I am using Asterisk via FreePbx to implement B2BUA.
As part of my task, I have created an Inbound Route, and set the Destination=Trunk, and selected one of my trunks with correct SIP credentials.
Everything seems to work fine except one sad issue.
When Asterisk dials the target SIP trunk, it prompts "Confirm Call" there, asking the destination side to press 1 to accept the call.
I need to remove this prompt.
It sounds stupid, but I can not find a way to do it anywhere in FreePbx Web GUI or in FreePbx online documentation.
Can someone suggest a solution to turn this Confirm Call feature Off for my FreePBX SIP trunk?
Some destination trunk settings
Asterisk Trunk Dial Options:
SIP/username#hostname.com
Sip Settings/Outgoing/Peer Details:
host=hostname.com
username=username
type=peer
port=5060
transport=tcp
tcpenable=yes
privacy=off
Piece of log showing the problem
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] pbx.c: Executing [tdial#ext-trunk:10] Dial("SIP/InTrunk-000013aa", "SIP/OutTrunk/username,300,SIP/username#hostname.com") in new stack
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] app_dial.c: Privacy DB is 'tdial', clid is '+18578888888'
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] netsock2.c: Using SIP RTP TOS bits 184
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] netsock2.c: Using SIP RTP CoS mark 5
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] app_dial.c: Called SIP/OutTrunk/username
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] res_musiconhold.c: Started music on hold, class 'none', on channel 'SIP/InTrunk-000013aa'
[2017-09-06 15:25:21] WARNING[16408][C-0000a153] format_wav.c: Read failed (type)
[2017-09-06 15:25:21] WARNING[16408][C-0000a153] file.c: Unable to open format wav
[2017-09-06 15:25:21] WARNING[16408][C-0000a153] res_musiconhold.c: Unable to open file '/var/lib/asterisk/moh/.nomusic_reserved/silence': No such file or directory
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] res_musiconhold.c: Stopped music on hold on SIP/InTrunk-000013aa
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] app_dial.c: SIP/OutTrunk-000013ab is ringing
[2017-09-06 15:25:23] VERBOSE[16408][C-0000a153] app_dial.c: SIP/OutTrunk-000013ab answered SIP/InTrunk-000013aa
[2017-09-06 15:25:23] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callpending.ulaw' (language 'en')
[2017-09-06 15:25:27] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callerintros/+18578888888.slin' (language 'en')
[2017-09-06 15:25:32] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callee-options.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'screen-callee-options.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'vm-sorry.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callerintros/+18578888888.slin' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callee-options.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'screen-callee-options.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'vm-sorry.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callerintros/+18578888888.slin' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callee-options.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'screen-callee-options.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'vm-sorry.ulaw' (language 'en')
[2017-09-06 15:25:41] NOTICE[16408][C-0000a153] app_dial.c: privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] pbx.c: Executing [tdial#ext-trunk:11] Set("SIP/InTrunk-000013aa", "CALLERID(number)=+18578888888") in new stack
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] pbx.c: Executing [tdial#ext-trunk:12] Set("SIP/InTrunk-000013aa", "CALLERID(name)=+18578888888") in new stack
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] pbx.c: Executing [tdial#ext-trunk:13] Hangup("SIP/InTrunk-000013aa", "") in new stack
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] pbx.c: Spawn extension (ext-trunk, tdial, 13) exited non-zero on 'SIP/InTrunk-000013aa'
By default freepbx not do confirm call. Check privacy settings,check "dial options" on trunk.
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] app_dial.c: Privacy DB is 'tdial', clid is '+18578888888'
You log is usless, use "core set verbose 3" to see dialplan transitions.
Related
I am having an issue/trying to make something work within Asterisk. I have a trunk to an Ascom Nurse call system and there is a basic function to dial from a handheld device into a patient room. I am able to establish the call from Asterisk to the nurse call server. The nurse call server sends a Refer message
to a different address on the same subnet but Asterisk cannot find that device. If I manually type a dialplan to route calls to that device it does connect but the system could have many addresses and it would be impossible to tell what address and dial number would be in the Refer message
Asterik 10.2.87.201
Ascom 10.2.87.1
Refer Message refer-to: sip:V1003B0G65605773L0#10.2.87.11
I need to be able to have Asterisk transfer to that ext and domain dynamically.
