Basic-bridge exited non-zero

[comfine]

Hello,

as we can see in this part of the log, there was an unwanted ending of the call directly after call was established. the caller could here the called Party for less then a second and then the call was dropped.

can somebody explain me what could be the cause of this? is it an asterisk-related bug, or some network-issue...or do we Need sip-debug enabled for more Information?

thanks in advance!

[2016-01-08 08:23:15] VERBOSE[3072][C-00002419] pbx.c: -- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/16-0000242d", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
[2016-01-08 08:23:15] VERBOSE[3072][C-00002419] pbx.c: -- Executing [s@macro-outbound-callerid:18] Set("SIP/16-0000242d", "CDR(outbound_cnum)=16") in new stack
[2016-01-08 08:23:15] VERBOSE[3072][C-00002419] pbx.c: -- Executing [s@macro-outbound-callerid:19] Set("SIP/16-0000242d", "CDR(outbound_cnam)=Mobil 2") in new stack
[2016-01-08 08:23:15] VERBOSE[3072][C-00002419] pbx.c: -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/16-0000242d", "0?sub-flp-2,s,1()") in new stack
[2016-01-08 08:23:15] VERBOSE[3072][C-00002419] pbx.c: -- Executing [s@macro-dialout-trunk:13] Set("SIP/16-0000242d", "OUTNUM=820382") in new stack
[2016-01-08 08:23:15] VERBOSE[3072][C-00002419] pbx.c: -- Executing [s@macro-dialout-trunk:14] Set("SIP/16-0000242d", "custom=SIP/arcor25602") in new stack
[2016-01-08 08:23:15] VERBOSE[3072][C-00002419] pbx.c: -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/16-0000242d", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)") in new stack
[2016-01-08 08:23:15] VERBOSE[3072][C-00002419] pbx.c: -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/16-0000242d", "0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))") in new stack
[2016-01-08 08:23:15] VERBOSE[3072][C-00002419] pbx.c: -- Executing [s@macro-dialout-trunk:17] Macro("SIP/16-0000242d", "dialout-trunk-predial-hook,") in new stack
[2016-01-08 08:23:15] VERBOSE[3072][C-00002419] pbx.c: -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/16-0000242d", "") in new stack
[2016-01-08 08:23:15] VERBOSE[3072][C-00002419] pbx.c: -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/16-0000242d", "0?bypass,1") in new stack
[2016-01-08 08:23:15] VERBOSE[3072][C-00002419] pbx.c: -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/16-0000242d", "1?Set(CONNECTEDLINE(num,i)=820382)") in new stack
[2016-01-08 08:23:15] VERBOSE[3072][C-00002419] pbx.c: -- Executing [s@macro-dialout-trunk:20] GotoIf("SIP/16-0000242d", "0?customtrunk") in new stack
[2016-01-08 08:23:15] VERBOSE[3072][C-00002419] pbx.c: -- Executing [s@macro-dialout-trunk:21] Dial("SIP/16-0000242d", "SIP/arcor25602/820382,300,Tt") in new stack
[2016-01-08 08:23:15] VERBOSE[3072][C-00002419] netsock2.c: == Using SIP RTP TOS bits 184
[2016-01-08 08:23:15] VERBOSE[3072][C-00002419] netsock2.c: == Using SIP RTP CoS mark 5
[2016-01-08 08:23:15] VERBOSE[3072][C-00002419] app_dial.c: -- Called SIP/arcor25602/820382
[2016-01-08 08:23:18] VERBOSE[3072][C-00002419] app_dial.c: -- SIP/arcor25602-0000242e is ringing
[2016-01-08 08:23:34] VERBOSE[3072][C-00002419] app_dial.c: -- SIP/arcor25602-0000242e answered SIP/16-0000242d
[2016-01-08 08:23:34] VERBOSE[3072][C-00002419] bridge_channel.c: -- Channel SIP/16-0000242d joined 'simple_bridge' basic-bridge <0a41b131-90fe-4d69-9518-ac7cb06d33e0>
[2016-01-08 08:23:34] VERBOSE[3074][C-00002419] bridge_channel.c: -- Channel SIP/arcor25602-0000242e joined 'simple_bridge' basic-bridge <0a41b131-90fe-4d69-9518-ac7cb06d33e0>
[2016-01-08 08:23:34] VERBOSE[3074][C-00002419] bridge_channel.c: -- Channel SIP/arcor25602-0000242e left 'simple_bridge' basic-bridge <0a41b131-90fe-4d69-9518-ac7cb06d33e0>
[2016-01-08 08:23:34] VERBOSE[3072][C-00002419] bridge_channel.c: -- Channel SIP/16-0000242d left 'simple_bridge' basic-bridge <0a41b131-90fe-4d69-9518-ac7cb06d33e0>
[2016-01-08 08:23:34] VERBOSE[3072][C-00002419] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 21) exited non-zero on 'SIP/16-0000242d' in macro 'dialout-trunk'
[2016-01-08 08:23:34] VERBOSE[3072][C-00002419] pbx.c: == Spawn extension (from-internal, 820382, 5) exited non-zero on 'SIP/16-0000242d'
[2016-01-08 08:23:34] VERBOSE[3072][C-00002419] pbx.c: -- Executing [h@from-internal:1] Hangup("SIP/16-0000242d", "") in new stack
[2016-01-08 08:23:34] VERBOSE[3072][C-00002419] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/16-0000242d'
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[david55]

To the extent that this can be debugged as an Asterisk issue, rather than a FreePBX one, you want the Asterisk Support forum. To the extent that it relates to FreePBX, you want http://community.freepbx.org/

In either case, you will need to enable SIP debugging, see https://wiki.asterisk.org/wiki/display/ ... nformation. However, I think that will just show that the called party cleared the call normally.

[comfine]

Thanks David for your reply.

normally an ending with non-zero in computerprograms means that there was an error, so i supposed that "exited non-zero" in asterisk means also, that there was some problem. is that assumption wrong?

i enabled sip-debug now to be able to analyze the problems in future.

[david55]

For Asterisk dialplan applications, zero means continue with the next dialplan step. Unless you override the behaviour with options, the Dial application will return non-zero if either party hangs up, and will return zero if the caller is still present, but the called was not answered.