search for: prescom

...obvious answer but maybe I'm misunderstanding the > question. > > exten => s,1,Dial(SIP/alice,20) > same => n,Playback(myannouncement) > same => n,NoOP(Whatever else you want to do goes here) > > On Mon, Aug 22, 2016 at 8:36 AM, Jean Aunis <jean.aunis at prescom.fr > <mailto:jean.aunis at prescom.fr>> wrote: > > Hello, > > I am searching a way to dial a SIP peer, and if it does not answer > within 20 seconds, play an announcement to the caller. This means > that the caller would hear a ring tone for 20 seconds...

...ll display a missed call. I would prefer to do everything in a single call. Le 22/08/2016 ? 17:57, John Kiniston a ?crit : > You could try using RetryDial() instead of Dial, It supports playing > an announcement. > > > On Mon, Aug 22, 2016 at 8:45 AM, Jean Aunis <jean.aunis at prescom.fr > <mailto:jean.aunis at prescom.fr>> wrote: > > Sorry, I forgot to write that the SIP peer must keep ringing while > the announcement is being played. > > > Le 22/08/2016 ? 17:42, John Kiniston a ?crit : >> This seems like the obvious answer b...

...striCon this year - www.astricon.net On 23 August 2016 at 12:52, Israel Gottlieb <isrlgb at gmail.com> wrote: > You could m and make a moh file that has ringing the first 30 sec and then > the anouncment > > ?????? 22 ????? 2016 7:19 PM,? "Jean Aunis" <jean.aunis at prescom.fr> ???: > > Thank you for the idea. The problem with RetryDial, is that it will cancel >> the first call, play the announce and then dial the SIP peer once again, so >> the telephone will display a missed call. I would prefer to do everything >> in a single call. >&gt...

Maybe try progress() instead of answer () ?????? 23 ????? 2016 7:19 PM,? "Jean Aunis" <jean.aunis at prescom.fr> ???: > Thank you, I just tried your suggestion. Strangely, the announcement is > played only if I try to dial a SIP peer which is not available (not > registered to be more precise). If the SIP peer is available, I only get > the ring tone, and never hear the announcement. Here...

On Thu, Jan 30, 2020 at 12:18 AM Jean Aunis <jean.aunis at prescom.fr> wrote: > Hello, > > I use UserEvents generated by the Message/ast_message_queue channel with > the UserEvent application. > > Regards > > Jean > Thanks Jean. We're looking at alternatives. > Le 29/01/2020 à 20:31, George Joseph a écrit : > > For...

On Thu, Jan 30, 2020 at 3:18 AM Jean Aunis <jean.aunis at prescom.fr> wrote: > Hello, > > I use UserEvents generated by the Message/ast_message_queue channel with > the UserEvent application. > Do you use any aspects of the channel itself in the user events, or merely the contents of the user event and what you've placed in it? -- Joshua...

Hello, I am searching a way to dial a SIP peer, and if it does not answer within 20 seconds, play an announcement to the caller. This means that the caller would hear a ring tone for 20 seconds, and only then hear the announcement if the callee did not answer. I know it is possible to do this with ARI, but in this particular case I do not want to use ARI. I would like to do this purely with

Hello, Has anyone already implemented some sort of call preemption in Asterisk ? I am trying to achieve something like this : - I want to limit the number of calls on a given SIP peer to 10 - on the other hand, some calls have higher priority than others - when the ceiling of 10 calls is reached and a call with a high priority is attempted, I would like to drop a call with a lower priority

Hello, I think there is an issue when DTMF are handled with SIP INFO and direct media is enabled. When I receive a SIP INFO, the logs tell me that a "DTMF begin" is generated, but no related "DTMF end" is generated, unless the call is ended. Here is an excerpt of the logs : *--- SIP INFO received **on **SIP/xxx-00000004:* [Dec 13 11:56:16] DTMF[18193][C-00000005]

On 2020-01-30 10:30 a.m., George Joseph wrote: > > > On Thu, Jan 30, 2020 at 12:18 AM Jean Aunis <jean.aunis at prescom.fr > <mailto:jean.aunis at prescom.fr>> wrote: > > Hello, > > I use UserEvents generated by the Message/ast_message_queue > channel with the UserEvent application. > > Regards > > Jean > > > Thanks Jean.  We're looking at alte...

For those of you who actually process SIP MESSAGE requests... Do you use any of the AMI events generated by the "Message/ast_msg_queue" channel? We want to change that channel to an "internal" channel that doesn't generate AMI events (for performance reasons) but we need to know if anyone's using them first. Thanks! -- George Joseph Asterisk Software Developer

...: ----------------------------------- * ASTERISK-29013 - res_pjsip: Asterisk doesn't stop sending invites (with auth) on 407 replies (Reported by Sebastian Damm) * ASTERISK-29108 - resource_endpoints.c : Memory leak if endpoint not found (Reported by Jean Aunis - Prescom) * ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing string when failing to add extension (Reported by Vieri) * ASTERISK-26424 - app_voicemail: Undocumented behavior from VMSayName (Reported by Eric Smith) * ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set corr...

