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Hello, I'm a new Asterisk user and I hope I haven't missed something, but I can't seem to find an answer to this issue. I have a Cisco SIP gateway terminating calls into a 7960 phone. The issue I would like to fix is if I have an incoming call without an ANI, such as directly from my TDM phone switch, Asterisk says the call is coming from the IP address of the Cisco gateway,

I'm trying to get Asterisk to pass thru calls using the G729 codec. I've got a 7960 phone and my gateway is an AS5400. I got the following messages when debugging SIP (7778881000 is the 7960): WARNING[1872]: channel.c:2115 ast_channel_make_compatible: 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

Hello, Earlier I was having problems getting g729 pass-thru to work correctly. I was able to fix this by putting allow=g729 as the first allow statement directly under the configuration of the user. I also put allow=ulaw in there below it. It seems the second "allow" statement is totally ignored, as asterisk only negotiates codecs based on the first statement. It looks as if asterisk

Hello, I'm wondering if anyone has benn able to successfully get g726-16 passthrouhg to work? I am wanting to use this codec instead of g729 as I'm running out of DSPs using a high complexity codec on the Ciscos. I would think it would work just as g729 does, which has been working fine for me, but it does not. G726-32 does work great however, but it's like Asterisk doesn't

Andrew Thompson wrote: > You should be able to set the inbound callerid from the switch/gateway > to a specific unknown in sip.conf file with just a callerid= line. > > The place I looked on the wiki didn't show a specific description for > the callerid= line, but that's what I thought I read for it somewhere. > >

Hello, I'm trying to run an ATA behind a NAT device, and am confused on exactly what the qualify config option does, other than send NOTIFY packets. Outbound calls work fine, but inbound calls do not go through. With qualify=yes and nat=yes, my show sip peers looks like: 7771111001/7771111001 10.0.0.10 D N 255.255.255.255 1222 OK (36 ms) So, it has established a