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


(Hunterman) #1

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 [s@crm-hangup: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 [s@crm-hangup:2] NoOp(“SIP/6000-0000006b”, “HANGUP CAUSE: 16”) in new stack
– Executing [s@crm-hangup: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 [h@ext-local:1] Macro(“PJSIP/anonymous-00000044”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/anonymous-00000044”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@crm-hangup:4] NoOp(“SIP/6000-0000006b”, “MASTER CHANNEL: 1550123825.209 = 1550123825.208”) in new stack
– Executing [s@crm-hangup:5] GotoIf(“SIP/6000-0000006b”, “1?return”) in new stack
– Goto (crm-hangup,s,8)
– Executing [s@crm-hangup: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 [s@macro-hangupcall:3] ExecIf(“PJSIP/anonymous-00000044”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“PJSIP/anonymous-00000044”, "SIP/6000-0000006b monior file= ") in new stack
– Executing [s@macro-hangupcall: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 [s@macro-hangupcall: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 [s@crm-hangup:1] NoOp(“PJSIP/anonymous-00000044”, “Sending Hangup to CRM”) in new stack
– Executing [s@crm-hangup:2] NoOp(“PJSIP/anonymous-00000044”, “HANGUP CAUSE: 16”) in new stack
– Executing [s@crm-hangup:3] ExecIf(“PJSIP/anonymous-00000044”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
– Executing [s@crm-hangup:4] NoOp(“PJSIP/anonymous-00000044”, “MASTER CHANNEL: 1550123825.208 = 1550123825.208”) in new stack
– Executing [s@crm-hangup:5] GotoIf(“PJSIP/anonymous-00000044”, “0?return”) in new stack
– Executing [s@crm-hangup:6] Set(“PJSIP/anonymous-00000044”, “__CRM_HANGUP=1”) in new stack
– Executing [s@crm-hangup: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 [s@crm-hangup: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


#2

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)


(Hunterman) #3

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.


#4

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


(Hunterman) #5

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


(system) closed #6

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