similar to: No remote address on RTP instance - On Ringing

Hello! I'm facing completely choppy sound. The wireshark trace shows, that there are a lot of codec changes without any trigger (means no options or reinvite or any other package). Background: The call is initiated by asterisk and is received by the same asterisk conference room via Phone extension -> asterisk -> provider A -> provider B -> asterisk. Asterisk initially sends

Hello, my logs are flooded with: WARNING: The 'stasis/m:cdr:aggregator-00000005' task processor queue reached 5000 scheduled tasks again. and then, when call came, I got this: ERROR: Oh dear... we couldn't allocate a port for RTP instance '0x6e1e680fd670' WARNING: Unable to cancel schedule ID 0. This is probably a bug (res_rtp_asterisk.c: dtls_srtp_stop_timeout_timer,

On Fri, Apr 14, 2023 at 5:44 AM Fourhundred Thecat <400thecat at gmx.ch> wrote: > Hello, > > my logs are flooded with: > > WARNING: The 'stasis/m:cdr:aggregator-00000005' task processor queue > reached 5000 scheduled tasks again. > This means there's a ton of channel events being created. > > and then, when call came, I got this: > > ERROR: Oh

The Asterisk Development Team would like to announce the release of Asterisk 13.16.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.16.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 14.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.5.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

> On 2023-04-14 11:06, Joshua C. Colp wrote: > On Fri, Apr 14, 2023 at 5:44 AM Fourhundred Thecat <400thecat at gmx.ch > <mailto:400thecat at gmx.ch>> wrote: > > Hello, > > my logs are flooded with: > > WARNING: The 'stasis/m:cdr:aggregator-00000005' task processor queue > reached 5000 scheduled tasks again. > > > This

Hello ! My asterisk log is full of messages like this: [2011-03-06 19:01:15] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit. [2011-03-06 19:01:20] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit. [2011-03-06 19:01:25] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit. [2011-03-06

Hi list, I've always about 50 concurrent SIP callers listening to several MOH streams (fed via mpg123) on Asterisk 11.4.0, 4x 2.2 GHz and the CPU usage is always at about 250% causing stuttering of the streams and delays when using Read() and Playback() (overall everything is just really slow/delayed). I've seen Asterisk doing that when the MOH stream that was fed via mpg123 was

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

Hi Joshua, Thank you for that. From the code it kind of looks like STRICT_RTP_LEARN_TIMEOUT is a minimum, not a maximum: if (!ast_sockaddr_isnull(&rtp->strict_rtp_address) && STRICT_RTP_LEARN_TIMEOUT < ast_tvdiff_ms(ast_tvnow(), rtp->rtp_source_learn.start)) { ast_verb(4, "%p -- Strict RTP learning complete - Locking on source address %s\n", Our call shows: #

Hello Everyone, We are looking for solutions where the transcoding is abstracted away from our * box (i.e., to the network layer) using some carrier grade gateway, or router. The reason for my post is to know about solutions people have used in the past, and how it fits into their overall architecture. Our transcoding needs consists mainly of u/alaw <-> g729, and gsm would also be good....

I don't know in that specific output what happened. Your best course of action is to add further logging or step through the logic with all of the knowledge you have of the RTP streams to understand what is happening. On Mon, Apr 17, 2023 at 8:52 PM David Cunningham <dcunningham at voisonics.com> wrote: > Hi Joshua, > > Thank you for that. From the code it kind of looks like

hi guys, i have a problem with 1.8 branch no matter which release of 1.8 i'm using. i can't make any sip calls, this is the error message i get on each call: [Jan 18 19:02:15] ERROR[1698] rtp_engine.c: No RTP engine was found. Do you have one loaded? [Jan 18 19:02:15] ERROR[1698] chan_sip.c: Got SDP but have no RTP session allocated. i'm sure that the rtp engine is loaded this is the

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

The Asterisk Development Team would like to announce the release of Asterisk 17.9.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.9.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 16.9.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.9.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 16.9.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.9.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 17.3.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 17.3.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 16.15.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.15.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: