No live call sound but appearing on recording

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No live call sound but appearing on recording

Postby vicieg » Mon Nov 15, 2021 12:40 pm

Hi all,

I've installed Vicidial and configured everything following the installation process as shown and it's now dialing however no voice heard while the call is running however when i record the call and i listen to the recording the other side sound shows up on the recording as it should be however not recording my voice and during the live call no sound at all

Vici version: VERSION: 2.14-833a
BUILD: 211106-1500

Carrier:
[********]
type=friend
host=************
Port=5060
insecure=port,invite
canreinvite=no
qualify=yes
allow=g729
allow=alaw
allow=ulaw
dtmfmode=rfc2833
context=trunkinbound
nat=force,comedia

Global string:
********* = SIP/*********

Dialplan:
exten => _+91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _+91NXXNXXXXXX,2,Dial(${***********}/${EXTEN:1},,tTor)
exten => _+91NXXNXXXXXX,3,Hangup

exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(${********}/+1${EXTEN:1},,tTor)
exten => _91XXXXXXXXXX,3,Hangup

exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(${*******}/+${EXTEN:1},,tTor)
exten => _91XXXXXXXXXX,3,Hangup
vicieg
 
Posts: 4
Joined: Fri Nov 05, 2021 10:08 am

Re: No live call sound but appearing on recording

Postby carpenox » Sat Nov 20, 2021 7:31 am

Do you hear the "only person in this conference" when you login?
Alma Linux 9.3 | Version: 2.14-911a | SVN Version: 3815 | DB Schema Version: 1710 | Asterisk 18.18.1
www.dialer.one -:- 1-833-DIALER-1 -:- https://linktr.ee/CyburDial -:- WhatsApp: +19549477572 -:- Skype: live:carpenox_3
carpenox
 
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Location: Coral Springs, FL

Re: No live call sound but appearing on recording

Postby vicieg » Sun Nov 21, 2021 10:23 am

carpenox wrote:Do you hear the "only person in this conference" when you login?


No sound at all.
I got back to the provider and they said to enable their ips on server's firewall. We have enabled their ips on ips list in the admin scection and disabled the firewall for trial. Yet issue still persisting.
vicieg
 
Posts: 4
Joined: Fri Nov 05, 2021 10:08 am

Re: No live call sound but appearing on recording

Postby carpenox » Sun Nov 21, 2021 4:58 pm

Type dahdi_cfg -v
Alma Linux 9.3 | Version: 2.14-911a | SVN Version: 3815 | DB Schema Version: 1710 | Asterisk 18.18.1
www.dialer.one -:- 1-833-DIALER-1 -:- https://linktr.ee/CyburDial -:- WhatsApp: +19549477572 -:- Skype: live:carpenox_3
carpenox
 
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Joined: Wed Apr 08, 2020 2:02 am
Location: Coral Springs, FL

Re: No live call sound but appearing on recording

Postby ambiorixg12 » Sun Nov 21, 2021 11:32 pm

With the firewall disabled, run this command rtp set debug on and make a test call, also show me the output of the sip show peers command
ambiorixg12
 
Posts: 448
Joined: Tue Sep 17, 2013 10:35 pm

Re: No live call sound but appearing on recording

Postby vicieg » Mon Nov 22, 2021 9:28 am

ambiorixg12 wrote:With the firewall disabled, run this command rtp set debug on and make a test call, also show me the output of the sip show peers command


