search for: scgm11

Hi, is it possible to detect 4 length pattern busy cadence detection on FXO lines in 1.8?? Here the tones are: 425Hz Pattern(0.2ms on, 0.2ms off, 0.2ms on, 0.6ms off) in asterisk 1.4 busy detect worked in asterisk 1.6 didn?t work and i was told that 1.6 can?t handle 4 length patterns, but what about 1.8?? for now I can only hangup by asking the provider polarity switch. Thanks best

>From this change (res_rtp_asterisk): ast 13.10 to 13.11 webrtc JSSIP stop working, failing with chan_sip.c:4083 retrans_pkt: Hanging up call 7238b48c11581d4166b899bf747a05f7 at 130.211.62.184:0 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). is there any way to configure to have the previous behaviour? Im trying to set

...------- * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported by snuffy) * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi (Reported by Morten Tryfoss) * ASTERISK-26520 - codec_opus: Generated fmtp line has no content (Reported by scgm11) * ASTERISK-26605 - codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded. (Reported by Richard Mudgett) * ASTERISK-26516 - pjsip: Memory corruption with possible memory leak. (Reported by Richard Mudgett) * ASTERISK-26592 - Latest libedit (3.1) defaults to...

...le? I've never compiled C# code before, and although a quick google suggests it shouldn't be too hard, I might need to know a few things like what version of .net it should be compiled with. The readme just points to the website. Thanks! On 18 March 2017 at 18:57, Sebastian Gutierrez <scgm11 at gmail.com> wrote: > Check this one: > > https://github.com/IntegraCCS/integradesigner > > You can do many things, document each node, and save xml with each > extension. > We?ve made it open source on Astricon 2015 you can extend it the way you > want. > > Hope i...

...e not been possible without your participation. Thank you! The following are the issues resolved in this release: Improvements made in this release: ----------------------------------- * ASTERISK-26558 - app_queue: add variable to know if the call is not answered after a queue (Reported by scgm11) * ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry (Reported by scgm11) * ASTERISK-26538 - codec_opus: Add sample to configs/samples/codecs.conf.sample (Reported by Kevin Harwell) * ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and '...

...eters (Reported by Joel Vandal) * ASTERISK-26890 - STUN server with non-default-route transport causes INVITE delay (Reported by George Joseph) * ASTERISK-26692 - res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip) (Reported by scgm11) * ASTERISK-26835 - res_rtp_asterisk: Crash when freeing RTCP address string (Reported by Niklas Larsson) * ASTERISK-26853 - res_rtp_asterisk: Crash in pjnath when receiving packet (Reported by Adagio) * ASTERISK-26613 - format_wav: wav16 format read file only by 32...

...eters (Reported by Joel Vandal) * ASTERISK-26890 - STUN server with non-default-route transport causes INVITE delay (Reported by George Joseph) * ASTERISK-26692 - res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip) (Reported by scgm11) * ASTERISK-26835 - res_rtp_asterisk: Crash when freeing RTCP address string (Reported by Niklas Larsson) * ASTERISK-26853 - res_rtp_asterisk: Crash in pjnath when receiving packet (Reported by Adagio) * ASTERISK-26613 - format_wav: wav16 format read file only by 32...

...26292 - app_confbridge: 3D-Conferencing via Binaural Synthesis (Reported by Dennis Guse) * ASTERISK-23828 - pjsip - Need a command to list active SIP subscriptions (Reported by Rusty Newton) * ASTERISK-26559 - app_queue: New service level calculation (Reported by scgm11) * ASTERISK-26658 - Add ability for dialplan show to display filenames/line numbers of registered extensions (Reported by Jonathan R. Rose) * ASTERISK-26527 - Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec (Reported by Badalian...

How are we all documenting complex dialplan? Is there something similar to Doxygen? I've got around 20 config files covering around 60 contexts and 40 variables. Of course, I've maintained a basic list of the major stuff, and documented the code throughout, but it's grown to the stage where it needs to be better documented, have a proper flowchart etc. Talking of flowcharts, I see

...de in this release: ----------------------------------- * ASTERISK-25063 - [patch]add X.509 subject alternative name support to Asterisk TLS support (Reported by Maciej Szmigiero) * ASTERISK-26558 - app_queue: add variable to know if the call is not answered after a queue (Reported by scgm11) * ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry (Reported by scgm11) * ASTERISK-26538 - codec_opus: Add sample to configs/samples/codecs.conf.sample (Reported by Kevin Harwell) * ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and '...

Same about me - need to re-enable membership all the time. Annoying (( ??, 12 ???. 2017 ?. ? 15:59, John Novack <jnovack at comcast.net>: > Not just gmail > Happening as well with Comcast.net > > My Comcast address is set to forward to another domain, as Comcast seems > to now block sending mail with a non Comcast "from" address. they turned > that on a couple

Hi, I'm noticing MixMonitor records 5 seconds aprox less of a call. The recording is iniciated via Queue and ends at the hungup. (gsm format), when I listen to the audio file, has 5 seconds missing at the end of the call. Any idea?? thanks ASt.1.6.0.1 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm noticing MixMonitor records 5 seconds aprox less of a call. The recording is iniciated via Queue and ends at the hungup. (gsm format), when I listen to the audio file, has 5 seconds missing at the end of the call. Any idea?? thanks ASt.1.6.0.1 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm noticing MixMonitor records 5 seconds aprox less of a call. The recording is iniciated via Queue and ends at the hungup. (gsm format), when I listen to the audio file, has 5 seconds missing at the end of the call. Any idea?? thanks ASt.1.6.0.1 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm having troubles using music on hold with realtime (ast 1.6.0.1). Everything seems ok, but no clases are show. If I try to make a cal nothing is played and says theres no moh class. Musiconhold.conf [general] cachertclasses=yes ; use 1 instance of moh class for all users who are using it, ; decrease consumable cpu cycles and memory

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Hi, Im trying to do this: 1) Originate a call between an external number and a ivr that do some things in background 2) after the originate I bridge the person that dial that extent with the external number I would like to have the ivr in background while the bridge is up for monitoring porpoises, but seems to stop processing when the local bridge is done other possibility could be

The 13.14 tar gz doesn?t even exists on the current or in the old releases folder. there seems to be an issue with the latest build not generating the artifacts? best regards On Feb 14, 2017, 11:04 -0300, Marcelo Terres <mhterres at gmail.com>, wrote: > Thanks Joshua. > Marcelo H. Terres <mhterres at gmail.com > IM: mhterres at jabber.mundoopensource.com.br >

Hi, I'm having problems with a new ISDN PRI in a new server. The cable is connected and the E1 modem seems to have issues with syncing (blinking light on the modem). versions: CentOS 6, asterisk 1.6.2.20, dahdi 2.5.0.1, libpri 1.4.12 ------------------------------------------------------------------------------------ dahdi show status T4XXP (PCI) Card 0 Span 1 RED 0