similar to: command - set sip_codec- does not work with asterisk-1.4.21

Hi, I am using Aserisk 1.4.23.1 and trying to use SIP_CODEC to define the codec being used. I have exclusively Polycom phones for this test, and basically I want all communications to use g729 (preferred codec), except for pagine 20 phones (which busts my g729 license count). In that case I want to use gsm. I have therefore specified Set(SIP_CODEC=gsm) I my dialplan before the appropriate

FYI, asterisk DOES now support g723, but you have to pay for it: http://store.yahoo.com/asteriskpbx/asteriskg729.html -----Original Message----- From: Dan Fernandez <danfernandez00@hotmail.com> Date: Mon, 5 May 2003 17:33:05 -0300 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work? Basically, since I?d like to use g723 for sip

I am wondering if the SIP protocol and its implementation in * allows for changing codecs mid-connection. I've seen some questions regarding this on the list, but I've not found any clear answers. I've also seen the SIP_CODEC variable, but it's not clear that it will change the codec on an existing call. Also, there are mentions of needing a reinvite to make the change, but most

Hello, I have a couple of 7960 and a quad T1 card on my asterisk box. I want to let the phones to use g729 when they "talk" to each other, but to use g711 when I'm going to route the call out of my network using the T1 card. Everything works just fine between the phones, but in order to be able to make calls through T1 I have to disallow the g729. For this purpose I have the

hi: when using chan_sip, I can use set SIP_CODEC in dialplan to change the codec of endpoint. this method didn't work with pjsip in asterisk 12/13. I found asterisk 12/13 has a new function PJSIP_MEDIA_OFFER. according to the description, it seems can set codec, but the document didn't offer any example. i try to use something like PJSIP_MEDIA_OFFER(alaw) but didn't work.

Hi, Could someone help me on configuring a H.323 trunk. I am trying to set up the following scenario: [SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)] I am using the following versions: Linux CentOS 3.3/2.4.21-.EL.co asterisk 1.0.1 pwlib_1.5.2 openh323_1.12.2 asterisk-oh323-0.6.3b Calling from Asterisk (2004) to the

We want some of our users to use G729, and some others to use ULAW. Our PSTN gateway provider supports both, so that's not a problem, and if I force him (the PSTN gateway) to allow G729 only, the outgoing call will take place with G729. The problem is that I want to have my PSTN provider configured to allow ULAW as a first priority, then G729. I did it like that: [mypstngate] type=friend

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: 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

I'm having a problem with a new SIP trunk. Calls within the UK to fixed lines are fine, but calls to mobiles have noticeably poorer audio quality. I thought it might have been a codec issue; we have used G.726 for internal and external calls (over primary ISDN and GSM). So I tried allowing "alaw", (G.711 A-law) which is the native codec used within the PSTN in this country,

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, 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 all, this is my actuel setup [SIP 2005]--[asterisk]--H.323 Trunk--[PBX]--[ext. 8900] Linux CentOS 3.3 (2.4.21-20.EL.c0) asterisk-1.0.1 asterisk-oh323-0.6.3b openh323_1.12.2 pwlib_1.5.2 Calling from SIPphone to the extension 8900 works always. Calling from 8900 to SIPphone works only sporadicly without any recognizeable pattern. Find below the output of the debug command: asterisk

Hi all, I have some OptiPoint400 standard SIP connected on Asterisk. They work pretty good. I only notice that calling from OptiPoint to OptiPoint doesn't show me the Caller ID name (only Caller ID number). But calling from an OptiPoint to a SoftClient (e.g X-Lite) shows me both on the softclient. Any ideas? Thx! sip.conf [2005] type=friend callerid="OptiPoint" <2005>

Hello, We're trying to receive G.711 (aLaw) faxes on the asterisk and convert them to tif. With T.38, we have several issues, so we are trying to use G.711, since the gateway is located in the same LAN, so there's no bandwidth/packet-lose issue. We also use on the same Asterisk Real-Time process for the extensions.conf My question: Is the following syntax for disabling T.38 support

Hi everyone, I'm wondering I would accomplish the following: I want to dial several SIP extensions simultaneously, HOWEVER, for different times (say ext 10 for 15 sec and ext 11 for 30 sec), and potentially with different headers (such as ALERT_INFO) and codecs for each extension. Obviously whoever picks up first gets the call. After the longest timeout expires (30 sec in this example) I want

All, Our users have a softphone client that supports the G723 Codec which we want to use for bandwidth reasons, however we do not wish (or have the funds) to license the codec on our Asterisk servers. We have G723 pass-through working between the clients just fine, however calls fail when terminating with Asterisk itself (i.e. Voicemail) or out to the PSTN due to transcoding issues. If it

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

On 03.10.19 15:08, Administrator TOOTAI wrote: > Before calling the gatreway add > > same = n,set(SIP_CODEC=alaw) > > [...] > Hey there, that doesn't work as it seems to be implemented for chan_sip only; I'm using chan_pjsip; sorry if I didn't explain myself properly. Anyway, in my case that would not really be an acceptable solution anyway, because I need the

hello, asteriskers: I compiled asterisk under arm-linux. i am using asterisk 1.4.2. i can run ./configure and menuselect with embedded modules. but running make comes out errors: ranlib libmxml.a make[3]: Leaving directory `/usr/src/asterisk-1.4.2/menuselect/mxml' cc -Wall -o menuselect.o -g -c -D_GNU_SOURCE menuselect.c cc -Wall -o menuselect_curses.o -g -c -D_GNU_SOURCE