This is a working example with a dialplan telling the system to manually send to the 10.2.87.11 address
exten => _VX.,n,Dial(SIP/${EXTEN}#10.2.87.11)
-- Executing [201*65609848#default:2] Dial("SIP/1341-000001cd", "SIP/T6/201*65609848") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/T6/201*65609848
> 0x7fba38016560 -- Strict RTP learning after remote address set to: 10.2.87.1:8766
-- SIP/T6-000001ce answered SIP/1341-000001cd
-- Channel SIP/T6-000001ce joined 'simple_bridge' basic-bridge <07b9ccbf-138d-423b-87cf-ad6c41336591>
-- Channel SIP/1341-000001cd joined 'simple_bridge' basic-bridge <07b9ccbf-138d-423b-87cf-ad6c41336591>
> Bridge 07b9ccbf-138d-423b-87cf-ad6c41336591: switching from simple_bridge technology to native_rtp
> Locally RTP bridged 'SIP/1341-000001cd' and 'SIP/T6-000001ce' in stack
> 0x7fb9c017aa00 -- Strict RTP learning after remote address set to: 192.168.21.82:16734
> Locally RTP bridged 'SIP/1341-000001cd' and 'SIP/T6-000001ce' in stack
> 0x7fb9c017aa00 -- Strict RTP switching to RTP target address 192.168.21.82:16734 as source
> Locally RTP bridged 'SIP/1341-000001cd' and 'SIP/T6-000001ce' in stack
-- Channel SIP/T6-000001ce left 'native_rtp' basic-bridge <07b9ccbf-138d-423b-87cf-ad6c41336591>
> Bridge 07b9ccbf-138d-423b-87cf-ad6c41336591: switching from native_rtp technology to simple_bridge
-- Channel SIP/1341-000001cd left 'simple_bridge' basic-bridge <07b9ccbf-138d-423b-87cf-ad6c41336591>
-- Executing [V1003B0G65609848L0#default:1] NoOp("SIP/1341-000001cd", "V1003B0G65609848L0") in new stack
-- Executing [V1003B0G65609848L0#default:2] Dial("SIP/1341-000001cd", "SIP/V1003B0G65609848L0#10.2.87.11") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/V1003B0G65609848L0#10.2.87.11
> 0x7fb9c017aa00 -- Strict RTP learning complete - Locking on source address 192.168.21.82:16734
> 0x7fba3802cb80 -- Strict RTP learning after remote address set to: 10.2.87.11:5012
-- SIP/10.2.87.11-000001cf answered SIP/1341-000001cd
-- Channel SIP/10.2.87.11-000001cf joined 'simple_bridge' basic-bridge <149e09d8-7d19-46fb-9135-43ffa33862e1>
-- Channel SIP/1341-000001cd joined 'simple_bridge' basic-bridge <149e09d8-7d19-46fb-9135-43ffa33862e1>
> Bridge 149e09d8-7d19-46fb-9135-43ffa33862e1: switching from simple_bridge technology to native_rtp
> Locally RTP bridged 'SIP/1341-000001cd' and 'SIP/10.2.87.11-000001cf' in stack
> 0x7fba3802cb80 -- Strict RTP switching to RTP target address 10.2.87.11:5012 as source
> 0x7fba3802cb80 -- Strict RTP learning complete - Locking on source address 10.2.87.11:5012
-- Channel SIP/1341-000001cd left 'native_rtp' basic-bridge <149e09d8-7d19-46fb-9135-43ffa33862e1>
-- Channel SIP/10.2.87.11-000001cf left 'native_rtp' basic-bridge <149e09d8-7d19-46fb-9135-43ffa33862e1>
== Spawn extension (default, V1003B0G65609848L0, 2) exited non-zero on 'SIP/1341-000001cd'
Not working when using a Transfer dialplan
> 0x7fb9c017aa00 -- Strict RTP learning after remote address set to: 192.168.21.82:16766
