problem with incoming calls

[asiawatcher]

so i have set up an INBOUND campaign users dids and it appears in the carriers section that my SIP account is registered

however when i call in to the box it isnt roputing the call to thew inbound group nor i hear the music on hold i set up

here is the log

any help would be greatly appreciated



cheers


- ViciBox Redux v.5.0.2-130821
linux-fsjm:~ # asterisk -vvvvvr
Asterisk 1.8.23.0-vici, Copyright (C) 1999 - 2012 Digium, Inc. and others.

Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for detail s.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.8.23.0-vici currently running on linux-fsjm (pid = 2633)
Verbosity is at least 21
[May 16 13:35:36] == Using SIP RTP CoS mark 5
[May 16 13:35:36] -- Executing [302118007994@trunkinbound:1] AGI("SIP/VIVA-0 0000016", "agi-DID_route.agi") in new stack
[May 16 13:35:36] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID _route.agi
[May 16 13:35:36] -- <SIP/VIVA-00000016>AGI Script agi-DID_route.agi complet ed, returning 0
[May 16 13:35:36] -- Executing [302118007994@trunkinbound:2] Hangup("SIP/VIV A-00000016", "") in new stack
[May 16 13:35:36] == Spawn extension (trunkinbound, 302118007994, 2) exited no n-zero on 'SIP/VIVA-00000016'
[May 16 13:35:36] -- Executing [h@trunkinbound:1] AGI("SIP/VIVA-00000016", " agi://127.0.0.1:4577/call_log--HVcauses ... ----------") in new stack
[May 16 13:35:36] -- <SIP/VIVA-00000016>AGI Script agi://127.0.0.1:4577/call _log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
linux-fsjm*CLI>

[striker]

1. have u created a DID with id 30211800799
2. what u have set the DID route = ?

[asiawatcher]

problem solved thanks a lot

i also got another problem while i uploaded files for music on hold i call and i hear silence why ? files are there i seen them with winscp and it says it cant find files


