search for: g729r8

...data before 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 require...

...Link. I want that all call incoming on the cisco 5300 are sent to Asterisk and all Asterisk outgoing call are sent to Cisco AS5300. Actually, i configure the AS5300: isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec preference 2 g729r8 codec preference 3 g723r63 codec preference 4 g711ulaw ! voice class codec 500 codec preference 1 g729r8 codec preference 2 g723r63 ! controller E1 0 framing NO-CRC4 pri-group timeslots 1-31 description E1 Beta-Test interface Serial0 no ip address shutdown clock rate 2015232 no fair-qu...

...ervice password-encryption ! hostname gw1 ! boot-start-marker boot system tftp mc3810-a2isv5-mz.123-10a.bin 192.168.5.104 boot-end-marker network-clock base-rate 56k no aaa new-model ip subnet-zero ! voice class codec 10 codec preference 1 g711ulaw codec preference 2 g711alaw codec preference 4 g729r8 codec preference 6 g729ar8 ! ! no voice confirmation-tone ! controller T1 0 mode cas framing esf linecode b8zs ds0-group 1 timeslots 1-4 type e&m-wink-start fdl both ! controller T1 1 mode cas framing esf linecode b8zs ds0-group 1 timeslots 1-4 type e&m-wink-start ! ! ! interface...

...ith asterisk, 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/20030...

Hello, I'm trying to receive faxes with asterisk. My configuration is like this: PSTN fax -> ISDN -> Cisco router with VoIP module -> Asterisk When I try to send a fax from PSTN fax I got the standard fax signal, Asterisk starts rxfax application and then call ends and there is no tif anywhere. On the fax display there is still one message: Calling... Part of my extensions.conf:

...NNEX-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

...le: No path to translate from 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 f...

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

...: 0 6995: Bridge Done : 0 6996: New Media : 0 6997: DSP DNLD Reqd : 0 6998: Media Dest addr/Port : 65.67.76.41:11980 6999: 7000: 006954: Dec 29 01:48:05.079: sipSPIHandleInviteMedia: 7001: Negotiated Codec : g711ulaw, bytes :160 7002: Preferred Codec : g729r8, bytes :20 7003: Preferred DTMF relay : 0 7004: Negotiated DTMF relay : 0 7005: Preferred and Negotiated NTE payloads: 101 0 7006: 7007: 006955: Dec 29 01:48:05.079: sipSPIDoQoSNegotiation - SDP body with media description 7008: 006956: Dec 29 01:48:05.079: sipSPIDoQoSNegotiation: Best effort call...

...codec g711ulaw sip ds0-num header-passing registrar server expires max 600 min 60 ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g711alaw codec preference 3 g723ar53 codec preference 4 g723ar63 codec preference 5 g723r53 codec preference 6 g728 codec preference 7 g729r8 ! ! ! ! ! ! ! ! ! username nvxcpe privilege 15 secret 5 $1$oAvO$zHweQ8lbhZjzROdYw4psM/ ! ! controller T1 1/0 framing esf linecode b8zs cablelength short 133 pri-group timeslots 1-8,24 description WAN ! ! ! interface FastEthernet0/0 no ip address shutdown duplex auto speed auto ! interface...

List users, It's been a while since I've posted here, but I've been hard at work pushing toward our large scale Asterisk goal and keeping up with this list can be a full time job by itself (I have19,543 unread list messages!!). This Friday, September 16th 2005, my team will be at the MCI Development Lab in Richardson, Texas testing our setup. We have a three server system

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