similar to: One SIP peer use 2 diff codecs?

Ok, here's an odd one. I would have opened a bug on this but last time I tried that I got flamed.. :) Problem: When proxy requests digest challenge (SIP) Asterisk responds normally with the exception that for some reason it changes the FROM: (Also changes Contact: )to what's in the original TO: line. Why on earth is it doing this?! It must be a bug, I've gone over my extensions.conf

Does anyone know if it's possible to set the codecs for a number via an Asterisk command? Ie, yes you can set the codecs in sip.conf for a user, but I'd like to have a command that can set the same thing so that it can be done without having to change sip.conf. Essentially I want the user to be able to prefix a code to their dialled number to set their preferred codec for that call.

Or perhaps how to configure and refer to two parallel IAX trunks with different codecs? I have a situation where I'm using G.729A as my IAX trunking codec. Now I need to push some short duration, low bitrate modem traffic over the link (a credit card terminal). Obviously the modem audio isn't going to survive the G.729 codec process intact, so for the times the device is used I'd like

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 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

Hi, I`d like to know the best way to manage codecs to let the CPU breath as much as possible. I understand transcoding is a big part of CPU usage on Asterisk, and I have the following situation: - A SIP provider that offers me G729 and ULAW (my choice, both as allowed) - Some of my calls are from G729 enabled phones to outside lines, and I`d like to send those calls using G279 to my

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

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

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 want to submit a bug about auto-dial , but I am not sure on which project the auto-dial comes, how to know about which project , auto-dial comes Thanks Joseph ___________________________________________________________ To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre.

I have a ALPHA version of my new ping AGI demo available. Access via: IAXTel 1-700-923-3645 or Dial(IAX2/guest@ext.fnords.org) When asked for an extension, enter 2101. This will bring you to the System Services menu. The Cepstral version of the ping is option 28, the Festival version of the ping is option 32. Please report problems and/or issues directly to me. I'm trying to get

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, 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 List; What configuration needed to let my FXS send and receive FAX? Regards Bilal

Hi All! I am facing some delay in conferencing. Using DIAX for Voip calls ,no hardware used yet I am using MeetMe to achieve conferencing and am having a lot of delays. Can anyone tell me how to reduce the delay Regards, Amna Saleem -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, understand that the bandwidth utilized for each call is dependent on the codec used, wonder if Asterisk can monitor the total bandwidth utilized and restrict/reject new calls when the resource is insufficient to support them reliably? Regards Andy Tan -- Andy Tan andytan@fastmail.fm -- http://www.fastmail.fm - Does exactly what it says on the tin

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

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

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 list, I'm planning and testing a distributed asterisk deployment throughout several sites; each will be connected to the PSTN and all of them among themselves via IAX trunks. Phones will be SIP. I guess I already "solved" (worked-around, actually) asterisk's codec negotiation limitations regarding local G.711 utilization vs. remote G.729 while minimizing