mod_conference: behavior when the conference is torn down before 31seconds
Hi all, I am experiencing a behavior that I cannot clearly understand. Basically I "autocall" a few phones into a conference with the sip_auto_answer set to true, as follows: <extension name="extension-intercom"> <condition field="destination_number" expression="^773$"> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true}"/> <action application="conference_set_auto_outcall" data="user/305"/> <action application="conference_set_auto_outcall" data="user/303"/> <action application="conference_set_auto_outcall" data="user/201"/> <action application="conference" data="412+flags{endconf|deaf}"/> <action application="conference" data="412 kick all"/> </condition> </extension> The conference establishes just fine and everyone can hear just fine. The "strange" behavior comes when the person calling to ext 773 hangs up before 31 seconds have passed, the rest of the phones stay up until they reach second 31 into the "conference". I am using snom phones and I see the BYE message arriving at the phones exactly at second 31 after the call establishes. The conference itself however does not exist after the person calling 773 hangs up (doing conference list on CLI shows NO active conferences). If the conference stays up more than 31 seconds, then when the person calling 773 hangs up, the rest of the phones hang up immediately as desired. Here's the log for a "page" that lasts less than 31 seconds: http://pastebin.freeswitch.org/9773 Here's the log of the phone for a "page" that lasts less than 31 seconds: http://pastebin.freeswitch.org/9774 Your inout is appreciated. Regards, Luis
Hello Luis, are you using encrypted TLS instead on SIP on this phone? I experienced a similar behaviour with 31 seocnds on TLS. Best regards Peter Luis F Urrea schrieb:
Hi all,
I am experiencing a behavior that I cannot clearly understand. Basically I "autocall" a few phones into a conference with the sip_auto_answer set to true, as follows:
<extension name="extension-intercom"> <condition field="destination_number" expression="^773$"> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true}"/> <action application="conference_set_auto_outcall" data="user/305"/> <action application="conference_set_auto_outcall" data="user/303"/> <action application="conference_set_auto_outcall" data="user/201"/> <action application="conference" data="412+flags{endconf|deaf}"/> <action application="conference" data="412 kick all"/> </condition> </extension>
The conference establishes just fine and everyone can hear just fine.
The "strange" behavior comes when the person calling to ext 773 hangs up before 31 seconds have passed, the rest of the phones stay up until they reach second 31 into the "conference".
I am using snom phones and I see the BYE message arriving at the phones exactly at second 31 after the call establishes.
The conference itself however does not exist after the person calling 773 hangs up (doing conference list on CLI shows NO active conferences).
If the conference stays up more than 31 seconds, then when the person calling 773 hangs up, the rest of the phones hang up immediately as desired.
Here's the log for a "page" that lasts less than 31 seconds:
http://pastebin.freeswitch.org/9773
Here's the log of the phone for a "page" that lasts less than 31 seconds:
http://pastebin.freeswitch.org/9774
Your inout is appreciated.
Regards,
Luis ------------------------------------------------------------------------
_______________________________________________ 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
the problem stems from the fact that you did an outcall to those other addresses. The 30 sec is the timeout waiting for those calls to establish. The outcome of those outbound calls must be determined before the conf will end. On Sat, Jul 18, 2009 at 7:01 AM, Peter P GMX <Prometheus001@gmx.net> wrote:
Hello Luis,
are you using encrypted TLS instead on SIP on this phone? I experienced a similar behaviour with 31 seocnds on TLS.
Best regards Peter
Luis F Urrea schrieb:
Hi all,
I am experiencing a behavior that I cannot clearly understand. Basically I "autocall" a few phones into a conference with the sip_auto_answer set to true, as follows:
<extension name="extension-intercom"> <condition field="destination_number" expression="^773$"> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true}"/> <action application="conference_set_auto_outcall" data="user/305"/> <action application="conference_set_auto_outcall" data="user/303"/> <action application="conference_set_auto_outcall" data="user/201"/> <action application="conference" data="412+flags{endconf|deaf}"/> <action application="conference" data="412 kick all"/> </condition> </extension>
The conference establishes just fine and everyone can hear just fine.
The "strange" behavior comes when the person calling to ext 773 hangs up before 31 seconds have passed, the rest of the phones stay up until they reach second 31 into the "conference".
I am using snom phones and I see the BYE message arriving at the phones exactly at second 31 after the call establishes.
The conference itself however does not exist after the person calling 773 hangs up (doing conference list on CLI shows NO active conferences).
If the conference stays up more than 31 seconds, then when the person calling 773 hangs up, the rest of the phones hang up immediately as desired.
Here's the log for a "page" that lasts less than 31 seconds:
http://pastebin.freeswitch.org/9773
Here's the log of the phone for a "page" that lasts less than 31 seconds:
http://pastebin.freeswitch.org/9774
Your inout is appreciated.
Regards,
Luis ------------------------------------------------------------------------
_______________________________________________ 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
_______________________________________________ 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/ Twitter: http://twitter.com/FreeSWITCH_wire 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)
-
Anthony Minessale
-
Luis F Urrea
-
Peter P GMX