similar to: SIP - how does * decide codec order of preference

Hi, I set a preference order of the codecs to my sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls of not registered phones disallow = all allow = g729 allow = g723 allow = alaw allow = ulaw Connected a 'Sipura SPA' sip phone to asterisk with g729 as its preferred codec. Problem: asterisk cannot make

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

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

Hi, We're holding ourselves back from moving to PJSIP as we don't appear to have figured out how to force codec preference in a dial plan. The 'PJSIP Advanced Codec Negotiation' document (https://wiki.asterisk.org/wiki/display/AST/PJSIP+Advanced+Codec+Negotiation) appears to ultimately be what we're after, but we're not comfortable running Asterisk 18 in production just

hi using the following in sip.conf, codec preferences aren't set, and asterisk uses alaw whatever I do, except force it to one specific in the [user] [general] disallow=all allow=g726 allow=g729 allow=gsm allow=alaw then, from 'sip show peer something' it tells me Codecs : 0x11a (gsm|alaw|g726|g729) Codec Order : (none) can someone please explaing why? this is

I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000. The codec order on each one is the next: SJPhone: GSM - iLBC - PCMA - PCMU GXP2000: G729 - GSM - PCMA - PCMU (I have a G729 license, so there's no problem with transcoding G729) In my sip.conf, I've defined the following codec order: disallow=all allow=g729 allow=gsm allow=g726 allow=alaw allow=ulaw And my

Hi, I'm looking into support for SS7 and I found an article (http://www.openss7.com/news13022002.html) which says that OpenSS7 supports Linux Support Inc's T400P-SS7 Quad T1 and E400P-SS7 Quad E1 PCI interface cards. It also says that Linux Support Inc is the primary sponsor of Asterisk. However I cannot find these cards on the Asterisk hardware page

hi Username : 1000012 Codecs : 0x11a (gsm|alaw|g726|g729) Codec Order : (gsm|g729|g726|alaw|ulaw) the above is from SIP SHOW PEER 1000012, and as it clearly shows, g.729 is preferred before alaw. If I dial this SIP - * - SIP from a phone with G.729 enabled, it uses G.729. However, if I dial from my cell phone - GSM - PSTN - * - SIP, the call uses ALAW, which I thought it

I've noticed the following situation: In two softphones, I've configured the next codec order for each one softphone 1: 1 - PCMA 2 - GSM softphone 2: 1 - GSM 2 - PCMA and in Asterisk, the order is: disallow=all allow=gsm allow=alaw If I call from softphone 1 to softphone 2, I presume that Asterisk should do transcoding (canreinvite is set to no):

Hi all, We use linux-2.4.20 in one of our products and we've found what looks to be a problem in the bridge module. (I know this is old code but we don't send our customers kernel upgrades unless we really have to!) The problem is that some of our bridges have become orphaned from the bridge module. Specifically ifconfig ourbridgename shows that the device "ourbridgename"

El Tue, 9 Jun 2020 09:46:32 -0600 George Joseph <gjoseph at digium.com> escribió: Hi George > > > > > > If transcoding is enabled Would it be possible to do the same but handle a > > 488 > > back from Bob and failover to another INVITE with Bob's allow list to > > handle > > transcoding? That way we would always try no-transcoding before

We are running Arch Linux as a new sever and only has samba4 available officially I am trying to migrate my samba 3 config to work with samba 4 I currently use samba to authenticate windows users to use our Linux shares. Then using the Unix groups setup in NIS to validate the users access to a particular share. Here is the problem. I can see the shares using samba 4 but it uses the

We are running Arch Linux as a new sever and only has samba4 available officially I am trying to migrate my samba 3 config to work with samba 4 I currently use samba to authenticate windows users to use our Linux shares using the unix groups as the valid users. Here is the problem. I can see the shares using samba 4 but it uses the "Domain users" group to write to the shares and not

Hi: I am useing asterisk 11.12. I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI use alaw. G722 is great when ip-phone talks to each other. but when ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to transcode to alaw. so I try to change the codec when dial from SIP to DAHDI. I tried to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP

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

Hello! I'm facing the following scenario: - Initial call opened to asterisk: SDP g722,alaw,ulaw - Outgoing call to provider started with Invite / SDP alaw, g726 and g729. - Provider sends 183 Session progress SDP: g729, alaw - Provider sends g729 rtp packages But: there is no license to transcode g729. What is asterisk doing? Asterisk decides to stop the call at all: - Sends cancel

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 list, Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle. Grandstream allows for 8 different codec specifications. I have defined them as 4 x G726 & 4 x alaw. Snom allow for 7 different codec specifications. I have defined them as 3 x G726 & 4 x G729. The SIP peers are both defined as : disallow=all allow=g726 allow=alaw allow=g729 allow=gsm This is the

Hi, I purchased a G.729 1 channel codec license from digium. And installed as per the documentation. Then configured the sip.conf to use the new codec. For that, I am added the following entries in sip.conf (via web interface, as i am using asterisknow 1.5) disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm After that, when try to call through the PSTN line I can hear the voice of