similar to: set different codec for different sip calls

hello: i want to test the g729 with asterisk. my scenario is sipp(ulaw)->asterisk1 with g729->asterisk2 with g729. I want to test g729 module with asterisk-1.4.21, when i make calls from asterisk 1 to asterisk 2, the asterisk 1 always send ulaw to asterisk 2. my sip in asterisk 1 is with codec g729 and enforce that use g729, the sip in asterisk 2 also work with G729 only, but asterisk 2

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

Hello, I'd like to implement some public sip uri's that poeple can call into and get an echo test. Is there a way to force a codec so that users can test various codecs? Something like: echo-test at example.com (negotiates whatever codec, is there a way to figure out what codec was negotiated and tell the user) echo-test-g711 at example.com (forces g711) echo-test-g729 at

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

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

I didn't know how else to caption this. I'm trying to play around with codec pass-through. I have two SIP phones, both with g729, behind two Asterisk servers. I set all the configs, SIP and IAX, to "disallow=all; allow=g729" on both servers. But the originating server won't even try to call the destination server: -- Executing Dial("SIP/btel-c7d7",

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 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, 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, 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

Hello, Has anyone experienced a segmentation fault in asterisk for using G729 against an AS5300 in SIP ? I'm having this problem. Also, any other codec I use gives me incompatible media (can be read in SIP DEBUG messages). AS5300 configured for multiple codecs, so is Asterisk. Tried G711u/A G723 and G.729. Any clues ? Regards, Jorge A. Info: Asterisk ver 1.0.7 stable Using AMPortal

Hi All, I'm trying to connect to a SIP carrier who never connected with Asterisk. I managed to connect with a sipura phone or a grandstream, no problem. When I configure asterisk, I'm able to send out calls to the carrier no problems, however, receiving calls doesn't work, and I keep getting the following messages: <-- SIP read from 69.xx.xx.xx:5060: INVITE

Is there any way to set Asterisk to choose what codec to allow for a new call based on current usage? In other words... be able to define a max number of ulaw calls, then after that only allowing g729? The idea here is that in general, a T-1 should be enough for our offices to have phone + citrix + some video (got good QoS in place already). But for usage spikes, user experience would be kept

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

I asked this question once before with no answer. Hopefully someone can help me as I cannot see a way to do this. I am wanting to differentiate inbound calls voice from FAX. The purpose of course voice gets g729 and FAX gets 711 (ulaw). The problem I'm having is everytime it matches the SIP peer (like it should) but it's always goes to the prefered codec. Anyone have suggestions on how 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

Hello. I have a codec problem to send calls from a SIP device to a H323 gateway. First I'll explain the scenario: - Asterisk 1.2.1 - The SIP phone can use any codec I want. - The H323 gateway can only use g729 (cause it's not under my administration) - SIP phone has g729 configured, so my asterisk doesn't need to "transcode" (I don't have licences for g729) - sip.conf

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 am currently trying out the asterisk@home (version 1) release of Asterisk, and I want to configure it as follows: Calls from regular telephony network (PSTN) come in through my VoIP provider over SIP and outgoing calls to the PSTN should be routed through the ViOP provider onto the PSTN network. I thus have no direct PSTN connection, but only a SIP connection. Incomming calls

Did anyone have a way to make codec negotiation work with Asterisk? This is something I would love to have working as well. I won't need PSTN -> G729 mixing. Just SIP -> SIP using G729 for calling remote offices via VPN, but everything else use G711. -----Original Message----- From: Brancaleoni Matteo [mailto:mbrancaleoni@espia.it] Sent: Wednesday, July 16, 2003 11:32 AM To: