similar to: Auto video call hangup

Howdy, I recently saw something strange with a call between *'s over IAX2. There are actually 3 *'s involved. The setup is like this: SIP phone ------(ulaw over LAN)------ *1 -------- IAX2 (ulaw over Internet) ---------*2--------(GSM over Internet) -----------*3--------(ulaw over LAN)------ SIP phone Now what is shown below is the Asterisk in the middle, that is doing the

Hi, I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to connect to a 1.2.5 box for PSTN. There are 15 users on the remote server, all connecting via SIP softphones. For some reason, there is an increasing number of calls where the callee does not get any audio although the caller can hear them perfectly. This happens between 5% and 10% of the time. If they hang up and call

Some more info: Just tried this on a server without using any NAT and no port forwarding, no masquerading, and I still have the same problem. So there goes that idea. I do not know what this VNAK error means. By the way, I am using the latest version (1.2.6) of asterisk, have also tried other versions with the same problem [1.0.9 (Ubuntu Breezy) and 1.0.7 (Debian Sarge) and 1.2.1 (Ubuntu

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, We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial invite from our upstream provider offers both g711 and g729. Asterisk presumably simply forwards the media from iax2 trunk encapsulation to sip

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

I am trying to enable full WebRTC support on asterisk-11.15 for installation on a CentOS 5 machine. Currently the distro cannot be upgraded to any later CentOS series. This CentOS series ships with openssl-0.9.8e, which lacks DTLS-SRTP support required for WebRTC. So I decided to build a parallel install of openssl. I chose the Fedora 21 package, openssl-1.0.1j, and built it on CentOS 5. The

Hello, 2 asterisk servers 16.8.0 version running on Debian 10.3 On one of them, I can't compile asterisk having error    [CC] res_rtp_asterisk.c -> res_rtp_asterisk.o res_rtp_asterisk.c:2674:3: error: ‘pj_ice_sess_cb’ {aka ‘struct pj_ice_sess_cb’} has no member named ‘on_valid_pair’   .on_valid_pair = ast_rtp_on_valid_pair,    ^~~~~~~~~~~~~ res_rtp_asterisk.c:2674:19: warning:

Hi, For years I've been running a minimal (ish) SIP based Asterisk with the modules based on chan-sip. For various reasons unrelated to Asterisk the machine the latest incarnation of this configuration has been updated to Debian Buster and thus to Asterisk 16. Since this upgrade I have a dependency problem related to res_rtp_asterisk.so. So the old config was: [modules] autoload=no load

Hi I'm trying to get the rebuilt parking functionality to work in Asterisk 12.0.0. In Asterisk 11.6.0 I managed to get a call to get parked by adding a dynamic feature in features.conf for the DMTF sequence *# which called a macro in extensions.conf, which then runned the ParkAndAnnounce application, and the call got parked. The syntax for ParkAndAnnounce I used was this (I don't

Le 13/03/2020 à 13:30, Joshua C. Colp a écrit : > On Fri, Mar 13, 2020 at 9:27 AM Administrator <admin at tootai.net > <mailto:admin at tootai.net>> wrote: > > Hello, > > 2 asterisk servers 16.8.0 version running on Debian 10.3 On one of > them, > I can't compile asterisk having error > >     [CC] res_rtp_asterisk.c ->

Hi all, I am trying to connect to a softphone application using an Iax channel on Asterisk 1.4.30. I can do outbound calls, from softphone to asterisk, but not inbound from asterisk to softphone. I get the following Debug: ---------------------------------------------------------------------- ---------------------------------------------------------------------- Tx-Frame Retry[000] -- OSeqno:

Hi, compiling the current git version on Centos 7 gives me: [CC] res_statsd.c -> res_statsd.o res_rtp_asterisk.c:2669:2: error: unknown field ‘on_valid_pair’ specified in initializer .on_valid_pair = ast_rtp_on_valid_pair, ^ res_rtp_asterisk.c:2669:2: warning: initialization from incompatible pointer type [enabled by default] res_rtp_asterisk.c:2669:2: warning: (near initialization

I have been using 11.4.0 for some time. All was fine. I downloaded 11.5, extracted, run ./configure, make, make install. I got a message about res_rtp_asterisk.so was not compiled in the 11.5 Sure enough I have rss_rtp_asterisk.c but not .o file and no .so file. I then looked in the config.log and nothing is in there about res_rtp_asterisk What's up? jerry -------------- next part

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: #

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

We use Asterisk 14 to proxy calls between two servers, 10.0.0.192 to 10.0.0.228. But sometimes another of our servers becomes listed as a SIP agent, even though the server's IP address isn't part of our sip.conf, extensions.conf, nor any other config I know of. For example in the log snippet below, the source server experienced an SDP renegotiation in the middle of a call, and seemingly as

Hello, I'm experiencing some problems with running IAX2 protocol on quite reliable link with G729A codec. My customer has 2mb FR link to the Internet used in about 20%. Ping statistics: 50 packets transmitted, 49 received, 2% packet loss, time 49496ms rtt min/avg/max/mdev = 9.308/13.126/33.307/4.851 ms Everything would be great, but the quality isn't good enough. I have 2mb/512kb DSL

Hello Asterisk list, I've facing a memory allocation issue that happens occasionally but on a consistent basis. The problem happens as follow, suddenly Asterisk starts consuming a lot of memory, in a rate of more than 1GB per hour. Kernel will eventually kill it via the OOM killer when memory is really exausted... This situation does not generate backtrace because Asterisk is responsive

Can anyone tell me why I keep getting these messages from IAXTEL? It does appear to register since I get lines like this: 2005-04-30 04:26:42 VERBOSE[1644]: -- Registered to '69.73.19.178', who sees us as 67.182.152.242:4569 But what is this? I don't think IAXTEL is working for me, since I can't dial 800 #s through it when I copy the iaxtel.com instructions. 2005-05-20