-- Executing [201*65609851#default:1] NoOp("SIP/1341-000001de", "201*65609851") in new stack
-- Executing [201*65609851#default:2] Dial("SIP/1341-000001de", "SIP/T6/201*65609851") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/T6/201*65609851
> 0x7fba4001ebf0 -- Strict RTP learning after remote address set to: 10.2.87.1:8766
-- SIP/T6-000001df answered SIP/1341-000001de
-- Channel SIP/T6-000001df joined 'simple_bridge' basic-bridge <9b4ac0b8-3bcf-4ef3-862a-46b36a7cddab>
-- Channel SIP/1341-000001de joined 'simple_bridge' basic-bridge <9b4ac0b8-3bcf-4ef3-862a-46b36a7cddab>
> Bridge 9b4ac0b8-3bcf-4ef3-862a-46b36a7cddab: switching from simple_bridge technology to native_rtp
> Locally RTP bridged 'SIP/1341-000001de' and 'SIP/T6-000001df' in stack
> 0x7fb9c017aa00 -- Strict RTP learning after remote address set to: 192.168.21.82:16766
> Locally RTP bridged 'SIP/1341-000001de' and 'SIP/T6-000001df' in stack
> 0x7fb9c017aa00 -- Strict RTP switching to RTP target address 192.168.21.82:16766 as source
> Locally RTP bridged 'SIP/1341-000001de' and 'SIP/T6-000001df' in stack
-- Channel SIP/T6-000001df left 'native_rtp' basic-bridge <9b4ac0b8-3bcf-4ef3-862a-46b36a7cddab>
> Bridge 9b4ac0b8-3bcf-4ef3-862a-46b36a7cddab: switching from native_rtp technology to simple_bridge
-- Channel SIP/1341-000001de left 'simple_bridge' basic-bridge <9b4ac0b8-3bcf-4ef3-862a-46b36a7cddab>
-- Executing [V1003B0G65609851L0#default:1] NoOp("SIP/1341-000001de", "V1003B0G65609851L0") in new stack
-- Executing [V1003B0G65609851L0#default:2] Transfer("SIP/1341-000001de", "SIP/V1003B0G65609851L0") in new stack
== Using SIP RTP CoS mark 5
> 0x7fb9c0174940 -- Strict RTP learning after remote address set to: 192.168.21.82:16770
-- Executing [V1003B0G65609851L0#default:1] NoOp("SIP/1341-000001e0", "V1003B0G65609851L0") in new stack
-- Executing [V1003B0G65609851L0#default:2] Transfer("SIP/1341-000001e0", "SIP/V1003B0G65609851L0") in new stack
-- Auto fallthrough, channel 'SIP/1341-000001e0' status is 'UNKNOWN'
== Using SIP RTP CoS mark 5
> 0x7fb9c0174940 -- Strict RTP learning after remote address set to: 192.168.21.82:16770
-- Executing [V1003B0G65609851L0#default:1] NoOp("SIP/1341-000001e1", "V1003B0G65609851L0") in new stack
-- Executing [V1003B0G65609851L0#default:2] Transfer("SIP/1341-000001e1", "SIP/V1003B0G65609851L0") in new stack
-- Auto fallthrough, channel 'SIP/1341-000001e1' status is 'UNKNOWN'
== Using SIP RTP CoS mark 5
> 0x7fb9c0174940 -- Strict RTP learning after remote address set to: 192.168.21.82:16770
-- Executing [V1003B0G65609851L0#default:1] NoOp("SIP/1341-000001e2", "V1003B0G65609851L0") in new stack
-- Executing [V1003B0G65609851L0#default:2] Transfer("SIP/1341-000001e2", "SIP/V1003B0G65609851L0") in new stack
-- Auto fallthrough, channel 'SIP/1341-000001e2' status is 'UNKNOWN'
> 0x7fb9c0174940 -- Strict RTP learning after remote address set to: 192.168.21.82:16770
-- Auto fallthrough, channel 'SIP/1341-000001de' status is 'ANSWER'
In short I need the last example to dynamically send to the domain in the refer message.
