope to see many of you at Astricon

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Can not transfer inbound call

October 3, 2018, 8:10 am

≫ Next: FWconsole trunk commands from the command line

≪ Previous: Asterisk 15 Security Fix Only Mode

@genscript wrote:

Hi,
We missed a call from 44xxxxxxxxxx. CDR shows me that the call was during 13:59. But our phone never rang
at that time. And the caller got the error “temporarily not available”.
And the CDR shows me the App was Playback and Destination was ‘t’.
The ISP check the RTP was normal.
Could you give me some ideas about troubleshoot?

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