search for: g729br8

...re I can implement * on this particular solution. My question is perhaps a slightly differently worded one than has been asked before, but it may be the case that it is the same question as I have seen already posted (with no 'definitive' answer): "Can g729 calls of type g729r8 or g729br8 from a Cisco AS5300 be terminated on Asterisk systems and sent out Zap interfaces?" If the answer is "Yes," then are there any specific patches I will need? Which of the two H323 drivers works? Both? Of course, I assume that the G729 licenses from Digium are required for each...

...ybody on the list who implemented Cisco ATA + * + Cisco 2600? I cannot get my calls from ATA to terminate to the Cisco gateway via *. I am not sure if it is my hardware problem. I'm getting the following "codec negotiation problem" from Cisco. 23:39:08: Unexpected VoIPCodec Type :g729br8 23:39:08: Unexpected VoIPCodec Type :gsmefr I appreciate any help I can get. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040108/75c7ec0e/attachment.htm

Hi to all, I got a Cisco 2651XM wired to an E1 PRI. What I want to do is to pass all incoming calls to my asterisk. This is my actual conf: http://pastebin.ca/270677 with this I'm able to call my number from outside, but the call stop on the 2600, infact I can hear the tone, but I'm not able to forward calls to my asterisk. Anyone got an idea of my errors? Thanks to all. -- .:FaberK:.

...erisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that digium's g.729 is not compatible with these codec found on cisco AS5300. Am I correct? Any advice will be helpful Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030922/c88a...

...Law 64000 bps g723ar53 G.723.1 ANNEX-A 5300 bps g723ar63 G.723.1 ANNEX-A 6300 bps g723r53 G.723.1 5300 bps g723r63 G.723.1 6300 bps g726r16 G.726 16000 bps g726r24 G.726 24000 bps g726r32 G.726 32000 bps g728 G.728 16000 bps g729br8 G.729 ANNEX-B 8000 bps g729r8 G.729 8000 bps

...om SIP/7778881000-2874(4) to SIP/as5400-35c1(256) WARNING[1872]: app_dial.c:1002 dial_exec: Had to drop call because I couldn't make SIP/7778881000-2874 compatible with SIP/as5400-35c1 I have the following setup in the AS5400: voice class codec 1 codec preference 1 g729r8 codec preference 2 g729br8 and the following in the sip.conf file: [as5400] disallow=all allow=g729 It looks like it should be pretty simple to get this working correctly. My extensions.conf file has the line: exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@as5400) so I think it would pass-thru fine since this is an outboun...

Does anyone in here know how to setup auto negotiation between g729 and g711ulaw on a cisco 5400? I would imagine it would be the same on a 3660. The problem I am having is natively the call is setup for g729 however when the call is transferred to voicemail it uses ULAW so when the cisco tries to connect to the voice mail I get a SIP error that the codec couldn't be negotiated. I need

Hi, I have a serious problem to configure Cisco AS5XXX and Asterisk , I trying to use asterisk for PSTN(A)----> Cisco AS5xxx ----> ASterisk---->PSTN(B) (No Nat, no Firewall) I hear (on the PSTN(A)) clearly what the other person is saying, but the other person (on the PSTN(B) side) hears nothing from PSTN(A). I use tcpdump for debug de rtp trafic, and ouput contains