similar to: Reply to INVITE with 1 codec

Using Asterisk 18.12.0, a little confused on how to configure my pjsip.conf file to determine the codec to use for a call I have 2 endpoints: [Alice] disallow:all allow:ulaw,alaw,g729 [Bob] disallow:all allow:ulaw,alaw,g729 Alice calls into Asterisk on ext 100 and then we dial Bob I want to wait until Bod side codec is chosen to answer Alice and have each channel use the codec chose on Bob

Hi, I followed the TLS/SRTP tutorial on the wiki [0] using Asterisk 11.8.1 on CentOS 6.5 x86_64 and CSipSimple on a Nexus with Android 4.4.x local wifi. The phone seems to register but directly after that things fall apart (turning SELinux off made no difference): *CLI> -- Registered SIP 'encrypted' at 10.0.0.137:58079 > Saved useragent

So the call used Alaw as Codec. > Am 13.06.2020 um 17:23 schrieb Luca Bertoncello <lucabert at lucabert.de>: > > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > Hi > >> Try "sip show peer <peername>" for a phone. > > So: > > mobile phone: > bpi*CLI> sip show peer 0049177xxxxxxx > > > > > * Name :

Dear all, I'm trying to use ulaw and alaw with Diax and Asterisk but I'm not able to, I got the following error message: Mar 20 11:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected connect attempt from 192.168.0.55, requested/capability 0x8/0xc incompatible with our capability 0xfe02. I do not understand why because my Asterisk box load these codecs properly! Does somebody

Starting with Asterisk 13.1 we are seeing this WARNING messages a lot in our logs and console: WARNING[25164][C-0004865e]: chan_sip.c:7364 sip_write: Can't send 10 type frames with SIP write) We found that line in function "sip_write" inside "chan_sip.c". In our previous version (11.2.1) we did not see those messages being printed (same verbosity level). We compared

Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon

Running build 1.8.5.0 (compiled from source) I seem to be having an issue with codec negotiation. I have a Grandstream HT503 FXO port connected to a pstn line, a Polycom SP501, and a SIP trunk with callwithus. What I'm essentially looking to accomplish is for ulaw or g729 (preferably ulaw) to be used to the Grandstream FXO or any other internal endpoint, and for g729 only to be used outbound

Hello, After I have re-read the "PJSIP Advanced Codec negotiation" document, it occurred to me that the desired behavior should actually happen automatically, just due to the codec negotiation logic, but it looks like asterisk doesn't actually follow the described logic which is likely a bug. Can you please follow with me through a simple sip call and see if I'm missing

Hello people, I've ran into two problem that I can't seem to be able to solve on my own. Here's my scenario (running Asterisk 13.28.1): In short: - Asterisk behaves unexpectedly (at least to me) when negotiating between endpoints             that have a different but intersecting set of codecs (preventing direct media flow).           - Also, when an endpoint sends RTP with an

I am unable to get re-invite to work on a new system. Also, unwanted transcoding is occurring on PSTN calls. The new system (FreePBX 2.11.0.37, Asterisk 11.9.0, CentOS 6.5) will eventually replace an old system (FreePBX 2.8.1, Asterisk 1.8.7.2, CentOS 5.8) currently in production. Both systems are on VPS with public IP addresses. Goals for the new system include: HD (g722) connections on

Hi, at my Asterisk box, I have a few of IAX2 phones (configured with alaw/ulaw/gsm codecs, in this order) and a PRI E1 line. In iax.conf I hav: disallow=all allow=alaw allow=ulaw allow=gsm During some incoming call, I read at console: -- Executing Dial("Zap/2-1", "IAX2/215|20|TtwW") in new stack -- Called 215 -- Call accepted by 10.97.1.7 (format ulaw) --

Hi, I'm a bit confused about how Asterisk decides in which order of preference it should list the different codecs in its SDP message during SIP call setup. In my sip.conf [general] section I've got disallow=all allow=gsm allow=ulaw allow=alaw But when Asterisk bridges a call from an E1 to VoIP it sends out an INVITE with the codecs listed in the following order of preference

Steve, My Problem is not a problem, with the codec negotiation between end points. But when asterisk does it with canreinvite=no, * do not do it right. I replied with a lengthy discussion about my findings here, This behavior can be reproduced. But '*' do not seem to do the negotiation correctly. http://lists.digium.com/pipermail/asterisk-users/2004-January/032197.html

Hi, I am taking over an asterisk system from another person and having an issue where a sip trunk is restricting the outgoing codecs to just g.729 I am dialing in from a Cisco 7960. The Invite from the Cisco has the folowing M line: m=audio 17022 RTP/AVP 18 0 8 101. So it is allowing g.729, ulaw and alaw. Asterisk is tandeming the call out over a SIP trunk sip.conf tandem trunk config:

How can I set the prefered codec between 2 calling parties ?? My Grandstream supports G729, alaw and gsm... in this order. The Zoiper softphone has alaw and gsm as codecs... in that order. Although there should be a matching codec found, my Grandstream can not call the Zoiper softphone. CLI shows : [Mar 11 17:47:21] WARNING[22367]: channel.c:3340 ast_channel_make_compatible: No path to

Hello, I have been trying to get my coders to work without a conversion. I have read all the available asterisk documentation and support groups without any luck. Here is my issue. (Please feel free to ask questions if you do not understand what I am talking about.) I am using Cisco ATA-186 set to g729 codec. (But it will switch to g711 if sip-server request g711) I have 2 SIP-services to

Hi: I have a mgcp.conf and a mgcp_additional.conf which records the special information about the extensions. And i found if i use ulaw in the general context in mgcp.conf,then all the registered extensions can make both outbound and inbound calls,the mgcp.conf is following: [general] port = 2727 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw ; can be disable and do no effect

I'm using "insecure=invite" with two different dial plans, it it working with one dial plan but not with the other. What other parameters could influence "insecure=invite" In sip.conf below "insecure=invite" is working OK [pstn-1270] type=friend secret=spa3k username=voice-1270 mailbox=369 host=dynamic insecure=invite canreinvite=no disallow=all allow=ulaw

Ok - updating the firmware on teh device - factory reset, re-config. Capabilities: us - (g726|slin16|ulaw|alaw|gsm), peer - audio=(ulaw|alaw|g726|slin16)/video=(nothing)/text=(nothing), combined - (g726|slin16|ulaw|alaw) Looking much better. Jerry On Wed, Jun 17, 2020 at 4:01 PM Jerry Geis <jerry.geis at gmail.com> wrote: > I thought - what about the software - maybe it needs updated.