similar to: Codec preference selection?

Displaying 20 results from an estimated 10000 matches similar to: "Codec preference selection?"

2006 Jan 31
1
Fw: Codec preference selection?
Hi! No one can help me with this?? ----- Original Message ----- From: Fran Sedano To: asterisk-users@lists.digium.com Sent: Monday, January 30, 2006 8:43 PM Subject: Codec preference selection? Hi; I'm trying to implement what is known by Cisco Callmanager as regions: Specify that when phones from zone A call to phones in zone B, use g729, but if they call to zone C, use g711. Any
2004 Jan 05
0
Codec Negotiation Does not seem to work as e xpected ?? Help Please !!
Steve, My Problem is not a problem, with the codec negotiation between end points. But when asterisk does it with canreinvite=no, * do not do it right. I replied with a lengthy discussion about my findings here, This behavior can be reproduced. But '*' do not seem to do the negotiation correctly. http://lists.digium.com/pipermail/asterisk-users/2004-January/032197.html
2003 Jul 18
0
FW: Sip codec preferences
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:
2010 Mar 17
3
SIP codec negotiation / manipulation
We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation. We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second
2004 Jan 05
2
Codec Negotiation Does not seem to work as expected ?? Help Please !!
Hello, I have been trying to get my coders to work without a conversion. I have read all the available asterisk documentation and support groups without any luck. Here is my issue. (Please feel free to ask questions if you do not understand what I am talking about.) I am using Cisco ATA-186 set to g729 codec. (But it will switch to g711 if sip-server request g711) I have 2 SIP-services to
2004 Nov 24
2
Codec control
How can i control the codec for the calls. For example I have 3 SIP phones registered to asterisk The firs two are in the local area network (behind nat)- I want to use g711 between them and to connect directly (canreinvite=yes) and the third is in internet - want all calls to it and from it to use g729 and media to go through asterisk. So if Phone 1 calls Phone 2 the codec to be g711, but when
2009 Oct 20
1
Is there a way to force a codec on an incoming sip uri call?
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
2011 Jun 13
5
No audio after a reinvite changing codec
Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 <----------------------> g711
2004 Dec 02
4
Codec Conversion
Hello, Is there an utility for asterisk for codec conversion? I tried google but i haven' got anything. I am trying to initiate a call with G711 codec to asterisk and i would like asterisk to call a gateway with an g729 codec, therefore making a codec conversion from g711 to g729. I know chan_oh323 does it by specifying the OUT_CODEC variable, but chan_h323 does not. And i was wondering is
2006 Dec 15
2
call from h323 to SIP
Hi i am trying to do the same thing: receive a call from a cisco callmanager and forward it to a SIP user. Asterisk is compiled with h323 support, and is configured as a gateway in the cisco callmanager. h323.conf: [general] port = 1720 bindaddr = 193.x.x.x ; this SHALL contain a single, valid IP address for this machine allow=all extension.conf: exten = 3298,1,Answer exten =
2007 Jun 28
2
Fax passthrough howto codec upspeed
Hello everybody, Just was wondering if somebody can help for G711 fax passthrough w/ asterisk. The issue I have is regarding codec upspeed when the call is already connected using G729 for example. The setup is fax---ATA---asterisk---Cisco---fax When codec upspeed should happen, ATA or Cisco will send a G711 reINVITE causing the codec to be switched over, but asterisk does NOT
2003 Oct 18
0
Oh323 cisco callamanager
hi , i'm testing asterisk like and Automatic attendant with a callmanager and vg200 gateway with 1 t1 everithing works finw but some times asterisk didnt not disconnect calls and star growing the number of connections from asterisk to callmanager , and when this connections get to 35 g711 , the asterisk hang. some one , ?? i'm using asterisk-0.5.0 and oh323 5.5 regards , victor
2006 Oct 13
1
3way calling / codec problem
I'm having problems with conference calls (3-way) when I have my codec forced to g729 in sip.conf. I'm using Grandstream 2000s. If enable both g711 and g729 then 3 way calling and transfers work. I'm not sure why this would matter? Here's the error: Oct 13 13:54:45 NOTICE[31184] chan_sip.c: No compatible codecs! Any help is greatly appreciated!
2007 Jun 06
4
Best Codec
We are evaluating starting a small VoIP consumer based platform. What is the best codec to use with customers using primarily DSL as internet connectivity? I know that g729 is the king-all, but I want to know what the rest of the professional are using out there. g729 has a cost involved, so does the cost really offset the performance? Or is it better to go with g711 to start off? We plan
2003 Nov 06
0
which codec will be used ?
hello, my situations is as follows. In our comapny we are planing to have *. I'm testing it now. If we will buy G729 codec for * ... UA(SIP) <-> FW <-> (SIP)*(H323) <-> (H323)GATEKEEPER(H323) <-> (H323)AS5300 <-> world the following equipment speeks G729: *, GK ,AS5300. All call from UA to another endpoing go through *, because of mixed SIP and H323
2006 Apr 19
1
Codec problem from SIP to H323
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
2008 Mar 27
1
ADPCM codec and IAXy device
Hi All; I need to buy one IAXy device, but I discovered that it supports only g711 and ADPCM codec, so I was wonder that it does not support g729 or GSM?! Anyway, what is that ADPCM and how much it consumes bandwitdh? Also, asterisk support such codec? What its name in the configuration? Any advise? Regards Bilal
2014 Jun 04
1
Renegotiate SIP audio codec after call is up
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2023 Jul 05
1
Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
Hello Michael, you are referring to the following behavior - did I get it correctly?: outbound broken: asterisk offers g722 / g711 to provider (callee), callee answers g711. Asterisk now transcodes between caller and callee (g722 <-> g711). inbound works: call from provider: g711 -> asterisk drops g722 and passes g711 to internal callee -> no transcoding. As far as I know,
2008 May 21
1
speex, ilbc and g729 codecs, GSM with IAX
Dears; I do not know if any had experience in using speex or ilbc with IAX and got good results, because I am facing a problem with GSM. I am facing a noise problem when I am using GSM with IAX trunk as following: IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk using GSM codec ---> Remote Asterisk Box ---> Digium Card (FXO) to terminate the call to the destination While no