Thank you for you help
I am not sure if this is correct but I got this to work by creating users for all the devices I needed to communicate to and adjusting my extensions.conf
exten => _VX.,n,Dial(SIP/${EXTEN}#${SIPDOMAIN})
fullname = 10.2.87.10
secret =
hasvoicemail = no
host = 10.2.87.10
userqphone = yes
qualify = no
hassip = yes
hasiax = no
callwaiting = yes
context = default
Consider the following asterisk dialplan. After the last sound file is played it hangs up immediately and the waitexten timeout parameter seems to be ignored. The behaviour I am trying to achieve is to wait a given number of seconds after the last sound file is completed to get a response and then hang up. The last sound file says "Dial * to hear these options again". If I set the waitexten timeout to say 60, it hangs up before the sound files are played. A shorter time allows them all to be played but then it hangs up immediately. Any suggestions for handling this a better way are welcome
[mainmenu]
exten => s,1,Wait(0.25)
same => 2,Answer()
same => 3,Background(/srv/asterisk/sounds/optionslist)
same => n,Background(/srv/asterisk/sounds/dial2cs)
same => n,Background(/srv/asterisk/sounds/dial3ma)
same => n,Background(/srv/asterisk/sounds/dial4ac)
same => n,Background(/srv/asterisk/sounds/dial0)
same => n,Background(/srv/asterisk/sounds/dialstar)
same => n,WaitExten(20)
exten => 2,1,Goto(cs,2,1)
exten => *,1,Goto(s,3)
console output
== Using SIP RTP CoS mark 5
-- Executing [+12345#public:1] Goto("SIP/xxx.pstn.twilio.com-00000044", "mainmenu,s,1") in new stack
-- Goto (mainmenu,s,1)
-- Executing [s#mainmenu:1] Wait("SIP/xxx.pstn.twilio.com-00000044", "0.25") in new stack
-- Executing [s#mainmenu:4] BackGround("SIP/xxx.pstn.twilio.com-00000044", "/srv/asterisk/sounds/dial2cs") in new stack
-- Executing [s#mainmenu:5] BackGround("SIP/xxx.pstn.twilio.com-00000044", "/srv/asterisk/sounds/dial3ma") in new stack
-- <SIP/xxx.pstn.twilio.com-00000044> Playing '/srv/asterisk/sounds/dial3ma.slin' (language 'en')
-- Executing [s#mainmenu:6] BackGround("SIP/xxx.pstn.twilio.com-00000044", "/srv/asterisk/sounds/dial4ac") in new stack
-- <SIP/xxx.pstn.twilio.com-00000044> Playing '/srv/asterisk/sounds/dial4ac.slin' (language 'en')
-- Executing [s#mainmenu:7] BackGround("SIP/xxx.pstn.twilio.com-00000044", "/srv/asterisk/sounds/dial0") in new stack
-- <SIP/xxx.pstn.twilio.com-00000044> Playing '/srv/asterisk/sounds/dial0.slin' (language 'en')
-- Executing [s#mainmenu:8] BackGround("SIP/xxx.pstn.twilio.com-00000044", "/srv/asterisk/sounds/dialstar") in new stack
-- <SIP/xxx.pstn.twilio.com-00000044> Playing '/srv/asterisk/sounds/dialstar.slin' (language 'en')
[
[Aug 4 05:03:28] WARNING[2225]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission xxx#0.0.0.0 for seqno 5305 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32001ms with no response
[Aug 4 05:03:28] WARNING[2225]: chan_sip.c:4204 retrans_pkt: Hanging up call xxx#0.0.0.0 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
== Spawn extension (mainmenu, s, 8) exited non-zero on 'SIP/xxx.pstn.twilio.com-00000044'
Asterisk 11.7.0~dfsg-1ubuntu1
I assume that you have a problem like in this topic Asterisk,SIP Retransmission timeout. Try to solve the problem with the NAT settings or firewall.
In this answer propose to set canreinvite=no in sip.conf.
I am currently working with freepbx version 2.11.038 using asterisk version 11.6. I am trying to set up an English Extension and a Spanish Extension ie when you ring one number you hit the ext 100049 which is for English and when you ring the other you hit ext 100050 which of for Spanish.
I need the voicemails of these extensions to be in the required language ie 100049 has English voice prompts and 100050 has Spanish voice prompts.
Currently when ringing both I get an English voice.
In the Settings->Advanced Sip Settings, I can set the language to say es. But then all the voice prompts are changed to Spanish on a global scale. I am looking to only do a single extension.
In Settings->Voicemail admin, I can set the various extensions voicemail language using a language code. The issue is that when a user calls both extensions they get the English voice prompts still. However if 100049 dial "*97" for voicemail they hear a password request in English, if 100050 dial "*97" for voicemail they are requested the password in Spanish. So this voicemail settings only seems to affect when a user dials into their own voicemail and not when an external calls hits their voicemail.
Here is my structure of the sound files
Here are my Extensions
Today I have installed the FreePBX language module, which seems to allow me to enter the extension language on the extension page rather than via the voicemail admin, but the result is the same, ext to "*97" language correct, external calls still gets default language.
