Issue with inbound call from PSTN - No Sounds, No live call, call drops. - GOautodial Open Source Omni-channel Contact Center Suite - GOautodial Open Source Omni-channel Contact Center Suite (predictive dialer + inbound IVR & ACD + non-voice)

Issue with inbound call from PSTN - No Sounds, No live call, call drops.

Added by Leopoldo Martinez almost 4 years ago

Hi, i finally could route inbound call to an agent, i was making all my test from my softphone to my goautodial, and all works fine (still working fine), but when i call from a real phone (PSTN Phone) the call drops, i can't hear any IVR, sound or nothing, just drop the call. please any help, this is really urgent.

Replies (9)

RE: Issue with inbound call from PSTN - No Sounds, No live call, call drops. - Added by Leopoldo Martinez almost 4 years ago

Hey, i could solve the problem above, was something with codecs, all fine now. But the life is not so easy and perfect, now I'm facing a new problem, the call from PSTN is ok, the can hear the IVR, but the can talk with an agent because the call drops inmediately, the error that i can see in asterisk cli is this one.

ERROR¹²⁰⁴⁴[C-000098fa]: member.c:404 member_exec: unable to create member

HERE IS THE COMPLETE TRACE OF ASTERISK CLI

Using SIP RTP CoS mark 5
Executing [910883259@trunkinbound:1] AGI in new stack
Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
<SIP/SOLIVESA-00003f17>AGI Script agi-DID_route.agi completed, returning 0
Executing [99909*78***DID@default:1] Answer("SIP/SOLIVESA-00003f17", "") in new stack
Executing [99909*78***DID@default:2] AGI in new stack
Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
<SIP/SOLIVESA-00003f17> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
<SIP/SOLIVESA-00003f17> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
Manager 'sendcron' logged on from 127.0.0.1
Called 127*000*000*001*78600089@default
Executing [127*000*000*001*78600089@default:1] Goto("Local/127*000*000*001*78600089@default-00005bc6;2", "default,78600089,1") in new stack
Goto (default,78600089,1)
Executing [78600089@default:1] Konference("Local/127*000*000*001*78600089@default-00005bc6;2", "8600089,qR") in new stack
Local/127*000*000*001*78600089@default-00005bc6;1 answered
Executing [83047777777777@vicidial-auto:1] Answer("Local/127*000*000*001*78600089@default-00005bc6;1", "") in new stack
Executing [83047777777777@vicidial-auto:2] Playback("Local/127*000*000*001*78600089@default-00005bc6;1", "ding") in new stack
<Local/127*000*000*001*78600089@default-00005bc6;1> Playing 'ding.gsm' (language 'en')
Manager 'sendcron' logged off from 127.0.0.1
Executing [83047777777777@vicidial-auto:3] Hangup("Local/127*000*000*001*78600089@default-00005bc6;1", "") in new stack
Spawn extension (vicidial-auto, 83047777777777, 3) exited non-zero on 'Local/127*000*000*001*78600089@default-00005bc6;1'
[Jun 4 20:56:45] WARNING⁹⁵²¹[C-000098eb]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
Executing [h@vicidial-auto:1] AGI") in new stack
<Local/127*000*000*001*78600089@default-00005bc6;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
[Jun 4 20:56:45] WARNING⁹⁵²²[C-000098ea]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
Executing [h@default:1] AGI") in new stack
<Local/127*000*000*001*78600089@default-00005bc6;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
<SIP/SOLIVESA-00003f17> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
<SIP/SOLIVESA-00003f17> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
<SIP/SOLIVESA-00003f17> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
<SIP/SOLIVESA-00003f17> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
<SIP/SOLIVESA-00003f17> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
<SIP/SOLIVESA-00003f17> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
<SIP/SOLIVESA-00003f17>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
Executing [127*000*000*001*8600089@default:1] Goto("SIP/SOLIVESA-00003f17", "default,8600089,1") in new stack
Goto (default,8600089,1)
Executing [8600089@default:1] Konference("SIP/SOLIVESA-00003f17", "8600089,R") in new stack
