Re: [Freeswitch-users] Status of mod_skinny? is it being maintained?
ring_ready worked! Why is the behavior different here vs. sip target phone? both are coming in from the same external caller. Does the sofia endpoint send back that ring ready status automatically? -- Nathan On 03/14/2013 03:50 PM, Anthony Minessale wrote:
The inbound SIP leg cancels the call before the ring time timeout. So the box calling this box [10.20.0.26] is canceling it after 10 seconds. so it has a shorter timeout than the box you are debugging and you don't last long enough to make it to VM.
Try adding the ring_ready app to your dial-plan right before you bridge to skinny. If that doesn't work try answer app, just to prove if you answer it that the down-stream leg will not cancel.
On Thu, Mar 14, 2013 at 3:43 PM, Nathan Neulinger <nneul@mst.edu <mailto:nneul@mst.edu>> wrote:
Sorry, new to the diagnostics... redoing now.
http://pastebin.freeswitch.__org/20692 <http://pastebin.freeswitch.org/20692>
On 03/14/2013 03:39 PM, Anthony Minessale wrote:
the sip trace is not in there still. I assume this was the actual log file and not a console cap. In that case you either need to add console level to the log map or just do sofia tracelevel debug so the traces are at debug level instead of console.
On Thu, Mar 14, 2013 at 3:23 PM, Nathan Neulinger <nneul@mst.edu <mailto:nneul@mst.edu> <mailto:nneul@mst.edu <mailto:nneul@mst.edu>>> wrote:
http://pastebin.freeswitch.____org/20691 <http://pastebin.freeswitch.__org/20691 <http://pastebin.freeswitch.org/20691>>
This is with a call coming in from an external sip trunk. If you want me to test from a sip from directly attached to freeswitch I can do that as well.
-- Nathan
On 03/14/2013 03:06 PM, Anthony Minessale wrote:
Can you repeat that log with "sofia global siptrace on" Something odd is happening with the SIP leg on the other side that I can't see,
On Thu, Mar 14, 2013 at 2:24 PM, Nathan Neulinger <nneul@mst.edu <mailto:nneul@mst.edu> <mailto:nneul@mst.edu <mailto:nneul@mst.edu>> <mailto:nneul@mst.edu <mailto:nneul@mst.edu>
<mailto:nneul@mst.edu <mailto:nneul@mst.edu>>>> wrote:
I know, but I definitely didn't hang up on the calling side.
For the SIP phones, I can let them ring right up to the full timeout. With the Cisco, it's stopping after 10s every time and not doing any rollover to next step in dialplan.
-- ------------------------------____----------------------------__--
Nathan Neulinger nneul@mst.edu <mailto:nneul@mst.edu> <mailto:nneul@mst.edu <mailto:nneul@mst.edu>> Missouri S&T Information Technology (573) 612-1412 <tel:%28573%29%20612-1412> <tel:%28573%29%20612-1412>
System Administrator - Architect
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH___wire <http://twitter.com/FreeSWITCH_wire>
AIM: anthm MSN:anthony_minessale@hotmail.__com <mailto:MSN%3Aanthony_minessale@hotmail.com> <mailto:MSN%3Aanthony___minessale@hotmail.com <mailto:MSN%253Aanthony_minessale@hotmail.com>> GTALK/JABBER/PAYPAL:anthony.__minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> <mailto:PAYPAL%3Aanthony.__minessale@gmail.com <mailto:PAYPAL%253Aanthony.minessale@gmail.com>> IRC: irc.freenode.net <http://irc.freenode.net> <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.__org <mailto:sip%3A888@conference.freeswitch.org> <mailto:sip%3A888@conference.__freeswitch.org <mailto:sip%253A888@conference.freeswitch.org>> googletalk:conf+888@__conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> <mailto:googletalk%3Aconf%__2B888@conference.freeswitch.__org <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org>> pstn:+19193869900 <tel:%2B19193869900>
-- ------------------------------__------------------------------ Nathan Neulinger nneul@mst.edu <mailto:nneul@mst.edu> Missouri S&T Information Technology (573) 612-1412 <tel:%28573%29%20612-1412> System Administrator - Architect
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
-- ------------------------------------------------------------ Nathan Neulinger nneul@mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect
Thanks to everyone for helping on this - fix is in and appears to be working beautifully. Problem looked to be that the skinny code was sending back the wrong ring_ready message. I would be very interested in getting some idea of who is using the skinny support though from a regression testing standpoint - I'm new to the codebase.... -- Nathan On 03/14/2013 03:59 PM, Nathan Neulinger wrote:
ring_ready worked!
