search for: colp

Hi, I would like to test the following COLP use case : Alice and Bob are both using a SIP phone registered on a Asterisk 10 server. Alice dials Bob's extension. While Bob's phone is ringing, Asterisk updates Alice phone screen with Bob's name, so that at a glance, Alice can check she dialed the correct number. Before diving int...

...n/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny (Reported by Matt Jordan) * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when ICE is not enabled (Reported by Joshua Colp) * ASTERISK-25383 - Core dumps on startup and shutdown with MALLOC_DEBUG enabled (Reported by yaron nahum) * ASTERISK-25423 - Caller gets no Connected line update during call pickup. (Reported by Richard Mudgett) * ASTERISK-25305 - Dynamic logger channels can be added multiple...

Hi, I've searched the asterisk.org and voip-info wiki sites, but not found an answer that seems to match. Hopefully this is a simple question. COLP is working very well on our system - Unfortunately it is working a bit TOO well in some circumstances. We have some "untrusted" trunks. On these trunks, an initial CallerID can be used, but any redirected caller numbers, COLP updates etc are not safe to accept. Sadly I cannot find how to...

...neprov: memory leak and heap-use-after-free (Reported by Badalian Vyacheslav) * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes returns garbage (Reported by Etienne Lessard) * ASTERISK-25751 - res_pjsip: Support pjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp) * ASTERISK-25606 - Core dump when using transports in sorcery (Reported by Martin Mou??ka) * ASTERISK-20987 - non-admin users, who join muted conference are not being muted (Reported by hristo) * ASTERISK-25737 - res_pjsip_outbound_registration: line option not in Alembic (Rep...

...message causes asterisk to crash (Reported by Ross Beer) * ASTERISK-27103 - core: ast_safe_system command injection possible. (Reported by Corey Farrell) * ASTERISK-27013 - res_rtp_asterisk: Media can be hijacked even with strict RTP enabled (Reported by Joshua Colp) * ASTERISK-27231 - res_rtp_asterisk: Allow remote SSRC to change due to renegotiation (Reported by Joshua Colp) * ASTERISK-27222 - core: Don't queue up multiple video update frames. (Reported by Joshua Colp) * ASTERISK-27212 - bridge_softmix: Quickly joining/leaving...

...ERISK-26220 - Add support for noreturn function attributes. (Reported by Corey Farrell) Bugs fixed in this release: ----------------------------------- * ASTERISK-27143 - bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues. (Reported by Joshua Colp) * ASTERISK-27142 - sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5 (Reported by Corey Farrell) * ASTERISK-25810 - say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say......

...ith PlayDTMF instead of only "sending" (Reported by lvl) * ASTERISK-28613 - func_curl: CURLOPT cannot set Content-Type header (Reported by Martin Tomec) * ASTERISK-28533 - func_jitterbuffer: Add support for video synchronization (Reported by Joshua C. Colp) * ASTERISK-17808 - [patch] Unregister a realtime moh class (Reported by Byron Clark) * ASTERISK-28489 - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI domain (Reported by Stas Kobzar) Bugs fixed in this release: ------------------------------...

On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Tue, Feb 28, 2023 at 9:50 AM David Cunningham < > dcunningham at voisonics.com> wrote: > >> Hello, >> >> Does anyone know if one of the "strictrtp" options disables RTP learning? >> As far as I can tell from...

...orks on 16.5 (Reported by Niklas Larsson) * ASTERISK-28521 - pjsip: Memory Leak (Reported by Mark) * ASTERISK-28523 - Asterisk 16.5.0 Memory leak (Reported by Cyril Rami��re) * ASTERISK-28538 - chan_pjsip: Deadlock on fax detection (Reported by Joshua C. Colp) * ASTERISK-28536 - Asterisk release candidates fail to build on FreeBSD (Reported by Guido Falsi) * ASTERISK-28499 - translate: Crash when frame does not have a "src" field set (Reported by Gregory Massel) * ASTERISK-28488 - pjsip mwi: n+1 sip notify's sent...

