search for: lbrcodec

Yesterday, I started to experiment with Cisco to Cisco SIP calls using the g729 codec. According to the documentation, both the ATA-186 and 7960 are able to make use of the g729. >From an earlier e-mail, I made a change to the configuration of the ATA, changing the values: LBRCodec:3 RxCodec: 3 TxCodec: 3 The first thing I noticed was that when I did a sip show channels, the format had changed from ULAW to ALAW. The problem that I am having is that when I attempt to call any of the 7960, I get the following: -- Executing Dial("SIP/1200-3bc9", "SIP/1000|50...

...00000000000000000000 Dhcp: 0 StaticIP: 192.168.1.27 StaticRoute: 192.168.1.1 StaticNetMask: 255.255.255.0 EPID0orSID0: . EPID1orSID1: . CA0orCM0: 192.168.1.59:2727 CA1orCM1: 0 CA0UID: 0 CA1UID: 0 MGCPVer: NCS1.0 RetxIntvl: 500 RetxLim: 10 MGCPPort: 2427 CodecName: PCMU,PCMA,G723,G729 LBRCodec: 3 PrfCodec: 1 AudioMode: 0x00350035 ConnectMode: 0x90000400 CallerIdMethod: 0xc0019e60 DNS1IP: 0.0.0.0 DNS2IP: 0.0.0.0 Domain: . NumTxFrames: 2 TOS: 0x000068b8 OpFlags: 0x00000002 VLANSetting: 0x0000002b Polarity: 0x00000000 FXSInputLevel: 0 FXSOutputLevel: -4 SigTimer: 0x00000064...

I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]: channel.c:1508 ast_set_read_format: Unable to find a path from G726 to SLINR Jul 9 13:15:3...

Clicking on the "For more information, click here" link on the Digium site nice brings back up the same page I was looking at before, without any additional G.729 information that I can see. I'm wondering if some kind asterisker out there could provide us neophytes with some "typical scenarios" where that codec would be useful to us. For instance, I assume that it

Hi, I'm using a Cisco ATA186 and iConnect to complete PSTN calls to the US. I've noticed that when I set the Cisco ATA to use LBRCodec to 0 (g.723 instead of g.729), AudioMode 0x00150015 and RxCodec, LxCodec to 0, (use g.723) Asterisk will connect to iConnect, successfully natively bridge the call and then about two seconds later not just drop the call, but terminate unexpectedly. The asterisk daemon will stop uncleanly. If...

I'm having a horrible experience getting a Cisco ATA-186 to work with *. I can make calls from the ATA with no problems. However, incoming calls make the ATA ring once, and then the call is disconnected. I have no problems with my Sipura 2000 or my Grandstream phones. I am running 2.16.1 sip code on the ATA 186. Neither * nor the ATA is behind a NAT. They are both on public IP addresses

...s.digium.com > Subject: [Asterisk-Users] ATA 186, firmware SIP 3.1 and codec g.726 > > I have a ATA 186 with SIP firmware 3.1 when I changed the configurations > to > use the g.726 codec I received many erros and the calls doesn't work. > > I changed the fields: > > - LBRCodec: 6 <- the code for g.726 > - TXCodec: 6 > - RxCodec: 6 > > The errors: > > Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to > calculate samples for format G726 > Jul 9 13:15:37 NOTICE[1192491824]: channel.c:1508 ast_set_read_format: > Unable to fi...