[Jun 4 20:56:46] ERROR⁹⁵¹⁴[C-000098e9]: member.c:404 member_exec: unable to create member
Spawn extension (default, 8600089, 1) exited non-zero on 'SIP/SOLIVESA-00003f17'
[Jun 4 20:56:46] WARNING⁹⁵¹⁴[C-000098e9]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
Executing [h@default:1] AGI") in new stack
<SIP/SOLIVESA-00003f17>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------) completed, returning 0
Manager 'sendcron' logged on from 127.0.0.1
Called 58600089@default
Executing [58600089@default:1] Konference("Local/58600089@default-00005bc7;2", "8600089,qLR") in new stack
Local/58600089@default-00005bc7;1 answered
Executing [8309@default:1] Answer("Local/58600089@default-00005bc7;1", "") in new stack
Executing [8309@default:2] Monitor("Local/58600089@default-00005bc7;1", "wav,20210604-155645_617899753_16514963_PILOT1") in new stack
Manager 'sendcron' logged off from 127.0.0.1
Executing [8309@default:3] Wait("Local/58600089@default-00005bc7;1", "3600") in new stack
Using SIP RTP CoS mark 5
Executing [910883259@trunkinbound:1] AGI in new stack
Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
<SIP/SOLIVESA-00003f18>AGI Script agi-DID_route.agi completed, returning 0
Executing [99909*78***DID@default:1] Answer("SIP/SOLIVESA-00003f18", "") in new stack
Executing [99909*78***DID@default:2] AGI in new stack
Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
<SIP/SOLIVESA-00003f18> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
<SIP/SOLIVESA-00003f18> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
<SIP/SOLIVESA-00003f18> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
<SIP/SOLIVESA-00003f18> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
<SIP/SOLIVESA-00003f18> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
<SIP/SOLIVESA-00003f18> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
<SIP/SOLIVESA-00003f18> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
<SIP/SOLIVESA-00003f18> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
<SIP/SOLIVESA-00003f18>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
Executing [8307@default:1] Answer("SIP/SOLIVESA-00003f18", "") in new stack
Executing [8307@default:2] Playback("SIP/SOLIVESA-00003f18", "vm-goodbye") in new stack
<SIP/SOLIVESA-00003f18> Playing 'vm-goodbye.gsm' (language 'en')
Manager 'sendcron' logged on from 127.0.0.1
Manager 'sendcron' from 127.0.0.1, hanging up channel: Local/58600089@default-00005bc7;2
[Jun 4 20:56:53] WARNING⁹⁵³⁴[C-000098ec]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
Executing [h@default:1] AGI") in new stack
Manager 'sendcron' logged off from 127.0.0.1
<Local/58600089@default-00005bc7;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------) completed, returning 0
Spawn extension (default, 8309, 3) exited non-zero on 'Local/58600089@default-00005bc7;1'
[Jun 4 20:56:53] WARNING⁹⁵³³[C-000098ed]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
Executing [h@default:1] AGI") in new stack
<Local/58600089@default-00005bc7;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------) completed, returning 0
[Jun 4 20:56:57] WARNING⁹⁵³⁵[C-000098ee]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
Executing [h@default:1] AGI") in new stack

Any one can help me? give me and advise or somethin, I would apreciate it.
Thanks.

RE: Issue with inbound call from PSTN - No Sounds, No live call, call drops. - Added by Wittie Manansala almost 4 years ago

Hi,

Please provide the following:

1. DID Settings screenshot
2. Ingroup Settings screenshot
3. Carrier Settings for your INBOUND
4. Asterisk CLI log when calling your inbound screenshot

Thanks

RE: Issue with inbound call from PSTN - No Sounds, No live call, call drops. - Added by Leopoldo Martinez almost 4 years ago

Wittie Manansala wrote in RE: Issue with inbound call from PSTN - No Sounds, No liv...:

Hi,

Please provide the following:

1. DID Settings screenshot

2. Ingroup Settings screenshot

3. Carrier Settings for your INBOUND

4. Asterisk CLI log when calling your inbound screenshot

Thanks

There is, all information you ask...