Why is the behavior different here vs. sip target phone? both are coming in from the same external caller. Does the sofia endpoint send back that ring ready status automatically?
-- Nathan
On 03/14/2013 03:50 PM, Anthony Minessale wrote:
The inbound SIP leg cancels the call before the ring time timeout. So the box calling this box [10.20.0.26] is canceling it after 10 seconds. so it has a shorter timeout than the box you are debugging and you don't last long enough to make it to VM.
Try adding the ring_ready app to your dial-plan right before you bridge to skinny. If that doesn't work try answer app, just to prove if you answer it that the down-stream leg will not cancel.
On Thu, Mar 14, 2013 at 3:43 PM, Nathan Neulinger <nneul@mst.edu <mailto:nneul@mst.edu>> wrote:
Sorry, new to the diagnostics... redoing now.
http://pastebin.freeswitch.__org/20692 <http://pastebin.freeswitch.org/20692>
On 03/14/2013 03:39 PM, Anthony Minessale wrote:
the sip trace is not in there still. I assume this was the actual log file and not a console cap. In that case you either need to add console level to the log map or just do sofia tracelevel debug so the traces are at debug level instead of console.
On Thu, Mar 14, 2013 at 3:23 PM, Nathan Neulinger <nneul@mst.edu <mailto:nneul@mst.edu> <mailto:nneul@mst.edu <mailto:nneul@mst.edu>>> wrote:
http://pastebin.freeswitch.____org/20691 <http://pastebin.freeswitch.__org/20691 <http://pastebin.freeswitch.org/20691>>
This is with a call coming in from an external sip trunk. If you want me to test from a sip from directly attached to freeswitch I can do that as well.
-- Nathan
On 03/14/2013 03:06 PM, Anthony Minessale wrote:
Can you repeat that log with "sofia global siptrace on" Something odd is happening with the SIP leg on the other side that I can't see,
On Thu, Mar 14, 2013 at 2:24 PM, Nathan Neulinger <nneul@mst.edu <mailto:nneul@mst.edu> <mailto:nneul@mst.edu <mailto:nneul@mst.edu>> <mailto:nneul@mst.edu <mailto:nneul@mst.edu>
<mailto:nneul@mst.edu <mailto:nneul@mst.edu>>>> wrote:
I know, but I definitely didn't hang up on the calling side.
For the SIP phones, I can let them ring right up to the full timeout. With the Cisco, it's stopping after 10s every time and not doing any rollover to next step in dialplan.
-- ------------------------------____----------------------------__--
Nathan Neulinger nneul@mst.edu <mailto:nneul@mst.edu> <mailto:nneul@mst.edu <mailto:nneul@mst.edu>> Missouri S&T Information Technology (573) 612-1412 <tel:%28573%29%20612-1412> <tel:%28573%29%20612-1412>
System Administrator - Architect
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH___wire <http://twitter.com/FreeSWITCH_wire>
AIM: anthm MSN:anthony_minessale@hotmail.__com <mailto:MSN%3Aanthony_minessale@hotmail.com> <mailto:MSN%3Aanthony___minessale@hotmail.com <mailto:MSN%253Aanthony_minessale@hotmail.com>> GTALK/JABBER/PAYPAL:anthony.__minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> <mailto:PAYPAL%3Aanthony.__minessale@gmail.com <mailto:PAYPAL%253Aanthony.minessale@gmail.com>> IRC: irc.freenode.net <http://irc.freenode.net> <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.__org <mailto:sip%3A888@conference.freeswitch.org> <mailto:sip%3A888@conference.__freeswitch.org <mailto:sip%253A888@conference.freeswitch.org>> googletalk:conf+888@__conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> <mailto:googletalk%3Aconf%__2B888@conference.freeswitch.__org <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org>> pstn:+19193869900 <tel:%2B19193869900>
-- ------------------------------__------------------------------ Nathan Neulinger nneul@mst.edu <mailto:nneul@mst.edu> Missouri S&T Information Technology (573) 612-1412 <tel:%28573%29%20612-1412> System Administrator - Architect
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
-- ------------------------------------------------------------ Nathan Neulinger nneul@mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect
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