Hi everybody. This is my scenario:
- SIP/6000 dials to SIP/6001
- I’m sending a manager action redirect, with both open channels to context of the MeetMe room (9999):
Action: Redirect
Channel: SIP/6000-00000018
ExtraChannel: SIP/6001-00000017
Exten: 9999
Context: casa
Priority: 1
- Both users can now speak over a MeetMe audio conference room.
BUT…
After this, the CALLED user (not the caller) hangup automatically after less time (from 1 minute to 1:40 sec).
I also try to change the SIP channels in my AMI actions putting Channel: SIP/6001-00000017 and ExtraChannel: SIP/6000-00000018 but with no changes. Always the CALLED one is hangupped.
I try to make a normal call from 6000 to 6001 and viceversa and there is not problems.
I try to make a normal call from 6000 to 9999 and from 6001 to 9999 and they call normally over the conference room.
The problem start only after the AMI Action REDIRECT.
How can I solve? Do you know if this is a knowed bug.
I’m using asterisk 1.6.2.11 with dahdi 2.4+dahdi 2.4 tools on Ubuntu 8.04.
This is the log:
[Sep 13 16:51:12] VERBOSE[5999] netsock.c: == Using SIP RTP CoS mark 5
[Sep 13 16:51:12] VERBOSE[7250] pbx.c: – Executing [6000@casa:1] Dial(“SIP/6001-00000017”, “SIP/6000”) in new stack
[Sep 13 16:51:12] VERBOSE[7250] netsock.c: == Using SIP RTP CoS mark 5
[Sep 13 16:51:12] VERBOSE[7250] app_dial.c: – Called 6000
[Sep 13 16:51:13] VERBOSE[7250] app_dial.c: – SIP/6000-00000018 is ringing
[Sep 13 16:51:17] VERBOSE[7250] app_dial.c: – SIP/6000-00000018 answered SIP/6001-00000017
[Sep 13 16:51:17] VERBOSE[7250] rtp.c: – Native bridging SIP/6001-00000017 and SIP/6000-00000018
[Sep 13 16:52:14] VERBOSE[7267] pbx.c: – Executing [9999@casa:1] MeetMe(“SIP/6000-00000018”, “9999,q”) in new stack
[Sep 13 16:52:14] VERBOSE[7267] config.c: == Parsing ‘/etc/asterisk/meetme.conf’: [Sep 13 16:52:14] VERBOSE[7267] config.c: == Found
[Sep 13 16:52:14] VERBOSE[7267] app_meetme.c: – Created MeetMe conference 1023 for conference ‘9999’
[Sep 13 16:52:14] VERBOSE[7250] pbx.c: == Spawn extension (casa, 9999, 1) exited non-zero on ‘SIP/6001-00000017’
[Sep 13 16:52:14] VERBOSE[7250] pbx.c: – Executing [9999@casa:1] MeetMe(“SIP/6001-00000017”, “9999,q”) in new stack
[Sep 13 16:52:46] VERBOSE[7267] pbx.c: == Spawn extension (casa, 9999, 1) exited non-zero on ‘SIP/6000-00000018’
[Sep 13 16:53:03] NOTICE[5999] chan_sip.c: Received SIP subscribe for peer without mailbox: 6000
[Sep 13 16:53:03] NOTICE[5999] chan_sip.c: Received SIP subscribe for peer without mailbox: 6001
[Sep 13 16:54:44] VERBOSE[7250] chan_dahdi.c: – Hungup ‘DAHDI/pseudo-602577886’
[Sep 13 16:54:44] VERBOSE[7250] pbx.c: == Spawn extension (casa, 9999, 1) exited non-zero on ‘SIP/6001-00000017’
[Sep 13 16:56:03] NOTICE[5999] chan_sip.c: Received SIP subscribe for peer without mailbox: 6000
[Sep 13 16:56:03] NOTICE[5999] chan_sip.c: Received SIP subscribe for peer without mailbox: 6001
THANK YOU VERY MUCH!