After call from GSM to Asterisk server to direct extension have problem

Hello everyone

Am to test to call from gsm mobile to Asterisk server to call direct extension like 6000,
I have ringing between GSM and Astersik but the problem when Am answer from extension the call go to be disconnect automatic why?

when disconnect the ssh show me:

0x7f638891f220 – Strict RTP learning after remote address set to: 192.168.29.5:5008
[2019-02-14 08:57:15] WARNING[3555][C-00000088]: channel.c:5600 set_format: Unable to find a codec translation path: (ulaw) -> (g723)
[2019-02-14 08:57:15] WARNING[3555][C-00000088]: channel.c:5600 set_format: Unable to find a codec translation path: (g723) -> (ulaw)
– Connected line update to PJSIP/anonymous-00000044 prevented.
– SIP/6000-0000006b answered PJSIP/anonymous-00000044
0x7f6400915bb0 – Strict RTP learning after remote address set to: 192.168.33.169:8012
[2019-02-14 08:57:15] WARNING[12472]: channel.c:5600 set_format: Unable to find a codec translation path: (g723) -> (alaw)
[2019-02-14 08:57:15] WARNING[12472]: channel.c:5600 set_format: Unable to find a codec translation path: (ulaw) -> (g723)
– Channel SIP/6000-0000006b joined ‘simple_bridge’ basic-bridge
– Channel PJSIP/anonymous-00000044 joined ‘simple_bridge’ basic-bridge
[2019-02-14 08:57:15] WARNING[12478][C-00000088]: channel.c:6560 ast_channel_make_compatible_helper: No path to translate from PJSIP/anonymous-00000044 to SIP/6000-0000006b
– Channel PJSIP/anonymous-00000044 left ‘simple_bridge’ basic-bridge
– Channel SIP/6000-0000006b left ‘simple_bridge’ basic-bridge
== Spawn extension (macro-dial-one, s, 55) exited non-zero on ‘PJSIP/anonymous-00000044’ in macro ‘dial-one’
– SIP/6000-0000006b Internal Gosub(crm-hangup,s,1) start
– Executing [[email protected]:1] NoOp(“SIP/6000-0000006b”, “Sending Hangup to CRM”) in new stack
== Spawn extension (macro-exten-vm, s, 19) exited non-zero on ‘PJSIP/anonymous-00000044’ in macro ‘exten-vm’
– Executing [[email protected]:2] NoOp(“SIP/6000-0000006b”, “HANGUP CAUSE: 16”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/6000-0000006b”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
== Spawn extension (ext-local, 6000, 2) exited non-zero on ‘PJSIP/anonymous-00000044’
– Executing [[email protected]:1] Macro(“PJSIP/anonymous-00000044”, “hangupcall,”) in new stack
– Executing [[email protected]:1] GotoIf(“PJSIP/anonymous-00000044”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [[email protected]:4] NoOp(“SIP/6000-0000006b”, “MASTER CHANNEL: 1550123825.209 = 1550123825.208”) in new stack
– Executing [[email protected]:5] GotoIf(“SIP/6000-0000006b”, “1?return”) in new stack
– Goto (crm-hangup,s,8)
– Executing [[email protected]:8] Return(“SIP/6000-0000006b”, “”) in new stack
== Spawn extension (from-internal, , 1) exited non-zero on ‘SIP/6000-0000006b’
– SIP/6000-0000006b Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
– Executing [[email protected]:3] ExecIf(“PJSIP/anonymous-00000044”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [[email protected]:4] NoOp(“PJSIP/anonymous-00000044”, "SIP/6000-0000006b monior file= ") in new stack
– Executing [[email protected]:5] AGI(“PJSIP/anonymous-00000044”, “attendedtransfer-rec-restart.php,SIP/6000-0000006b,”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
– <PJSIP/anonymous-00000044>AGI Script attendedtransfer-rec-restart.php completed, returning 0
– Executing [[email protected]:6] Hangup(“PJSIP/anonymous-00000044”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘PJSIP/anonymous-00000044’ in macro ‘hangupcall’
== Spawn extension (ext-local, h, 1) exited non-zero on ‘PJSIP/anonymous-00000044’
– PJSIP/anonymous-00000044 Internal Gosub(crm-hangup,s,1) start
– Executing [[email protected]:1] NoOp(“PJSIP/anonymous-00000044”, “Sending Hangup to CRM”) in new stack
– Executing [[email protected]:2] NoOp(“PJSIP/anonymous-00000044”, “HANGUP CAUSE: 16”) in new stack
– Executing [[email protected]:3] ExecIf(“PJSIP/anonymous-00000044”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
– Executing [[email protected]:4] NoOp(“PJSIP/anonymous-00000044”, “MASTER CHANNEL: 1550123825.208 = 1550123825.208”) in new stack
– Executing [[email protected]:5] GotoIf(“PJSIP/anonymous-00000044”, “0?return”) in new stack
– Executing [[email protected]:6] Set(“PJSIP/anonymous-00000044”, “__CRM_HANGUP=1”) in new stack
– Executing [[email protected]:7] AGI(“PJSIP/anonymous-00000044”, “sangomacrm.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
– <PJSIP/anonymous-00000044>AGI Script sangomacrm.agi completed, returning 0
– Executing [[email protected]:8] Return(“PJSIP/anonymous-00000044”, “”) in new stack
== Spawn extension (ext-local, h, 1) exited non-zero on ‘PJSIP/anonymous-00000044’
– PJSIP/anonymous-00000044 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=

note:
Unable to find a codec translation path: (ulaw) -> (g723),
in my astersik server Am enable the codec (ulaw) -> (g723),but my ip phone is Grandstream GXP2160

any help to solve it?
THANKS

Your error is a codec mismatch and it seems your Asterisk server can’t transcode for some reason. You need to use the same codec on both ends.

2019-02-14 08:57:15] WARNING[12472]: channel.c:5600 set_format: Unable to find a codec translation path: (g723) -> (alaw)
[2019-02-14 08:57:15] WARNING[12472]: channel.c:5600 set_format: Unable to find a codec translation path: (ulaw) -> (g723)

I know that my friend,
But in my gsm gateway device not include codec “ulaw and alaw”, in astersik server selected all codec, not working…
Gsm device worked with codec “G.723.1,PCMA,G.729AB,G.723.1,PCMU”
I will try it to solve the problem for another method.

PCMU=ulaw
PCMA=alaw
Use one of those on both sides

Thank you for your replay and your information
Iam go to test very soon

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