Hi, It's necessary to pay attention to the modularity of the system and overall scoping concerns when making this type of decision. The XMLRPC is a loadable module loaded by the core at runtime. "show" is an API command registered by mod_commands.so Because of this, it is not possible to extend the knowledge of XMLRPC to the 'show' command because there is no guarantee either module will be loaded. So, it's the API interface that is designed with a [string-in / string-out] prototype and the XMLRPC is only providing an interface to the API interface of freeswitch. This independance is mandatory and a fundamental of my design principle and anything else must be abandoned despite how tempting because the API interface is called from other methods such as a direct http connection, an event socket or from a scripting language such as javascript. However, It is completely possible to add support to the show command to parse a special input param and deliver that string-out as XML for instance: "show channels as xml" "show channels as csv" Each API command must provide it's own complete interpretation of the input and decision about what to output and since the entire system only has one "show" command it is not very difficult to implement this and it would be happily accepted as a patch to mod_commands.c Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Jonas Gauffin <jonas.gauffin@gmail.com> To: freeswitch-dev@lists.freeswitch.org Sent: Tuesday, March 6, 2007 4:08:42 AM Subject: Re: [Freeswitch-dev] XML-RPC Correction: <array> <data> <value><string>0983de41-2562-fa44-8c0c-a6e5747b2c17,2007-03-06</string></value> <value><dateTime.iso8601>20070306T10:42:01</dateTime.iso8601></value> <value><string>PortAudio/1234</string></value> </data> </array> You can also use the <struct> tag to be able to assign names to each value. More info: http://www.xmlrpc.com/spec/ On 3/6/07, Jonas Gauffin <jonas.gauffin@gmail.com> wrote:
Hello
I've played around with the XML-RPC implementation and I have some suggestions. The output of all commands are now simple text strings. That makes the result parsing quite hard for applications that would like to use the interface (and it's really telnet commands wrapped into a xmlrpc body).
For instance, take the "show channels" result. It currently looks like this (have removed the xml rpc body):
uuid,created,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,read_codec,read_rate,write_codec,write_rate 0983de41-2562-fa44-8c0c-a6e5747b2c17,2007-03-06 10:42:01,PortAudio/1234,CS_EXECUTE,FreeSwitch,5555551212,,1234,bridge,sofia/mydomain.com/1234@conference.freeswitch.org,L16,8000,L16,8000 37138427-cdb7-324c-b200-7123c063d473,2007-03-06 10:42:02,sofia/mydomain.com/1234@conference.freeswitch.org,CS_LOOPBACK,FreeSwitch,5555551212,,mydomain.com/1234@conference.freeswitch.org,,,PCMU,8000,PCMU,8000
2 total.
That should have been wrapped into xml instead:
<channels count="2"> <channel> <uuid>0983de41-2562-fa44-8c0c-a6e5747b2c17,2007-03-06</uuid> <created>2007-03-06 10:42:01</created> <name>PortAudio/1234</name> [... and all other information ...] </channel> </channels>
Same goes for all other commands implemented in XML-RPC.
Best regards, Jonas
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org ____________________________________________________________________________________ Get your own web address. Have a HUGE year through Yahoo! Small Business. http://smallbusiness.yahoo.com/domains/?p=BESTDEAL
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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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
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?
You have to remember that FreeSWITCH isn't a sip stack but Sofia-SIP is our stack of choice. I suspect that the NUA layer isn't passing this up to our layer to deal with. I'm sure we'll have to bring this to their attention for a fix ;) But you can open a bug on this at jira.freeswitch.org. Thanks, Brian
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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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
Is it enough to use the latest tarball? I cannot access via SVN from my work PC, so I can try this? 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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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
Just an FYI we do run svn on port 81 to bypass transparent proxy. /b On Mar 7, 2007, at 11:27 AM, Sluschny, Thomas wrote:
Is it enough to use the latest tarball? I cannot access via SVN from my work PC, so I can try this?
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
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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
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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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
participants (4)
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Anthony Minessale
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Brian West
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Michael Jerris
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Sluschny, Thomas