Here are the log files for the calls made to ext 100049, the English Extension
-- Executing [vmx#macro-vm:10] NoOp("SIP/NodoProvicnial-000001cd", "Checking if ext 100049 is enabled: ") in new stack
-- Executing [vmx#macro-vm:11] GotoIf("SIP/NodoProvicnial-000001cd", "1?s-CHANUNAVAIL,1") in new stack
-- Goto (macro-vm,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL#macro-vm:1] Macro("SIP/NodoProvicnial-000001cd","get-vmcontext,100049") in new stack
-- Executing [s#macro-get-vmcontext:1] Set("SIP/NodoProvicnial-000001cd", "VMCONTEXT=default") in new stack
-- Executing [s#macro-get-vmcontext:2] GotoIf("SIP/NodoProvicnial-000001cd","0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing [s#macro-get-vmcontext:300] NoOp("SIP/NodoProvicnial-000001cd","") in new stack
-- Executing [s-CHANUNAVAIL#macro-vm:2] VoiceMail("SIP/NodoProvicnial-000001cd", "100049#default,u") in new stack
-- <SIP/NodoProvicnial-000001cd> Playing 'vm-theperson.gsm' (language 'en')[2014-08-07 11:10:52] NOTICE[4184][C-000000d1]: channel.c:4259 __ast_read: Dropping incompatible voice frame on SIP/NodoProvicnial-000001cd of format g729 since our native format has changed to (alaw)
-- <SIP/NodoProvicnial-000001cd> Playing 'digits/1.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001cd> Playing 'digits/0.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001cd> Playing 'digits/0.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001cd> Playing 'digits/0.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001cd> Playing 'digits/4.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001cd> Playing 'digits/9.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001cd> Playing 'vm-isunavail.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001cd> Playing 'vm-intro.gsm' (language 'en')
== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/NodoProvicnial-000001cd' in macro 'vm'
== Spawn extension (macro-exten-vm, s, 14) exited non-zero on 'SIP/NodoProvicnial-000001cd' in macro 'exten-vm'
== Spawn extension (from-did-direct, 100049, 2) exited non-zero on 'SIP/NodoProvicnial-000001cd'
-- Executing [h#from-did-direct:1] Macro("SIP/NodoProvicnial-000001cd", "hangupcall,") in new stack
-- Executing [s#macro-hangupcall:1] GotoIf("SIP/NodoProvicnial-000001cd", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s#macro-hangupcall:3] ExecIf("SIP/NodoProvicnial-000001cd", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s#macro-hangupcall:4] Hangup("SIP/NodoProvicnial-000001cd", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/NodoProvicnial-000001cd' in macro 'hangupcall'
== Spawn extension (from-did-direct, h, 1) exited non-zero on 'SIP/NodoProvicnial-000001cd'
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/NodoProvicnial-000001cd
== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/NodoProvicnial-000001cd' in macro 'vm'
== Spawn extension (macro-exten-vm, s, 14) exited non-zero on 'SIP/NodoProvicnial-000001cd' in macro 'exten-vm'
== Spawn extension (from-did-direct, 100049, 2) exited non-zero on 'SIP/NodoProvicnial-000001cd'
-- Executing [h#from-did-direct:1] Macro("SIP/NodoProvicnial-000001cd", "hangupcall,") in new stack
-- Executing [s#macro-hangupcall:1] GotoIf("SIP/NodoProvicnial-000001cd", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s#macro-hangupcall:3] ExecIf("SIP/NodoProvicnial-000001cd", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s#macro-hangupcall:4] Hangup("SIP/NodoProvicnial-000001cd", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/NodoProvicnial-000001cd' in macro 'hangupcall'
== Spawn extension (from-did-direct, h, 1) exited non-zero on 'SIP/NodoProvicnial-000001cd'
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/NodoProvicnial-000001cd
Here are the logs for ext 100050 ie Spanish Extension:
-- Executing [vmx#macro-vm:10] NoOp("SIP/NodoProvicnial-000001d0", "Checking if ext 100050is enabled: ") in new stack
-- Executing [vmx#macro-vm:11] GotoIf("SIP/NodoProvicnial-000001d0", "1?s-NOANSWER,1") in new stack
-- Goto (macro-vm,s-NOANSWER,1)