Download all files

clipboard-202106072010-gxh6m.png (25.6 KB) clipboard-202106072010-gxh6m.png
clipboard-202106072010-j0ina.png (21.6 KB) clipboard-202106072010-j0ina.png
clipboard-202106072011-9qqua.png (38.7 KB) clipboard-202106072011-9qqua.png
clipboard-202106072011-x6rlv.png (40.3 KB) clipboard-202106072011-x6rlv.png
clipboard-202106072014-tbcj6.png (49.6 KB) clipboard-202106072014-tbcj6.png
clipboard-202106072014-q8vos.png (217 KB) clipboard-202106072014-q8vos.png

RE: Issue with inbound call from PSTN - No Sounds, No live call, call drops. - Added by Demian Biscocho almost 4 years ago

This is a codec issue. Have you tried using G711/Ulaw?

<quote>
Executing [8600089@default:1] Konference("SIP/SOLIVESA-00003f17", "8600089,R") in new stack
[Jun 4 20:56:46] ERROR9514[C-000098e9]: member.c:404 member_exec: unable to create member
</quote>

RE: Issue with inbound call from PSTN - No Sounds, No live call, call drops. - Added by Leopoldo Martinez almost 4 years ago

Demian Lizandro Biscocho wrote in RE: Issue with inbound call from PSTN - No Sounds, No liv...:

This is a codec issue. Have you tried using G711/Ulaw?

<quote>
Executing [8600089@default:1] Konference("SIP/SOLIVESA-00003f17", "8600089,R") in new stack
[Jun 4 20:56:46] ERROR9514[C-000098e9]: member.c:404 member_exec: unable to create member
</quote>

Yes, as you see in one of the pics that our friend Wittie Manansala asked, you can see it

allow=ulaw

What other thing can i try?

RE: Issue with inbound call from PSTN - No Sounds, No live call, call drops. - Added by Leopoldo Martinez almost 4 years ago

Wittie Manansala wrote in RE: Issue with inbound call from PSTN - No Sounds, No liv...:

Hi,

Please provide the following:

1. DID Settings screenshot
2. Ingroup Settings screenshot
3. Carrier Settings for your INBOUND
4. Asterisk CLI log when calling your inbound screenshot

Thanks

HI, have you check it out?

RE: Issue with inbound call from PSTN - No Sounds, No live call, call drops. - Added by Leopoldo Martinez almost 4 years ago

Demian Lizandro Biscocho wrote in RE: Issue with inbound call from PSTN - No Sounds, No liv...:

This is a codec issue. Have you tried using G711/Ulaw?

<quote>
Executing [8600089@default:1] Konference("SIP/SOLIVESA-00003f17", "8600089,R") in new stack
[Jun 4 20:56:46] ERROR9514[C-000098e9]: member.c:404 member_exec: unable to create member
</quote>

Hi, I'm still facing the same error

I'm using
G711/Ulaw = Doesn't work
G711/Alaw = Doesn't work
G729 = Doesn't work
GSM = Doesn't work

please i need to solve this.

clipboard-202106162143-xnevz.png (226 KB) clipboard-202106162143-xnevz.png

RE: Issue with inbound call from PSTN - No Sounds, No live call, call drops. - Added by Leopoldo Martinez almost 4 years ago

Well, I have managed to get some calls in, but not others. I am going to leave samples of the call that I manage to enter against the other 2 that do not. I'm pretty sure one of them is due to the fact that the number I called from is in private mode, but the other, I have no idea why the call won't accept it.

here is the acepted call.

This one is a non acepted call.

And this is the anonymous or private number

I have installed g729, and i hace active GSM, ALAW, ULAW, G729. I triyed activating and deactivation some codec, the result is allways de same.

Download all files

clipboard-202106262146-nhue8.png (177 KB) clipboard-202106262146-nhue8.png
clipboard-202106262149-gva8c.png (195 KB) clipboard-202106262149-gva8c.png
clipboard-202106262150-yk3ok.png (200 KB) clipboard-202106262150-yk3ok.png

RE: Issue with inbound call from PSTN - No Sounds, No live call, call drops. - Added by Wittie Manansala almost 4 years ago

Hi,

Have you tried to coordinate the issue to your VOIP Provider? If Yes, what is there advise/recommendation?

Try also to route your inbound call in musiconhold or specific extension and check if calls goes through.

Thanks

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