- ViciBox Redux v.5.0.2-130821

Verbosity is at least 21
[May 21 12:22:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 21 12:22:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 21 12:22:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 21 12:22:02] == Manager 'sendcron' logged off from 127.0.0.1
[May 21 12:22:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 21 12:22:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 21 12:22:08] > Refreshing DNS lookups.
[May 21 12:22:08] > ast_get_srv: SRV lookup for '_sip._udp.viva.gr' mapped to host telephony.viva.gr, port 5060
[May 21 12:22:08] == Using SIP RTP CoS mark 5
[May 21 12:22:08] -- Executing [302118007994@trunkinbound:1] AGI("SIP/VIVA-00000001", "agi-DID_route.agi") in new stack
[May 21 12:22:08] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[May 21 12:22:08] -- <SIP/VIVA-00000001>AGI Script agi-DID_route.agi completed, returning 0
[May 21 12:22:08] -- Executing [99909*3***DID@default:1] Answer("SIP/VIVA-00000001", "") in new stack
[May 21 12:22:09] -- Executing [99909*3***DID@default:2] AGI("SIP/VIVA-00000001", "agi-VDAD_ALL_inbound.agi") in new stack
[May 21 12:22:09] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[May 21 12:22:09] WARNING[13365]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:22:09] WARNING[13365]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:22:09] WARNING[13365]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin)
[May 21 12:22:11] -- Started music on hold, class 'default', on SIP/VIVA-00000001
[May 21 12:22:11] WARNING[13365]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin)
[May 21 12:22:11] WARNING[13365]: res_musiconhold.c:329 ast_moh_files_next: Unable to open file '/var/lib/asterisk/default/0001_test_stereo_48000Hz_8bit_PCM': No such file or directory
[May 21 12:22:11] WARNING[13365]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:22:11] WARNING[13365]: res_musiconhold.c:329 ast_moh_files_next: Unable to open file '/var/lib/asterisk/default/0000_sip-silence': No such file or directory
[May 21 12:22:11] -- Stopped music on hold on SIP/VIVA-00000001
[May 21 12:22:14] WARNING[13365]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:22:14] WARNING[13365]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:22:14] WARNING[13365]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:22:14] WARNING[13365]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin)
[May 21 12:22:15] WARNING[13365]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:22:15] WARNING[13365]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:22:15] WARNING[13365]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:22:16] -- Started music on hold, class 'default', on SIP/VIVA-00000001
[May 21 12:22:16] WARNING[13365]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin)
[May 21 12:22:16] WARNING[13365]: res_musiconhold.c:329 ast_moh_files_next: Unable to open file '/var/lib/asterisk/default/0001_test_stereo_48000Hz_8bit_PCM': No such file or directory
[May 21 12:22:16] WARNING[13365]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:22:16] WARNING[13365]: res_musiconhold.c:329 ast_moh_files_next: Unable to open file '/var/lib/asterisk/default/0000_sip-silence': No such file or directory
[May 21 12:22:16] -- Stopped music on hold on SIP/VIVA-00000001
[May 21 12:22:32] -- <SIP/VIVA-00000001>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 4
[May 21 12:22:32] == Spawn extension (default, 99909*3***DID, 2) exited non-zero on 'SIP/VIVA-00000001'
[May 21 12:22:32] -- Executing [h@default:1] AGI("SIP/VIVA-00000001", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[May 21 12:22:32] -- <SIP/VIVA-00000001>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[May 21 12:23:02] == Manager 'sendcron' logged on from 127.0.0.1
[May 21 12:23:02] == Manager 'sendcron' logged off from 127.0.0.1
[May 21 12:23:02] == Manager 'sendcron' logged on from 127.0.0.1
[May 21 12:23:02] == Manager 'sendcron' logged off from 127.0.0.1
[May 21 12:23:07] == Manager 'sendcron' logged on from 127.0.0.1
[May 21 12:23:07] == Manager 'sendcron' logged off from 127.0.0.1
[May 21 12:24:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 21 12:24:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 21 12:24:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 21 12:24:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 21 12:24:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 21 12:24:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 21 12:24:56] == Using SIP RTP CoS mark 5
[May 21 12:24:56] -- Executing [302118007994@trunkinbound:1] AGI("SIP/VIVA-00000002", "agi-DID_route.agi") in new stack
[May 21 12:24:56] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[May 21 12:24:56] -- <SIP/VIVA-00000002>AGI Script agi-DID_route.agi completed, returning 0
[May 21 12:24:56] -- Executing [99909*3***DID@default:1] Answer("SIP/VIVA-00000002", "") in new stack
[May 21 12:24:56] -- Executing [99909*3***DID@default:2] AGI("SIP/VIVA-00000002", "agi-VDAD_ALL_inbound.agi") in new stack
[May 21 12:24:56] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[May 21 12:24:56] WARNING[13681]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:24:56] WARNING[13681]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:24:56] WARNING[13681]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin)
[May 21 12:24:58] -- Started music on hold, class 'default', on SIP/VIVA-00000002
[May 21 12:24:58] WARNING[13681]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:24:58] WARNING[13681]: res_musiconhold.c:329 ast_moh_files_next: Unable to open file '/var/lib/asterisk/default/0001_test_stereo_48000Hz_8bit_PCM': No such file or directory
[May 21 12:24:58] WARNING[13681]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:24:58] WARNING[13681]: res_musiconhold.c:329 ast_moh_files_next: Unable to open file '/var/lib/asterisk/default/0000_sip-silence': No such file or directory
[May 21 12:24:58] -- Stopped music on hold on SIP/VIVA-00000002
[May 21 12:25:01] WARNING[13681]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:25:01] WARNING[13681]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:25:01] WARNING[13681]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:25:01] WARNING[13681]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin)
[May 21 12:25:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 21 12:25:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 21 12:25:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 21 12:25:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 21 12:25:02] WARNING[13681]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:25:02] WARNING[13681]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:25:02] WARNING[13681]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:25:03] -- Started music on hold, class 'default', on SIP/VIVA-00000002
[May 21 12:25:03] WARNING[13681]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:25:03] WARNING[13681]: res_musiconhold.c:329 ast_moh_files_next: Unable to open file '/var/lib/asterisk/default/0001_test_stereo_48000Hz_8bit_PCM': No such file or directory
[May 21 12:25:03] WARNING[13681]: channel.c:5211 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[May 21 12:25:03] WARNING[13681]: res_musiconhold.c:329 ast_moh_files_next: Unable to open file '/var/lib/asterisk/default/0000_sip-silence': No such file or directory
[May 21 12:25:03] -- Stopped music on hold on SIP/VIVA-00000002
[May 21 12:25:05] -- <SIP/VIVA-00000002>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 4
[May 21 12:25:05] == Spawn extension (default, 99909*3***DID, 2) exited non-zero on 'SIP/VIVA-00000002'
[May 21 12:25:05] -- Executing [h@default:1] AGI("SIP/VIVA-00000002", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[May 21 12:25:05] -- <SIP/VIVA-00000002>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[May 21 12:25:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 21 12:25:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 21 12:26:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 21 12:26:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 21 12:26:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 21 12:26:02] == Manager 'sendcron' logged off from 127.0.0.1
[May 21 12:26:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 21 12:26:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 21 12:27:02] == Manager 'sendcron' logged on from 127.0.0.1
[May 21 12:27:02] == Manager 'sendcron' logged on from 127.0.0.1
[May 21 12:27:02] == Manager 'sendcron' logged off from 127.0.0.1
[May 21 12:27:02] == Manager 'sendcron' logged off from 127.0.0.1
[May 21 12:27:07] == Manager 'sendcron' logged on from 127.0.0.1
[May 21 12:27:07] == Manager 'sendcron' logged off from 127.0.0.1
[May 21 12:27:08] > Refreshing DNS lookups.
[May 21 12:27:08] > ast_get_srv: SRV lookup for '_sip._udp.viva.gr' mapped to host telephony.viva.gr, port 5060
linux-fsjm*CLI>

[geoff3dmg]

Your carrier is sending you the audio for the inbound call as g729. However you do not have that codec enabled. So asterisk is unable to convert the g729 to gsm. Either enable g729 as an allowed codec or ask your carrier to change the audio format of inbound calls to gsm.

[boybawang]

you need to install g729 on your server