similar to: which codec over iax => pstn

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,

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

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

Hello, Anyone? I have hard time to believe this is not possible with chan_pjsip. Anyway, may I ask how people handle the following scenario which I imagine should be quite common: - I have internal extensions talk to each other using g722. so their codec setting (with chan_sip now) is "allow=g722,ulaw" - I have carriers trunks that handle ulaw only (allow=ulaw) - calls between

Hi All, I am wondering that how I can setup different codec for different extensions in my dial plan. scanario will when user X (Sip) call 111 extension in default context. The Asterisk response should be in GSM codec When user X (Sip) call 222 extension in default context. the Asterisk response should be in G711 Codec Actually I want to setup an extension for FAX receiving (rx_fax) and

I receive an INVITE/SDP containing: m=audio 11310 RTP/AVP 3 0 101 which I interpret as gsm, ulaw, rfc2833. and I reply with an OK/SDP containing: m=audio 15884 RTP/AVP 0 3 101 which I interpret as ulaw, gsm, rfc2833. How can I tell which codec was actually used for the call? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards

The number of codecs is overwhelming to me. What do ya'll consider the best codec for conserving bandwidth? [I realize at the cost of quality] Secondly, what do you think the best codec for voice quality is? Yours, Matthew

Using FireFly, all other codecs but G711 Ulaw is selected. But whenever I place a call, I get: Jan 24 16:07:06 WARNING[30654495]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Umm, wtf? I thought Inband was ONLY supported on G.711 u-law. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have looked at the codec's available but I don't know how get the highest quality recorded message. If a user calls in over the normal telephone network is this limited to the carriers codec or the codec at the asterisk side? Would I get a higher quality result using VoIP rather than the normal network? Any help would be appreciated. Thanks Fats.

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

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

Hi List; Where I determine the codec to be used for the SIP Trunk (between Asterik and another SIP softswitch)? Regards Bilal ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

Have asterisk@home 1.2.1 The server is on an internal network eg 10.10.10.10 It is NAT'd 1:1 via Checkpoint firewall to external public IP eg 50.50.50.50 The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on extension 1055. Outbound calls to 1055 work perfectly. Inbound calls from 1055 get picked up as if it were an external call (see below) and goes straight to the ring

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

Has anyone done anything with Asterisk using the G.711 codec? Also, is there a uncompressed option so that you could assign a single port to be unconpressed audio? Thanks! Stu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030602/118e6ac7/attachment.htm

Hello! I'm facing completely choppy sound. The wireshark trace shows, that there are a lot of codec changes without any trigger (means no options or reinvite or any other package). Background: The call is initiated by asterisk and is received by the same asterisk conference room via Phone extension -> asterisk -> provider A -> provider B -> asterisk. Asterisk initially sends

We are planning to run an outbound only campaign. A 20-second voice message will be played to callers and our dialer on machine1 will send to machine2-asterisk (1.4) instructions to dial 400 calls, play the message and hang up. This will be done for about 1 million phones. The asterisk box will communicate via SIP to a voice carrier. the voice carrier will then place the calls on pstn. The codec

> On Thu, May 10, 2018 at 11:44:14AM -0700, Steve Edwards wrote: >> I receive an INVITE/SDP containing: >> >> m=audio 11310 RTP/AVP 3 0 101 >> >> which I interpret as gsm, ulaw, rfc2833. >> >> and I reply with an OK/SDP containing: >> >> m=audio 15884 RTP/AVP 0 3 101 >> >> which I interpret as ulaw, gsm, rfc2833. >>

Hi list, I'm with a little problem on codec negotiation between a cisco827 and asterisk. My sip.conf is like that: [general] port = 5060 bindaddr = 0.0.0.0 context = default amaflags = default allow=g729 allow=gsm allow=alaw allow=ulaw ;disallow=all and cisco like that: dial-peer voice 6 voip destination-pattern 0T session protocol sipv2 session target ipv4:<asterisk-ip>

Hi, Want to do some experiments with the G.723 codecs - where can I download the 723 source code for Asterisk? I know there are some ongoing discussion regarding patents and license fees for the g.723 but I have some hardware on which I only have the 723 and need to test it privately. Thanks! Dan _________________________________________________________________ Use MSN Messenger to send