search for: onlyperson

...piece of hardware: Post 1) ********************************************************* Anyone ever seen MeetMe cause * to crash? Specifically, it happens consistantly if someone begins to enter a conference and then decides to hangup while Allison is introducing them - like playing back "conf-onlyperson". This has been seen with the MeetMe participant connecting via IAX and SIP (not saying it doesn't happen with Zap, just that I haven't seen it). The box is * 1.2.5, Zaptel 1.2.4, a TDM400P loaded with 3xFXO cards, Mandriva 2006 Free. Symptoms of the crash: once the participant hangs...

Anyone ever seen MeetMe cause * to crash? Specifically, it happens consistantly if someone begins to enter a conference and then decides to hangup while Allison is introducing them - like playing back "conf-onlyperson". This has been seen with the MeetMe participant connecting via IAX and SIP (not saying it doesn't happen with Zap, just that I haven't seen it). The box is * 1.2.5, Zaptel 1.2.4, a TDM400P loaded with 3xFXO cards, Mandriva 2006 Free. Symptoms of the crash: once the participant hangs...

Hi Brent > Anyone ever seen MeetMe cause * to crash? Specifically, it happens > consistantly if someone begins to enter a conference and then decides to > hangup while Allison is introducing them - like playing back > "conf-onlyperson". This has been seen with the MeetMe participant > connecting via IAX and SIP (not saying it doesn't happen with Zap, just > that I haven't seen it). Thank you for the hint. Now finaly I can 100% reproduce the problem. Yes, if I hang up during Playing 'conf-onlyperson'...

In article <200603181001.08589.benoit.panizzon@imp.ch>, benoit.panizzon@imp.ch says... > Thank you for the hint. Now finaly I can 100% reproduce the problem. Yes, if I > hang up during Playing 'conf-onlyperson' my machine freezes. It's not a GSM > Enconding problem as I suspected first, this happens with every encoding. > > magma*CLI> > -- Executing Answer("SIP/11-9d7c", "") in new stack > -- Executing MeetMe("SIP/11-9d7c", "555"...

...Answer("DAHDI/1-1", "") in new stack -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1", "1") in new stack -- Executing [266 at internal:3] ConfBridge("DAHDI/1-1", "1") in new stack -- <DAHDI/1-1> Playing 'conf-onlyperson.ulaw' (language 'en') ....... extensions.conf: [globals] ....... GOTO_ON_BLINDXFR="internal,266,1" features.conf: [featuremap] blindxfer => #1 But: -- Executing [s at DialOut:14] Dial("DAHDI/1-1", "motif/xxxx/+1234567890a at voice.google.com,,rTt&quot...

I just installed asterisk@home to see how it works and seems there is a problem with sounds... I dont hear any announcements or recordings... sounds are on /var/lib/asterisk/sounds and the logs show this: -- Created MeetMe conference 1023 for conference '8200' -- Playing 'conf-onlyperson' (language 'en') But I dont hear anything... any experiences with this kind of errors and/or asterisk@home? Thx! __________________________________________________________________ Anton Krall

...24 16:16:59 WARNING[1732]: channel.c:2597 ast_request: No channel type registered for 'zap' Oct 24 16:16:59 WARNING[1732]: app_meetme.c:465 build_conf: Unable to open pseudo channel - trying device -- Created MeetMe conference 1023 for conference '5000' -- Playing 'conf-onlyperson' (language 'en') -- Hungup 'IAX2/xxx.yyy.142.204:4569-2' However, I have zaptel and ztdummy drivers loaded: demeter:(acd1)asterisk # lsmod Module Size Used by ztdummy 3464 0 zaptel 218756 1 ztdummy usbhid...

...t;, "") in new stack >> -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1", "1") in new stack >> -- Executing [266 at internal:3] ConfBridge("DAHDI/1-1", "1") in new stack >> -- <DAHDI/1-1> Playing 'conf-onlyperson.ulaw' (language 'en') >> ....... >> >> >> extensions.conf: >> >> [globals] >> ....... >> GOTO_ON_BLINDXFR="internal,266,1" >> >> features.conf: >> >> [featuremap] >> blindxfer => #1 >> >&...

...------------------------ > Channel SIP/cc101-0887f498 was answered. -- Executing MeetMe("SIP/cc101-0887f498", "8600051") in new stack == Parsing '/etc/asterisk/meetme.conf': Found -- Created MeetMe conference 1023 for conference '8600051' -- Playing 'conf-onlyperson' (language 'en') == Manager 'sendcron' logged off from 127.0.0.1 -- Executing Ringing("SIP/66.54.140.46-08874e68", "") in new stack -- Executing Wait("SIP/66.54.140.46-08874e68", "1") in new stack -- Executing Answer("SIP/66.54.140.46...

...ecuting [266 at internal:2] SendDTMF("DAHDI/1-1", "1") in new >>>> stack >>>> -- Executing [266 at internal:3] ConfBridge("DAHDI/1-1", "1") in >>>> new stack >>>> -- <DAHDI/1-1> Playing 'conf-onlyperson.ulaw' (language 'en') >>>> ....... >>>> >>>> >>>> extensions.conf: >>>> >>>> [globals] >>>> ....... >>>> GOTO_ON_BLINDXFR="internal,266,1" >>>> >>>> features...