...Proceeding() (Reported by lvl) * ASTERISK-28685 - check_expr2: linking (when hardening) and cross-compiling troubles (Reported by Sebastian Kemper) * ASTERISK-28755 - SIP/Stasis: SIP headers not transmitted in the "variables" field (Reported by Jean Aunis - Prescom) * ASTERISK-28754 - ASTERISK-28738 Causes Audio Issue After Hold (Reported by Ross Beer) * ASTERISK-28716 - ICE: pjnath shouldn't wait for ICE to complete before allowing sending (Reported by Benjamin Keith Ford) * ASTERISK-28697 - res_pjsip: Named ACL does not...

...Proceeding() (Reported by lvl) * ASTERISK-28685 - check_expr2: linking (when hardening) and cross-compiling troubles (Reported by Sebastian Kemper) * ASTERISK-28755 - SIP/Stasis: SIP headers not transmitted in the "variables" field (Reported by Jean Aunis - Prescom) * ASTERISK-28754 - ASTERISK-28738 Causes Audio Issue After Hold (Reported by Ross Beer) * ASTERISK-28716 - ICE: pjnath shouldn't wait for ICE to complete before allowing sending (Reported by Benjamin Keith Ford) * ASTERISK-28697 - res_pjsip: Named ACL does not...

...missing comment mark on line 115 (Reported by Andrew Siplas) * ASTERISK-28430 - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF (Reported by under) * ASTERISK-29108 - resource_endpoints.c : Memory leak if endpoint not found (Reported by Jean Aunis - Prescom) * ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing string when failing to add extension (Reported by Vieri) * ASTERISK-26424 - app_voicemail: Undocumented behavior from VMSayName (Reported by Eric Smith) * ASTERISK-29091 - Crash when ast_translator_build_path...

...using Proceeding() (Reported by lvl) * ASTERISK-28764 - res_rtp_asterisk: Improve NACK support and seqno handling (Reported by Joshua C. Colp) * ASTERISK-28755 - SIP/Stasis: SIP headers not transmitted in the "variables" field (Reported by Jean Aunis - Prescom) * ASTERISK-28685 - check_expr2: linking (when hardening) and cross-compiling troubles (Reported by Sebastian Kemper) * ASTERISK-28754 - ASTERISK-28738 Causes Audio Issue After Hold (Reported by Ross Beer) * ASTERISK-28697 - res_pjsip: Named ACL does not update on r...

Hi Is it a normal behavior of Asterisk put a call on hold when receive a Session Progress with media address 0.0.0.0 in SDP? I believe the call on hold should be initiate with a re-invite. Thanks -- Att, Rafael Saraiva -------------- next part -------------- An HTML attachment was scrubbed... URL:

...g troubles (Reported by Sebastian Kemper) * ASTERISK-28764 - res_rtp_asterisk: Improve NACK support and seqno handling (Reported by Joshua C. Colp) * ASTERISK-28755 - SIP/Stasis: SIP headers not transmitted in the "variables" field (Reported by Jean Aunis - Prescom) * ASTERISK-28754 - ASTERISK-28738 Causes Audio Issue After Hold (Reported by Ross Beer) * ASTERISK-28697 - res_pjsip: Named ACL does not update on reload if changed (Reported by Timothy Vanderaerden) * ASTERISK-28746 - res_pjsip_outbound_registration keeps retrying...

...g troubles (Reported by Sebastian Kemper) * ASTERISK-28764 - res_rtp_asterisk: Improve NACK support and seqno handling (Reported by Joshua C. Colp) * ASTERISK-28755 - SIP/Stasis: SIP headers not transmitted in the "variables" field (Reported by Jean Aunis - Prescom) * ASTERISK-28754 - ASTERISK-28738 Causes Audio Issue After Hold (Reported by Ross Beer) * ASTERISK-28697 - res_pjsip: Named ACL does not update on reload if changed (Reported by Timothy Vanderaerden) * ASTERISK-28746 - res_pjsip_outbound_registration keeps retrying...

...g troubles (Reported by Sebastian Kemper) * ASTERISK-28764 - res_rtp_asterisk: Improve NACK support and seqno handling (Reported by Joshua C. Colp) * ASTERISK-28755 - SIP/Stasis: SIP headers not transmitted in the "variables" field (Reported by Jean Aunis - Prescom) * ASTERISK-28754 - ASTERISK-28738 Causes Audio Issue After Hold (Reported by Ross Beer) * ASTERISK-28697 - res_pjsip: Named ACL does not update on reload if changed (Reported by Timothy Vanderaerden) * ASTERISK-28746 - res_pjsip_outbound_registration keeps retrying...