[Freeswitch-users] Status of mod_skinny? is it being maintained?

Nathan Neulinger nneul at mst.edu
Thu Mar 14 23:59:13 MSK 2013


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 at mst.edu <mailto:nneul at 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 at mst.edu <mailto:nneul at mst.edu> <mailto:nneul at mst.edu
>         <mailto:nneul at 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 at mst.edu <mailto:nneul at mst.edu>
>         <mailto:nneul at mst.edu <mailto:nneul at mst.edu>> <mailto:nneul at mst.edu <mailto:nneul at mst.edu>
>
>                  <mailto:nneul at mst.edu <mailto:nneul at 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 at mst.edu <mailto:nneul at mst.edu> <mailto:nneul at mst.edu <mailto:nneul at 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 at hotmail.__com <mailto:MSN%3Aanthony_minessale at hotmail.com>
>         <mailto:MSN%3Aanthony___minessale at hotmail.com <mailto:MSN%253Aanthony_minessale at hotmail.com>>
>         GTALK/JABBER/PAYPAL:anthony.__minessale at gmail.com <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>         <mailto:PAYPAL%3Aanthony.__minessale at gmail.com <mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
>         IRC: irc.freenode.net <http://irc.freenode.net> <http://irc.freenode.net> #freeswitch
>
>         FreeSWITCH Developer Conference
>
>         sip:888 at conference.freeswitch.__org <mailto:sip%3A888 at conference.freeswitch.org>
>         <mailto:sip%3A888 at conference.__freeswitch.org <mailto:sip%253A888 at conference.freeswitch.org>>
>         googletalk:conf+888 at __conference.freeswitch.org <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>         <mailto:googletalk%3Aconf%__2B888 at conference.freeswitch.__org
>         <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
>         pstn:+19193869900 <tel:%2B19193869900>
>
>
>     --
>     ------------------------------__------------------------------
>     Nathan Neulinger nneul at mst.edu <mailto:nneul at 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 at hotmail.com <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <mailto:sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900

-- 
------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect



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