There were some fatal known issues with 1.12.5.  We are currently testing a recent snapshot after 1.12.5 where I believe those fatal issues are resolved and that should merge into tree in the next week or so.  In the mean time, if you are using it for anything more than lab testing, I would suggest using current sofia darcs tree over 1.12.5.

Mike

---

From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Sluschny, Thomas
Sent: Monday, March 19, 2007 6:09 AM
To: freeswitch-dev@lists.freeswitch.org
Subject: Re: [Freeswitch-dev] Decline not routed from sip stack tofreeswitch?

I moved to sofia 1.12.5 and used the latest version, now everything is OK:

recv 357 bytes from udp/[192.168.1.1]:5060 at 10:00:06.054861:
   ------------------------------------------------------------------------
   SIP/2.0 603 Decline
   Via: SIP/2.0/UDP 192.168.1.111;rport=5060;branch=z9hG4bK5B55XgBQUjmXK
   From: "FreeSwitch" <sip:5555551212@192.168.1.111>;tag=53SQH40g915NQ
   To: <sip:1@192.168.1.1>;tag=2626728529
   Call-ID: 74190746-50a3-122a-2780-39a48cb53b8d
   CSeq: 81030866 INVITE
   Warning: 399 192.168.1.1 "Information_Call_1 is already active"
   Content-Length: 0

   ------------------------------------------------------------------------
send 312 bytes to udp/[192.168.1.1]:5060 at 10:00:06.132991:
   ------------------------------------------------------------------------
   ACK sip:1@192.168.1.1 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bK5B55XgBQUjmXK
   Max-Forwards: 70
   From: "FreeSwitch" <sip:5555551212@192.168.1.111>;tag=53SQH40g915NQ
   To: <sip:1@192.168.1.1>;tag=2626728529
   Call-ID: 74190746-50a3-122a-2780-39a48cb53b8d
   CSeq: 81030866 ACK
   Content-Length: 0

   ------------------------------------------------------------------------
2007-03-19 11:00:06 [DEBUG] mod_sofia.c:4845 event_callback() event [nua_r_invite] status [603][Decline] session: sofia/192.168.1.111/1@192.168.1.1
2007-03-19 11:00:06 [DEBUG] mod_sofia.c:4845 event_callback() event [nua_i_state] status [603][Decline] session: sofia/192.168.1.111/1@192.168.1.1
2007-03-19 11:00:06 [DEBUG] mod_sofia.c:2912 sip_i_state() Channel sofia/192.168.1.111/1@192.168.1.1 entering state [terminated]

Thomas

---

Von: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] Im Auftrag von Michael Jerris
Gesendet: Mittwoch, 7. März 2007 17:08
An: freeswitch-dev@lists.freeswitch.org
Betreff: Re: [Freeswitch-dev] Decline not routed from sip stack tofreeswitch?

I believe I fixed this issue last week, and your line numbers look off from current tree.  What svn revision are you using?  Can you try with current trunk to confirm it is still an issue.?

Mike

---

From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Sluschny, Thomas
Sent: Wednesday, March 07, 2007 5:48 AM
To: freeswitch-dev@lists.freeswitch.org
Subject: [Freeswitch-dev] Decline not routed from sip stack to freeswitch?

Hi,

like the subjects says:

Decline not routed from sip stack (sofia) to freeswitch (mod_sofia)?

Here is a logging with "set TPORT_LOG=1" to see the SIP messages.

Freeswitch has sent an INVITE, the answer was RINGING an then DECLINE.

But: the DECLINE is not received in mod_sofia!!!

I expected a function call of event_callback() from sofia stack, but there is none!!

So the freeswitch waits until timeout occurs:

_______________________________________________________________________________________________________________________________

2007-03-06 18:49:50 [DEBUG] mod_sofia.c:4373 event_callback() event [nua_r_invite] status [180][Ringing] session: sofia/192.168.1.111/3@192.168.1.101
2007-03-06 18:49:50 [DEBUG] mod_sofia.c:4373 event_callback() event [nua_i_state] status [180][Ringing] session: sofia/192.168.1.111/3@192.168.1.101
2007-03-06 18:49:50 [DEBUG] mod_sofia.c:2565 sip_i_state() Channel sofia/192.168.1.111/3@192.168.1.101 entering state [proceeding]
recv 323 bytes from udp/[192.168.1.101]:5060 at 17:49:50.818256:
   ------------------------------------------------------------------------
   SIP/2.0 603 Decline
   Via: SIP/2.0/UDP 192.168.1.111;rport=5060;branch=z9hG4bK9a2m30XmH07Fj
   From: "FreeSwitch" <sip:5555551212@192.168.1.111>;tag=8cXm6erteSH3N
   To: <sip:3@192.168.1.101>;tag=3823224407
   Call-ID: ec363031-46ad-122a-2780-39a48cb53b8d
   CSeq: 80483359 INVITE
   Warning: Playing 16kHz-File
   Content-Length: 0

   ------------------------------------------------------------------------
2007-03-06 18:50:00 [NOTICE] switch_ivr.c:2752 switch_ivr_originate() Hangup sofia/192.168.1.111/2@192.168.1.101 [CS_HOLD] [NO_ANSWER]
2007-03-06 18:50:00 [INFO] switch_channel.c:969 switch_channel_perform_hangup() Kill sofia/192.168.1.111/2@192.168.1.101 [1]
2007-03-06 18:50:00 [DEBUG] switch_core.c:2903 switch_core_session_run() (sofia/192.168.1.111/2@192.168.1.101) State HANGUP
2007-03-06 18:50:00 [DEBUG] mod_sofia.c:1282 sofia_on_hangup() Channel sofia/192.168.1.111/2@192.168.1.101 hanging up, cause: NO_ANSWER

_______________________________________________________________________________________________________________________________

Can I configure sofia to supress the event_callback() for some messages (e. g. Decline)?

Or: has anyone ever seen a DECLINE in freeswitch?

Thanks in advance

Thomas