-- Executing [s-NOANSWER#macro-vm:1] Macro("SIP/NodoProvicnial-000001d0", "get-vmcontext,100050") in new stack
-- Executing [s#macro-get-vmcontext:1] Set("SIP/NodoProvicnial-000001d0","VMCONTEXT=default") in new stack
-- Executing [s#macro-get-vmcontext:2] GotoIf("SIP/NodoProvicnial-000001d0", "0?00:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing [s#macro-get-vmcontext:300] NoOp("SIP/NodoProvicnial-000001d0", "") in new stack
-- Executing [s-NOANSWER#macro-vm:2] VoiceMail("SIP/NodoProvicnial-000001d0","100050#default,u") in new stack
-- <SIP/NodoProvicnial-000001d0> Playing 'vm-theperson.gsm' (language 'en')[2014-08-07 11:14:50] NOTICE[4191][C-000000d3]: channel.c:4259 __ast_read: Dropping incompatible voice frame on SIP/NodoProvicnial-000001d0 of format g729 since our native format has changed to (alaw)
-- <SIP/NodoProvicnial-000001d0> Playing 'digits/1.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001d0> Playing 'digits/0.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001d0> Playing 'digits/0.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001d0> Playing 'digits/0.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001d0> Playing 'digits/5.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001d0> Playing 'digits/0.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001d0> Playing 'vm-isunavail.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001d0> Playing 'vm-intro.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001d0> Playing 'beep.gsm' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/100050/tmp/JKE7il format: wav49, 0x7fccb8064b98
-- x=1, open writing: /var/spool/asterisk/voicemail/default/100050/tmp/JKE7il format: gsm, 0x7fccb8060a78
-- x=2, open writing: /var/spool/asterisk/voicemail/default/100050/tmp/JKE7il format: wav, 0x7fccb807eac8
-- User hung up
I have asked this on the FreePBX forum also please see the following link.
FreePbx Forum Post
Any advice would be great as I am at a bit of a loss. Any other information I can provide please let me know.
You have download module called "Languages" via module admin and setup other language.
After that you can change language in dialplan, for example select in ivr on press 1 language es and go next destination as you set it.
Also you can change language in extensions tab, that will work for outbound calls from thoose extension.
I am using asterisk 1.8 trying to save voicemail details in database but cli shows this
[Apr 4 18:11:54] NOTICE[6879][C-00000006]: chan_sip.c:25503 handle_request_invite: Call from '' (37.59.2.156:5071) to extension '9900972597869877' rejected because extension not found in context 'default'.
-- <DAHDI/i1/9634434640-3> Playing 'digits/4.gsm' (language 'yes')
-- <DAHDI/i1/9634434640-3> Playing 'digits/7.gsm' (language 'yes')
-- <DAHDI/i1/9634434640-3> Playing 'digits/5.gsm' (language 'yes')
-- <DAHDI/i1/9634434640-3> Playing 'digits/9.gsm' (language 'yes')
-- <DAHDI/i1/9634434640-3> Playing 'digits/5.gsm' (language 'yes')
-- <DAHDI/i1/9634434640-3> Playing 'digits/0.gsm' (language 'yes')
-- <DAHDI/i1/9634434640-3> Playing 'digits/0.gsm' (language 'yes')
-- <DAHDI/i1/9634434640-3> Playing 'vm-isunavail.gsm' (language 'yes')
-- <DAHDI/i1/9634434640-3> Playing 'vm-intro.gsm' (language 'yes')
-- <DAHDI/i1/9634434640-3> Playing 'beep.gsm' (language 'yes')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/4759500/tmp/E2vm3O format: wav49, 0xb7416ecc
-- x=1, open writing: /var/spool/asterisk/voicemail/default/4759500/tmp/E2vm3O format: gsm, 0xb741f5b4
-- x=2, open writing: /var/spool/asterisk/voicemail/default/4759500/tmp/E2vm3O format: wav, 0xb741fcdc
-- Span 1: Channel 0/1 got hangup request, cause 16
-- User hung up
== Parsing '/var/spool/asterisk/voicemail/default/4759500/INBOX/msg0000.txt': Found
[Apr 4 18:12:14] WARNING[6963][C-00000004]: app_voicemail.c:4086 insert_data_cb: SQL Direct Execute failed!
[Apr 4 18:12:14] WARNING[6963][C-00000004]: res_odbc.c:608 ast_odbc_direct_execute: SQL Execute error! Verifying connection to asterisk [asterisk-connector]...
