Hi, I am trying to use FS to make outgoing SIP calls. I have a number of gateways that can make the call. However, if one of them is down or has some other problem then I would like to detect that quickly. I intended to use the provisional '100 Trying' message for this... if it hasn't been received in a couple of seconds then go on and try the next gateway. But I can't find a flag/event/state which corresponds to receipt of this message. Can anyone tell me where I should be looking? I put a debug print in sofia_event_callback for every event but there doesn't seem to be one fired for this condition. Thanks in advance. {P^/ John
That 100 trying is handled deep in the sip stack. The author of sofia said it would be a big job to bring that up to the even callback. Someone may be able to persuade him to allow you to pass a global timeout waiting for 100 or something but no solution exists atm On Tue, Apr 21, 2009 at 3:32 AM, John Dalgliesh <johnd@defyne.org> wrote:
Hi,
I am trying to use FS to make outgoing SIP calls. I have a number of gateways that can make the call. However, if one of them is down or has some other problem then I would like to detect that quickly.
I intended to use the provisional '100 Trying' message for this... if it hasn't been received in a couple of seconds then go on and try the next gateway.
But I can't find a flag/event/state which corresponds to receipt of this message. Can anyone tell me where I should be looking? I put a debug print in sofia_event_callback for every event but there doesn't seem to be one fired for this condition.
Thanks in advance.
{P^/ John
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Hi Anthony, Thanks for the reply! While waiting for my question to appear on the list yesterday (6H delay at yoda.ostag.org... is first post moderated?) I went deep into the SIP stack and figured out the solution: You just have to give NTATAG_PASS_100(1) as one of the tags for nua_create. Then you get a sofia event for it. I guess the author has made it easier since your last discussion. I have changed my mod_sofia to do this. I also added a channel flag which is set if any response has been received from the remote end (be it 100, 18X, 2XX, etc.). The flag is now tested by switch_ivr_originate to time-out a call quickly. Would you/anyone be interested in a patch to do this? If so please let me know the procedure for posting patches etc. {P^/ On Tue, 21 Apr 2009 at 10:03 -0500, Anthony Minessale wrote:
That 100 trying is handled deep in the sip stack. The author of sofia said it would be a big job to bring that up to the even callback. Someone may be able to persuade him to allow you to pass a global timeout waiting for 100 or something but no solution exists atm
On Tue, Apr 21, 2009 at 3:32 AM, John Dalgliesh <johnd@defyne.org> wrote:
Hi,
I am trying to use FS to make outgoing SIP calls. I have a number of gateways that can make the call. However, if one of them is down or has some other problem then I would like to detect that quickly.
I intended to use the provisional '100 Trying' message for this... if it hasn't been received in a couple of seconds then go on and try the next gateway.
But I can't find a flag/event/state which corresponds to receipt of this message. Can anyone tell me where I should be looking? I put a debug print in sofia_event_callback for every event but there doesn't seem to be one fired for this condition.
Thanks in advance.
{P^/ John
Patches should be posted to http://jira.freeswitch.org, Thanks for digging in. /b On Apr 21, 2009, at 10:22 PM, John Dalgliesh wrote:
Hi Anthony,
Thanks for the reply!
While waiting for my question to appear on the list yesterday (6H delay at yoda.ostag.org... is first post moderated?) I went deep into the SIP stack and figured out the solution: You just have to give NTATAG_PASS_100(1) as one of the tags for nua_create. Then you get a sofia event for it. I guess the author has made it easier since your last discussion.
I have changed my mod_sofia to do this. I also added a channel flag which is set if any response has been received from the remote end (be it 100, 18X, 2XX, etc.). The flag is now tested by switch_ivr_originate to time-out a call quickly.
Would you/anyone be interested in a patch to do this? If so please let me know the procedure for posting patches etc.
{P^/
Brian West brian@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com
I guess maybe he added it as a result of our last conversation and never told me. yes, if you do it elegantly we can add that patch. submit what you have to jira in we can do code review. On Tue, Apr 21, 2009 at 10:38 PM, Brian West <brian@freeswitch.org> wrote:
Patches should be posted to http://jira.freeswitch.org, Thanks for digging in.
/b
On Apr 21, 2009, at 10:22 PM, John Dalgliesh wrote:
Hi Anthony,
Thanks for the reply!
While waiting for my question to appear on the list yesterday (6H delay at
yoda.ostag.org... is first post moderated?) I went deep into the SIP stack
and figured out the solution: You just have to give NTATAG_PASS_100(1) as one of the tags for nua_create. Then you get a sofia event for it. I guess
the author has made it easier since your last discussion.
I have changed my mod_sofia to do this. I also added a channel flag which is set if any response has been received from the remote end (be it 100, 18X, 2XX, etc.). The flag is now tested by switch_ivr_originate to time-out a call quickly.
Would you/anyone be interested in a patch to do this? If so please let me know the procedure for posting patches etc.
{P^/
Brian West brian@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
participants (3)
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Anthony Minessale
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Brian West
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John Dalgliesh