[asterisk-dev] Asterisk v1.6.0.10 broken ? Channel auto hangup...

[Serge Berney]

Hi everybody,

 

I’m using, compiled from source, Asterisk v1.6.0.10.

All is working excepted when user is calling, after a while, communication hang-up without error ! L

 

Ø  I receive a lot of “Unable to parse INFO message … with no human readable content” is that normal ?

Ø  What can I activate to enable more deep debug trace ?

Ø  Is there any known bug on chan_sip about this issue ?

 

Here’s the console log :

 

WARNING[3883] chan_sip.c: Unable to parse INFO message from 29ad1b5a3ef99c246f43b10e59109377 at 192.168.61.10. Content œ" ¬iÆ¡ !/¡ €) 

DEBUG[3883] chan_sip.c: Trying to put 'SIP/2.0 41' onto UDP socket destined for 192.168.11.189:5060

DEBUG[3922] rtp.c: Got RTCP report of 64 bytes

DEBUG[3883] chan_sip.c: Session timer stopped: 1422 - 6086bfea3d423e4f2a29c71842af0e8c at 192.168.61.10

DEBUG[3883] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 192.168.11.177:5060

DEBUG[3922] channel.c: Returning from native bridge, channels: SIP/sip_user1-082379b8, SIP/isdn-sip-08232a30

DEBUG[3922] channel.c: Hanging up channel 'SIP/isdn-sip-08232a30'

DEBUG[3922] chan_sip.c: Hangup call SIP/isdn-sip-08232a30, SIP callid 29ad1b5a3ef99c246f43b10e59109377 at 192.168.61.10

DEBUG[3922] chan_sip.c: Trying to put 'BYE sip:00' onto UDP socket destined for 192.168.11.189:5060

DEBUG[3922] rtp.c: Channel '<unspecified>' has no RTP, not doing anything

DEBUG[3922] app_dial.c: Exiting with DIALSTATUS=ANSWER.

DEBUG[3922] app_macro.c: Spawn extension (macro-dial_isdn,s,3) exited non-zero on 'SIP/sip_user1-082379b8' in macro 'dial_isdn'

VERBOSE[3922] logger.c:   == Spawn extension (macro-dial_isdn, s, 3) exited non-zero on 'SIP/sip_user1-082379b8' in macro 'dial_isdn'

DEBUG[3922] pbx.c: Spawn extension (kin_sortants,0041264848200,1) exited non-zero on 'SIP/sip_user1-082379b8'

VERBOSE[3922] logger.c:   == Spawn extension (kin_sortants, 0041264848200, 1) exited non-zero on 'SIP/sip_user1-082379b8'

DEBUG[3922] channel.c: Soft-Hanging up channel 'SIP/sip_user1-082379b8'

DEBUG[3922] channel.c: Hanging up channel 'SIP/sip_user1-082379b8'

DEBUG[3922] chan_sip.c: Hangup call SIP/sip_user1-082379b8, SIP callid 6086bfea3d423e4f2a29c71842af0e8c at 192.168.61.10

DEBUG[3883] chan_sip.c: Stopping retransmission on '29ad1b5a3ef99c246f43b10e59109377 at 192.168.61.10' of Request 103: Match Found

 

For information :

192.168.11.177 is the phone (Aastra)

192.168.11.189 is the PSTN gateway

192.168.61.10 is the Asterisk box

 

I tested few weeks ago with version 1.6.0.8 same bugs appear !

 

But all is working with version 1.4.23 and 1.4.24.1 (actually running)

 

Thanks in advance !

 

Serge Berney
Kin SA
Knowledge Integrated Networks
Rue du Bois-du-Lan 8
Case Postale 221
1217 Meyrin 1
Tél: +41 22 989 0 999
Fax: +41 22 989 0 998
s.berney at kinonline.net <mailto:s.berney at kinonline.net> 
www.kinonline.net <http://www.kinonline.net/> 

 

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