Live call without any sound

i have install goautodial 4.0 on server with iso and all set , but a lot of call are silent without any sound

all codec are selected in carrier , any help

Replies (6)

RE: Live call without any sound - Added by Levy Ryan Nolasco over 4 years ago

When your agent login do they hear the voice prompt "You are the only person in this conference?", Can you post your asterisk logs and sip show channels while attempting a call.

RE: Live call without any sound - Added by ahmed elgendy over 4 years ago

yes , agent hear the voice prompt


    -- Executing [55558600053@default:1] Hangup("Local/55558600053@default-00000052;2", "") in new stack
  == Spawn extension (default, 55558600053, 1) exited non-zero on 'Local/55558600053@default-00000052;2'
[Sep 23 05:42:49] WARNING[18331][C-0000008d]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
    -- Executing [h@default:1] AGI("Local/55558600053@default-00000052;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
    -- <Local/55558600053@default-00000052;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Called 99998633272530@default
    -- Executing [99998633272530@default:1] Dial("Local/99998633272530@default-00000053;2", "SIP/8633272530@kamailio,,tTo") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/8633272530@kamailio
    -- SIP/kamailio-00000048 is ringing
    -- Local/99998633272530@default-00000053;1 is ringing
       > 0x7f0ee4007ab0 -- Strict RTP learning after remote address set to: 192.168.22.237:30452
    -- SIP/kamailio-00000048 answered Local/99998633272530@default-00000053;2
    -- Local/99998633272530@default-00000053;1 answered
    -- Executing [8600051@default:1] Konference("Local/99998633272530@default-00000053;1", "8600051,R") in new stack
  == Manager 'sendcron' logged off from 127.0.0.1
    -- Channel SIP/kamailio-00000048 joined 'simple_bridge' basic-bridge <1f5a58c6-f20c-4dc0-a265-41c02e84343e>
    -- Channel Local/99998633272530@default-00000053;2 joined 'simple_bridge' basic-bridge <1f5a58c6-f20c-4dc0-a265-41c02e84343e>
       > 0x7f0ee4007ab0 -- Strict RTP switching to RTP target address 192.168.22.237:30452 as source
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
       > 0x7f0ee4007ab0 -- Strict RTP learning complete - Locking on source address 192.168.22.237:30452
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Called 8600051@default
    -- Executing [8600051@default:1] Konference("Local/8600051@default-00000054;2", "8600051,R") in new stack
    -- Local/8600051@default-00000054;1 answered
    -- Executing [871238876519076876985@default:1] AGI("Local/8600051@default-00000054;1", "agi://127.0.0.1:4577/call_log") in new stack
  == Manager 'sendcron' logged off from 127.0.0.1
    -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=95594039))
    -- <Local/8600051@default-00000054;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing [871238876519076876985@default:2] Dial("Local/8600051@default-00000054;1", "SIP/19076876985@commpeak1,,tTo") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/19076876985@commpeak1
[Sep 23 05:43:08] WARNING[18393][C-00000090]: file.c:774 ast_openstream_full: File confbridge-join does not exist in any format
[Sep 23 05:43:08] WARNING[18380][C-0000008f]: file.c:774 ast_openstream_full: File confbridge-join does not exist in any format
[Sep 23 05:43:08] WARNING[18380][C-0000008f]: file.c:774 ast_openstream_full: File confbridge-join does not exist in any format
       > 0x7f0f0018a680 -- Strict RTP learning after remote address set to: 104.196.22.135:26760
    -- SIP/commpeak1-00000049 is making progress passing it to Local/8600051@default-00000054;1
       > 0x7f0f0018a680 -- Strict RTP switching to RTP target address 104.196.22.135:26760 as source
       > 0x7f0f0018a680 -- Strict RTP learning complete - Locking on source address 104.196.22.135:26760
    -- SIP/commpeak1-00000049 answered Local/8600051@default-00000054;1
    -- Channel SIP/commpeak1-00000049 joined 'simple_bridge' basic-bridge <ac8b93dd-ff4d-4d53-b5c8-62a128a185b7>
    -- Channel Local/8600051@default-00000054;1 joined 'simple_bridge' basic-bridge <ac8b93dd-ff4d-4d53-b5c8-62a128a185b7>
    -- Channel SIP/commpeak1-00000049 left 'simple_bridge' basic-bridge <ac8b93dd-ff4d-4d53-b5c8-62a128a185b7>
    -- Channel Local/8600051@default-00000054;1 left 'simple_bridge' basic-bridge <ac8b93dd-ff4d-4d53-b5c8-62a128a185b7>
  == Spawn extension (default, 871238876519076876985, 2) exited non-zero on 'Local/8600051@default-00000054;1'
    -- Executing [h@default:1] AGI("Local/8600051@default-00000054;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----16-----SIP 200 OK)") in new stack
    -- <Local/8600051@default-00000054;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----16-----SIP 200 OK) completed, returning 0
[Sep 23 05:43:24] WARNING[18393][C-00000090]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
    -- Executing [h@default:1] AGI("Local/8600051@default-00000054;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
    -- <Local/8600051@default-00000054;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
[Sep 23 05:43:24] WARNING[18380][C-0000008f]: file.c:774 ast_openstream_full: File confbridge-leave does not exist in any format
  == Manager 'sendcron' logged on from 127.0.0.1
[Sep 23 05:43:24] NOTICE[18415]: manager.c:4309 action_hangup: Request to hangup non-existent channel: SIP/commpeak1-00000049
  == Manager 'sendcron' logged off from 127.0.0.1
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RE: Live call without any sound - Added by Ratanraj Singh over 4 years ago

Hi All,
I am also troubling with this issue randomly.As one call is dial fine with sound and tone and when we dial second number then no sound come,however call is connected but issue is with sound.
and when we logout and re-login then its work for 2 or 3 calls and again the the sound issue is come.

Can you please help on this how to fix the sound issue,
Note: We are not using any soft phone.

RE: Live call without any sound - Added by Wittie Manansala over 4 years ago

Hi,

Try to enable ulaw codec only in your carrier settings or coordinate to your VOIP provider to know what are the required codec/s.

Thanks

RE: Live call without any sound - Added by Ratanraj Singh over 4 years ago

Ulaw is already enabled in my career setting. PFB

[ola2020]
disallow=all
allow=gsm
allow=ulaw
allow=alaw
type=peer
dtmfmode=info
context=trunkinbound
qualify=yes
insecure=very
nat=yes
host=10.164.1.63

Protocol =SIP

Dial Plan

exten => _7259609482.,1,AGI
exten => _7259609482.,2,Dial(SIP/${EXTEN:10}@ola2020,,tTo)
exten => _7259609482.,3,Hangup

However I am using GSM gateway, Please be advise on this whether the GSM gateway is compatible with goautodial 4 or not.

careeer setting.JPG (69.7 KB) careeer setting.JPG

RE: Live call without any sound - Added by Wittie Manansala about 4 years ago

Hi,

Please try this:

[ola2020]
disallow=all
allow=ulaw
type=peer
dtmfmode=info
context=trunkinbound
qualify=yes
insecure=very
nat=yes
host=10.164.1.63

Thanks

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