This is what happened after running a test call


[Nov 22 15:40:19] WARNING[14133]: chan_sip.c:4092 retrans_pkt: Retransmission timeout reached on transmission 1809207821-470336134-1295739426 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 32000ms with no response
[Nov 22 15:40:20] NOTICE[14133][C-000161e5]: chan_sip.c:19559 send_check_user_failure_response: Failed to authenticate device <sip:108@208.73.203.12>;tag=1150507664 for INVITE, code = -1
[Nov 22 15:40:29] NOTICE[14133]: chan_sip.c:28907 handle_request_register: Registration from '"2002"<sip:2002@208.73.203.12>' failed for '62.4.15.143:9655' - Wrong password
[Nov 22 15:40:29] WARNING[14133]: chan_sip.c:4092 retrans_pkt: Retransmission timeout reached on transmission 10291571-1175619377-1520310029 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 31999ms with no response
[Nov 22 15:40:32] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 22 15:40:32] -- Called 8600051@default
[Nov 22 15:40:32] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000096;2", "8600051,F") in new stack
[Nov 22 15:40:32] -- Local/8600051@default-00000096;1 answered
[Nov 22 15:40:32] -- Executing [916504894546@default:1] AGI("Local/8600051@default-00000096;1", "agi://127.0.0.1:4577/call_log") in new stack
[Nov 22 15:40:32] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TESTOUT))
[Nov 22 15:40:32] -- <Local/8600051@default-00000096;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov 22 15:40:32] -- Executing [916504894546@default:2] Dial("Local/8600051@default-00000096;1", "SIP/TwilioVici/+116504894546,,tTor") in new stack
[Nov 22 15:40:32] == Using SIP RTP CoS mark 5
[Nov 22 15:40:32] -- Called SIP/TwilioVici/+116504894546
[Nov 22 15:40:32] -- Created MeetMe conference 1023 for conference '8600051'
[Nov 22 15:40:32] -- <Local/8600051@default-00000096;2> Playing 'conf-onlyperson.gsm' (language 'en')
[Nov 22 15:40:32] -- SIP/TwilioVici-000000a2 is ringing
[Nov 22 15:40:33] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 22 15:40:37] > 0x7fac24051bf0 -- Strict RTP learning after remote address set to: 34.203.251.73:18668
[Nov 22 15:40:37] -- SIP/TwilioVici-000000a2 answered Local/8600051@default-00000096;1
[Nov 22 15:40:37] -- Channel SIP/TwilioVici-000000a2 joined 'simple_bridge' basic-bridge <a8048fa2-6cca-4e44-beb0-55da3a637ca3>
[Nov 22 15:40:37] -- Channel Local/8600051@default-00000096;1 joined 'simple_bridge' basic-bridge <a8048fa2-6cca-4e44-beb0-55da3a637ca3>
[Nov 22 15:40:37] > 0x7fac24051bf0 -- Strict RTP switching to RTP target address 34.203.251.73:18668 as source
[Nov 22 15:40:42] > 0x7fac24051bf0 -- Strict RTP learning complete - Locking on source address 34.203.251.73:18668
[Nov 22 15:40:52] WARNING[14133]: chan_sip.c:4092 retrans_pkt: Retransmission timeout reached on transmission 2038966071-281624881-274530353 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 32000ms with no response
[Nov 22 15:40:52] NOTICE[14133][C-000161e8]: chan_sip.c:19559 send_check_user_failure_response: Failed to authenticate device <sip:108@208.73.203.12>;tag=1331978843 for INVITE, code = -1
[Nov 22 15:41:01] NOTICE[14133]: chan_sip.c:28907 handle_request_register: Registration from '<sip:1278@208.73.203.12>' failed for '20.115.25.73:61481' - Wrong password
[Nov 22 15:41:02] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 22 15:41:02] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 22 15:41:02] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 22 15:41:02] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 22 15:41:07] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 22 15:41:07] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 22 15:41:24] WARNING[14133]: chan_sip.c:4092 retrans_pkt: Retransmission timeout reached on transmission 90054547-247896282-262465640 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 32000ms with no response
[Nov 22 15:41:30] WARNING[14133]: chan_sip.c:4166 retrans_pkt: Timeout on 1897702625-1189523988-1507116835 on non-critical invite transaction.
vicieg
 
Posts: 4
Joined: Fri Nov 05, 2021 10:08 am

Re: No live call sound but appearing on recording

Postby vicieg » Mon Nov 22, 2021 10:46 am

vicieg wrote:
ambiorixg12 wrote:With the firewall disabled, run this command rtp set debug on and make a test call, also show me the output of the sip show peers command


