search for: sip_codec

...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 communication between endpoints and * does not support it, I need to change codecs when a user wants to check voicemail, use a zap channel, etc. I have configured sip.conf and extensions.conf as below but when I try it I keep getting the f...

...ip 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 reports the condec compatibility problem. both of asterisks can show g729 are there. =================================== exten => 2005,1,Answer exten => 2005,2,Set(${SIP_CODEC}=g729) // does not work exten => 2005,3,DIAL(SIP/1000 at 192.168.2.127,30,r) exten => 2005,4,Hangup ===use sipp t call asterisk 1 then forward to asterisk 2 with sip 1000. ------------------sip.conf ---------------------------------------------------- [1000] username=1000 allowtransfer=yes ty...

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)...

...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 following configuration using yesterday's cvs. [sip.conf] disallow=all allow=g729 allow=ulaw [extensions.conf] exten => 123456,1,SetVar,SIP_CODEC=ulaw exten => 123456,2,Dial(${TRUNK}/${EXTEN}) The problem is with the SetVar function, the debug shows that the function is executed, but after that, * sends the media capability to the phone with g729 as preferred codec. Is there any work around? or simply setVar wasn't meant for this...

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 of the sample sip.conf contexts I've used for setting up our sip channels reccommend "canreinvite=no". Does that preclude a...

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 PJ...

...as a first priority, then G729. I did it like that: [mypstngate] type=friend host=192.168.0.100 port=5060 context=pstn-in canreinvite=no disallow=all allow=ulaw allow=g729 Then, in the outgoing context for our G729 SIP customers, I've put something like that: exten => _0NXXXXXXXX,1,setvar(SIP_CODEC=g729) exten => _0NXXXXXXXX,2,Dial(SIP/0041${EXTEN:1}@mypstngate,90) What happens now when placing a call is very interesting. As you can see, Asterisk wants to change the codec to g729, but on the outgoing call to the PSTN gateway it remains ULAW. Like this, I'm using up one of my G729 lic...

...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 codec change after dahdi answered the channel. so everything is broken. the call log like below: [2014-09-23 21:18:46] VERBOSE[11634][C-0000000d] pbx.c: -- Executing [s at macro-dialout-trunk-predial-hook:2] Set("SIP/222-00000004", "SIP_COD...

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

...allowing "alaw", (G.711 A-law) which is the native codec used within the PSTN in this country, but this made no improvement. We had disallow=all allow=g726 in the [general] section of sip.conf. In the section for one of the phones, I added allow=alaw and then inserted Set(SIP_CODEC=alaw) in the relevant part of extensions.conf. For good measure, I also added NoOp(Codec was ${SIP_CODEC}) in the "h" extension. The messages in the Asterisk CLI appeared to show that the audio codec was correctly being set to "alaw", and on hangup I got "Codec was al...

...appear to be able to set an inheritable variable for the subsequent PJSIP leg of the call, to exclusively only offer the codec we negotiated for the first leg of the call. If for example we have chan_iax2 incoming that we wish to send out via pjsip. With chan_sip, this works: exten => s,n,Set(_SIP_CODEC_OUTBOUND=${CHANNEL(audioreadformat)}) With pjsip, this gives an error: exten => s,n,Set(_PJSIP_MEDIA_OFFER(audio)=!all,${CHANNEL(audioreadformat)}) Error: ERROR[26925][C-00020b9c] pbx_functions.c: Function _PJSIP_MEDIA_OFFER not registered I'd image things haven't changed since 20...

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.

...s example) I want to move to the next priority, say voice mail. I think this is pushing the ability of the Dial application a bit, but I'm curious to hear what suggestions others have. I have already patched the Dial app some time ago to allow me to pass different variables for each extension (SIP_CODEC and ALERT_INFO) and this works quite well -- but I don't think this is the cleanest way of doing it. I could certainly tweak it to have different timeouts per extension, but I'm looking for ideas/feedback before I do this. Thanks... --Luki

>> allow=ulaw >Why don't you remove this? Because I need some other users to be able to call out using ULAW over the same PSTN gateway... -Manuel ___________________________________________________ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com

...BSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:0100000000 at 52.22.22.22:5160> Content-Length: 0 <------------> [2020-09-19 23:42:19] VERBOSE[15153][C-00021a1f] pbx.c: Executing [0100000000 at incoming:1] Set("SIP/Upstream-00021a0d", "1?SIP_CODEC=g729") in new stack [2020-09-19 23:42:19] VERBOSE[15153][C-00021a1f] pbx.c: Executing [0100000000 at incoming:2] NoOp("SIP/Upstream-00021a0d", "SIP Call ID: 7030be5a09d89a9543234da051897a49 at 41.11.11.11") in new stack [2020-09-19 23:42:19] VERBOSE[15153][C-00021a1f] pbx.c...

...2637][C-00021a1f] chan_sip.c: Capabilities: us - (g722|alaw|g729), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|g729) <snip> [2020-09-19 23:42:19] VERBOSE[15153][C-00021a1f] pbx.c: Executing [0100000000 at incoming:1] Set("SIP/Upstream-00021a0d", "1?SIP_CODEC=g729") in new stack [2020-09-19 23:42:19] VERBOSE[15153][C-00021a1f] pbx.c: Executing [0100000000 at incoming:2] NoOp("SIP/Upstream-00021a0d", "SIP Call ID: 7030be5a09d89a9543234da051897a49 at 41.11.11.11<mailto:7030be5a09d89a9543234da051897a49 at 41.11.11.11>") in ne...

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

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

...on the same Asterisk Real-Time process for the extensions.conf My question: Is the following syntax for disabling T.38 support correct? vm*CLI> -- Executing Set("SIP/Proxy-00000000", "t38pt_udptl=no") vm*CLI> -- Executing Set("SIP/Proxy-00000000", "SIP_CODEC=aLaw") vm*CLI> -- Executing Answer("SIP/Proxy-00000000", "") The aLaw Set command is taken into consideration, because the SDP of the OK that follows these lines includes only codec 8, but when the ReceiveFAX command is executed, Asterisk immediately sends a T.38...