...; Jan 27 11:02:23 WARNING[6133]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device -- Created MeetMe conference 1023 for conference '81' Jan 27 11:02:23 DEBUG[6133]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'conf-onlyperson' (language 'si') Jan 27 11:02:23 DEBUG[6133]: chan_iax2.c:5346 socket_read: Ooh, voice format changed to 2 Jan 27 11:02:27 DEBUG[6133]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Jan 27 11:02:27 DEBUG[6133]: channel.c:1128 ast_settimeout: Scheduling timer at 0...

...- Executing [300@nway-conf:4] Set("SIP/ca1-ab110040", "DYNAMIC_FEATURES=") in new stack -- Executing [300@nway-conf:5] MeetMe("SIP/ca1-ab110040", "300|pdMX") in new stack -- Created MeetMe conference 1023 for conference '300' -- Playing 'conf-onlyperson' (language 'en') [Apr 19 16:14:15] WARNING[22995]: rtp.c:874 ast_rtcp_read: RTCP Read too short -- Started music on hold, class 'default', on SIP/ca1-ab110040 -- User entered '33' -- Executing [s@macro-nway-conf-start:4] Set("SIP/ua1-ac750040", "DYNA...

Hi, is the following possible? I would like to transfer a call to my "personal" MeetMe conference room and get transferred there automatically as well. Currently I can transfer the call to the conference, have to hangup and then call the conference number myself. I would love to have this in one quick function. Moreover is there a way to disable the "You are currently the only

...-- Executing MeetMe("IAX2/200@200/1", "81|pMs") in new stack == Parsing '/etc/asterisk/meetme.conf': Found == Parsing '/etc/asterisk/meetme_additional.conf': Found -- Created MeetMe conference 1023 for conference '81' -- Playing 'conf-onlyperson' (language 'si') -- Started music on hold, class 'default', on IAX2/200@200/1 ------------------------------------------------------------------------- Any hint, any advice ? Regards, Rob.

...t;DAHDI/1-1", "") in new stack > -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1", "1") in new stack > -- Executing [266 at internal:3] ConfBridge("DAHDI/1-1", "1") in new stack > -- <DAHDI/1-1> Playing 'conf-onlyperson.ulaw' (language 'en') > ....... > > > extensions.conf: > > [globals] > ....... > GOTO_ON_BLINDXFR="internal,266,1" > > features.conf: > > [featuremap] > blindxfer => #1 > > But: > > -- Executing [s at DialOut:14] Dial(&quot...

...;> -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1", "1") in new >>> stack >>> -- Executing [266 at internal:3] ConfBridge("DAHDI/1-1", "1") in >>> new stack >>> -- <DAHDI/1-1> Playing 'conf-onlyperson.ulaw' (language 'en') >>> ....... >>> >>> >>> extensions.conf: >>> >>> [globals] >>> ....... >>> GOTO_ON_BLINDXFR="internal,266,1" >>> >>> features.conf: >>> >>> [featur...

...etme I get some message saying: File channel.c, Line 1399 (ast_set_write_format): Unable to find a path from 2 to 1 This probably means that GSM<->G.723.1 transcoding is not supported (which is normal). However, when I try to use a voice file pre-encoded with G.723.1 codec (for example conf-onlyperson.g723) I not getting the " Unable to find a path from 2 to 1" message, but Asterisk segfaults. My guess is that asterisk is probably capable of playing/streaming files to g.723.1 endpoints if the files to be played are already encoded with g723 codec. Right? Is this feature supported fir...

...:49:53 WARNING[4555]: channel.c:1913 ast_request: No channel type registered for 'zap' 2005-06-28 16:49:53 WARNING[4555]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device ~ -- Created MeetMe conference 1023 for conference '1234' ~ -- Playing 'conf-onlyperson' (language 'en') Vr-VoIP1*CLI> Disconnected from Asterisk server Executing last minute cleanups -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.0 (GNU/Linux) iD8DBQFCwXQFJUm/Bor63CERAtE7AKCZ/IUG2IK/myBoc8iHsR7uV5PmDgCgzaTp hwQpYpTzLrp7p72beDciw+Q= =rJYD -----END PGP SIGNATURE----...

...-- Executing MeetMe("IAX2/300@300/3", "37455|APM") in new stack == Parsing '/etc/asterisk/meetme.conf': Found -- Created MeetMe conference 1023 for conference '37455' -- Playing 'conf-getpin' (language 'en') -- Playing 'conf-onlyperson' (language 'en') -- Started music on hold, class 'default', on IAX2/300@300/3 Aug 8 21:20:32 WARNING[19742]: app_meetme.c:962 conf_run: Unable to write frame to channel: Resource temporarily unavailable -- Stopped music on hold on IAX2/300@300/3 -- Hungup '...

...-- Executing MeetMe("SIP/9002-08f9feb8", "12345|p") in new stack == Parsing '/etc/asterisk/meetme.conf': Found -- Created MeetMe conference 1023 for conference '12345' -- Playing 'conf-getpin' (language 'en') -- Playing 'conf-onlyperson' (language 'en') -- Executing Playback("SIP/9001-08fb34d0", "conf-hasentered") in new stack -- Playing 'conf-hasentered' (language 'en') -- Executing Wait("SIP/9001-08fb34d0", "2") in new stack -- Executing MeetMe(...