This is what happened after running a test call


[Nov 22 15:40:19] WARNING[14133]: chan_sip.c:4092 retrans_pkt: Retransmission timeout reached on transmission 1809207821-470336134-1295739426 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 32000ms with no response
[Nov 22 15:40:20] NOTICE[14133][C-000161e5]: chan_sip.c:19559 send_check_user_failure_response: Failed to authenticate device <sip:108@208.73.203.12>;tag=1150507664 for INVITE, code = -1
[Nov 22 15:40:29] NOTICE[14133]: chan_sip.c:28907 handle_request_register: Registration from '"2002"<sip:2002@208.73.203.12>' failed for '62.4.15.143:9655' - Wrong password
[Nov 22 15:40:29] WARNING[14133]: chan_sip.c:4092 retrans_pkt: Retransmission timeout reached on transmission 10291571-1175619377-1520310029 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 31999ms with no response
[Nov 22 15:40:32] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 22 15:40:32] -- Called 8600051@default
[Nov 22 15:40:32] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000096;2", "8600051,F") in new stack
[Nov 22 15:40:32] -- Local/8600051@default-00000096;1 answered
[Nov 22 15:40:32] -- Executing [916504894546@default:1] AGI("Local/8600051@default-00000096;1", "agi://127.0.0.1:4577/call_log") in new stack
[Nov 22 15:40:32] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TESTOUT))
[Nov 22 15:40:32] -- <Local/8600051@default-00000096;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov 22 15:40:32] -- Executing [916504894546@default:2] Dial("Local/8600051@default-00000096;1", "SIP/TwilioVici/+116504894546,,tTor") in new stack
[Nov 22 15:40:32] == Using SIP RTP CoS mark 5
[Nov 22 15:40:32] -- Called SIP/TwilioVici/+116504894546
[Nov 22 15:40:32] -- Created MeetMe conference 1023 for conference '8600051'
[Nov 22 15:40:32] -- <Local/8600051@default-00000096;2> Playing 'conf-onlyperson.gsm' (language 'en')
[Nov 22 15:40:32] -- SIP/TwilioVici-000000a2 is ringing
[Nov 22 15:40:33] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 22 15:40:37] > 0x7fac24051bf0 -- Strict RTP learning after remote address set to: 34.203.251.73:18668
[Nov 22 15:40:37] -- SIP/TwilioVici-000000a2 answered Local/8600051@default-00000096;1
[Nov 22 15:40:37] -- Channel SIP/TwilioVici-000000a2 joined 'simple_bridge' basic-bridge <a8048fa2-6cca-4e44-beb0-55da3a637ca3>
[Nov 22 15:40:37] -- Channel Local/8600051@default-00000096;1 joined 'simple_bridge' basic-bridge <a8048fa2-6cca-4e44-beb0-55da3a637ca3>
[Nov 22 15:40:37] > 0x7fac24051bf0 -- Strict RTP switching to RTP target address 34.203.251.73:18668 as source
[Nov 22 15:40:42] > 0x7fac24051bf0 -- Strict RTP learning complete - Locking on source address 34.203.251.73:18668
[Nov 22 15:40:52] WARNING[14133]: chan_sip.c:4092 retrans_pkt: Retransmission timeout reached on transmission 2038966071-281624881-274530353 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 32000ms with no response
[Nov 22 15:40:52] NOTICE[14133][C-000161e8]: chan_sip.c:19559 send_check_user_failure_response: Failed to authenticate device <sip:108@208.73.203.12>;tag=1331978843 for INVITE, code = -1
[Nov 22 15:41:01] NOTICE[14133]: chan_sip.c:28907 handle_request_register: Registration from '<sip:1278@208.73.203.12>' failed for '20.115.25.73:61481' - Wrong password
[Nov 22 15:41:02] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 22 15:41:02] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 22 15:41:02] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 22 15:41:02] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 22 15:41:07] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 22 15:41:07] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 22 15:41:24] WARNING[14133]: chan_sip.c:4092 retrans_pkt: Retransmission timeout reached on transmission 90054547-247896282-262465640 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 32000ms with no response
[Nov 22 15:41:30] WARNING[14133]: chan_sip.c:4166 retrans_pkt: Timeout on 1897702625-1189523988-1507116835 on non-critical invite transaction.


after running rtp set debug on
[Nov 22 17:44:45] NOTICE[14133]: chan_sip.c:28907 handle_request_register: Registration from '"1900" <sip:1900@208.73.203.12>' failed for '173.212.243.187:5159' - Wrong password
[Nov 22 17:44:45] NOTICE[14133]: chan_sip.c:28907 handle_request_register: Registration from '"1900" <sip:1900@208.73.203.12>' failed for '173.212.243.187:5159' - Wrong password
[Nov 22 17:44:45] NOTICE[14133]: chan_sip.c:28907 handle_request_register: Registration from '"1900" <sip:1900@208.73.203.12>' failed for '173.212.243.187:5159' - Wrong password
[Nov 22 17:44:45] NOTICE[14133]: chan_sip.c:28907 handle_request_register: Registration from '"1900" <sip:1900@208.73.203.12>' failed for '173.212.243.187:5159' - Wrong password
[Nov 22 17:44:45] NOTICE[14133]: chan_sip.c:28907 handle_request_register: Registration from '"1900" <sip:1900@208.73.203.12>' failed for '173.212.243.187:5159' - Wrong password
[Nov 22 17:44:45] NOTICE[14133]: chan_sip.c:28907 handle_request_register: Registration from '"1900" <sip:1900@208.73.203.12>' failed for '173.212.243.187:5159' - Wrong password
vicieg
 
Posts: 4
Joined: Fri Nov 05, 2021 10:08 am

Re: No live call sound but appearing on recording

Postby ambiorixg12 » Mon Nov 22, 2021 1:28 pm

[Nov 22 15:41:24] WARNING[14133]: chan_sip.c:4092 retrans_pkt: Retransmission timeout reached on transmission 90054547-247896282-262465640 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 32000ms with no response


Retransmission timeout warning is related to NAT issue follow the link on the warning that will give you some hints to fix the issue
ambiorixg12
 
Posts: 448
Joined: Tue Sep 17, 2013 10:35 pm


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