...ing after remote > address set" should reset the rtp_source_learn.start timestamp, and yet the > "Strict RTP learning complete" messages are less than 5000ms after that. > What could be happening? > > Thanks again. > > > On Tue, 18 Apr 2023 at 10:40, Joshua C. Colp <jcolp at sangoma.com> wrote: > >> It's probably best if you read the logic[1]. There's an entire comment >> that talks about how it works. >> >> [1] >> https://github.com/asterisk/asterisk/blob/20/res/res_rtp_asterisk.c#L8158 >> >> On Mon,...

Hello, (sorry for not continuing the thread, I had set the list to digest). Would UnicastRTP be able to output u-law frames directly? If so, I think that is all I need. Does anyone know what the EAGI output is? Raw RTP? Best regards, Luca -------------- next part -------------- An HTML attachment was scrubbed... URL:

On 30/09/2020 14:59, Joshua C. Colp wrote: > latest version of 16 on Ubuntu 16.12.0~dfsg-1 ? -- sergio.

Hi, I would just like to know if any work was ever done on COLP or its related cousins? The last evidence of it seems to be about 2 years old when K.Flemming and Olle both showed some mild interest. I am not sure how well that code would apply to today's Asterisk. (I realise that this is sort of a duplicate posting, sorry about that.) Thanks for any feedb...

Hi, Does asterisk version 1.6.2.12 support COLP (COnnected Line Presentation) Regards Arjan Kroon Mobillion BV -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110330/937ec861/attachment.htm>

On 11/14/17 4:27 PM, Joshua Colp wrote: > On Tue, Nov 14, 2017, at 06:25 PM, Carlos Chavez wrote: >> On 11/14/17 3:55 PM, Joshua Colp wrote: >> >>> On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: >>>> I followed the blog post and I can get video from the conference if >>&gt...

...ith PlayDTMF instead of only "sending" (Reported by lvl) * ASTERISK-28613 - func_curl: CURLOPT cannot set Content-Type header (Reported by Martin Tomec) * ASTERISK-28533 - func_jitterbuffer: Add support for video synchronization (Reported by Joshua C. Colp) * ASTERISK-17808 - [patch] Unregister a realtime moh class (Reported by Byron Clark) * ASTERISK-28489 - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI domain (Reported by Stas Kobzar) * ASTERISK-28375 - res_pjsip: New configuration setting to...

...ith PlayDTMF instead of only "sending" (Reported by lvl) * ASTERISK-28613 - func_curl: CURLOPT cannot set Content-Type header (Reported by Martin Tomec) * ASTERISK-28533 - func_jitterbuffer: Add support for video synchronization (Reported by Joshua C. Colp) * ASTERISK-17808 - [patch] Unregister a realtime moh class (Reported by Byron Clark) * ASTERISK-28489 - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI domain (Reported by Stas Kobzar) * ASTERISK-28375 - res_pjsip: New configuration setting to...

...s_pjsip: Change default "From user" from UUID to something more palatable (Reported by Mark Michelson) * ASTERISK-25252 - ARI: Add the ability to manipulate log channels (Reported by Matt Jordan) * ASTERISK-25259 - chan_pjsip: Add rtptimeout support (Reported by Joshua Colp) * ASTERISK-25238 - ARI: Support push configuration (Reported by Matt Jordan) * ASTERISK-25173 - ARI: Add the ability to load/reload/unload an Asterisk module (Reported by Matt Jordan) * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation (Reported by Dwayne Hubbar...

...use the second "Strict RTP learning after remote address set" should reset the rtp_source_learn.start timestamp, and yet the "Strict RTP learning complete" messages are less than 5000ms after that. What could be happening? Thanks again. On Tue, 18 Apr 2023 at 10:40, Joshua C. Colp <jcolp at sangoma.com> wrote: > It's probably best if you read the logic[1]. There's an entire comment > that talks about how it works. > > [1] > https://github.com/asterisk/asterisk/blob/20/res/res_rtp_asterisk.c#L8158 > > On Mon, Apr 17, 2023 at 7:10 PM David C...

Le lun. 26 août 2019 à 12:07, Joshua C. Colp <jcolp at digium.com> a écrit : > ... > > libpjnath is the ICE/STUN/TURN library which is used by res_rtp_asterisk > for that functionality. If you're using WebRTC or ICE/STUN/TURN, then you > would be using that library. > Yes, I'm using ICE/STUN/TURN. That explai...