[Apr 4 18:12:14] WARNING[6963][C-00000004]: app_voicemail.c:4086 insert_data_cb: SQL Direct Execute failed!
[Apr 4 18:12:14] WARNING[6963][C-00000004]: app_voicemail.c:4202 store_file: SQL Execute error!
[INSERT INTO voicemail (dir,msgnum,recording,context,macrocontext,callerid,origtime,duration,mailboxuser,mailboxcontext,flag,msg_id) VALUES (?,?,?,?,?,?,?,?,?,?,?,?)]
== Spawn extension (voice, 4759500, 4) exited non-zero on 'DAHDI/i1/9634434640-3'
-- Hungup 'DAHDI/i1/9634434640-3'
[Apr 4 18:12:14] WARNING[6963][C-00000004]: res_odbc.c:608 ast_odbc_direct_execute: SQL Execute error! Verifying connection to asterisk [asterisk-connector]...
any help would be appreciated
You have no odbc connection or no table.
For exact issue check
asterisk -rx "odbc show"
I am Novice in asterisk I installed Asterisk but now when I calling with telephony call cannot
come into asterisk &
I config ed Outgoing call bat call cannot out asterisk when I write(asterisk -vvvvvr)
& I calling with outdoor display for me
-- Executing [09396464991#DLPN_Main:1] Macro("SIP/6001-00000000", "trunkdial -failover-0.3,DAHDI/g2/09396464991,DAHDI/g1/09396464991,trunk_1,trunk_1") in newstack
-- Executing [s#macro-trunkdial-failover-0.3:1] GotoIf("SIP/6001-00000000","0?1-fmsetcid,1") in new stack
-- Executing [s#macro-trunkdial-failover-0.3:2] GotoIf("SIP/6001-00000000","0?1-setgbobname,1") in new stack
-- Executing [s#macro-trunkdial-failover-0.3:3] Set("SIP/6001-00000000", "CALLERID(num)=6001") in new stack
-- Executing [s#macro-trunkdial-failover-0.3:4] Set("SIP/6001-00000000", "CALLERID(all)=") in new stack
-- Executing [s#macro-trunkdial-failover-0.3:5] GotoIf("SIP/6001-00000000","0?1-dial,1") in new stack
-- Executing [s#macro-trunkdial-failover-0.3:6] Set("SIP/6001-00000000", "CALLERID(all)=") in new stack
-- Executing [s#macro-trunkdial-failover-0.3:7] Set("SIP/6001-00000000", "CALLERID(all)=") in new stack
-- Executing [s#macro-trunkdial-failover-0.3:8] Goto("SIP/6001-00000000", "1-dial,1") in new stack
-- Goto (macro-trunkdial-failover-0.3,1-dial,1)
-- Executing [1-dial#macro-trunkdial-failover-0.3:1] Dial("SIP/6001-00000000:", "DAHDI/g2/09396464991") in new stack
[Mar 10 13:40:04] **WARNING[2106]: channel.c:5627 ast_request: No channel type registered for 'DAHDI'**
[Mar 10 13:40:04] WARNING[2106]: app_dial.c:2274 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 66 - Channel not implemented)== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [1-dial#macro-trunkdial-failover-0.3:2] GotoIf("SIP/6001-00000000", "20 > 0?1-CHANUNAVAIL,1:1-out,1") in new stack
-- Goto (macro-trunkdial-failover-0.3,1-CHANUNAVAIL,1)
-- Executing [1-CHANUNAVAIL#macro-trunkdial-failover-0.3:1] Dial("SIP/6001-00000000", "DAHDI/g1/09396464991") in new stack [Mar 10 13:40:04] WARNING[2106]: channel.c:5627 ast_request: No channel type registered for 'DAHDI'
**[Mar 10 13:40:04] WARNING[2106]: app_dial.c:2274 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 66 - Channel not implemented)== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [1-CHANUNAVAIL#macro-trunkdial-failover-0.3:2] Hangup("SIP/6001-00000000", "") in new stack**== Spawn extension (macro-trunkdial-failover-0.3, 1-CHANUNAVAIL, 2) exited non-zero on 'SIP/6001-00000000' in macro 'trunkdial-failover-0.3'== Spawn extension (DLPN_Main, 09396464991, 1) exited non-zero on 'SIP/6001-00
1) First check your dahdi config is present and it is ok. Use following to check it
dahdi_cfg -vvvv
2) check that your asterisk have pri/dahdi support.