search for: vtnoc

...terisk GUI Version: 2.0 The system was completely set up using the Asterisk GUI with a couple tweaks in users.conf that via:talk wants. Here is what happens: 1. Asterisk verifies connection to the server and we get this. (CLI output) -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host galvatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host optimusprime.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host megatron.vtnoc.net, port 50...

...when we were on the initial server we registered with but when we got pushed to another server the DTMF would fail till I did a sip reload or restarted Astersk. Now we get no DTMF ever. System set up. Asterisk 1.4.22 Asterisk GUI 2.0 users.conf [trunk_1] context = DID_trunk_1 host = galvatron.vtnoc.net username = user name secret = password trunkname = via:talk - galvatron ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no fromuser = user name authuser = user name insecure = port,invite dtmf = rfc2833 dtmfmode = rfc2833 relaxdtmf = yes...

...se two settings with no change. This is on asterisk 1.2.27 that's been working fine in production for about 3 months now. Here's the section from sip.conf (the way it had been working all along): [viatalk] type=peer secret=(yep it's right) username=(yep it's right) host=newyork-1.vtnoc.net canreinvite=no insecure=very qualify=yes context=incoming-viatalk dtmfmode=inband ; Choices are inband, rfc2833, or info ;relaxdtmf=yes ; Relax dtmf handling Thanks in advance for any help. I've got all incoming calls on Viatalk shunted to an extension in the mean...

...XXXXXXXX at default-c101,1 of format ulaw since our native format has changed to unknown ************************************** SIP.CONF Example Line *************************************** [trunk0] authuser=191691245XX username=191691245XX fromuser=191691245XX secret=12345 fromdomain=richmond-1.vtnoc.net host=richmond-1.vtnoc.net dtmf=auto dtmfmode=inband insecure=port,invite qualify=yes type=peer canreinvite=yes call-limit=2 context=viatalk -- /Nick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/2...

...when we were on the initial server we registered with but when we got pushed to another server the DTMF would fail till I did a sip reload or restarted Astersk. Now we get no DTMF ever. System set up. Asterisk 1.4.22 Asterisk GUI 2.0 users.conf [trunk_1] context = DID_trunk_1 host = galvatron.vtnoc.net username = user name secret = password trunkname = via:talk - galvatron ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no fromuser = user name authuser = user name insecure = port,invite dtmf = rfc2833 dtmfmode = rfc2833 relaxdtmf = yes...

...r entries, and incoming and outgoing calls work, but incoming calls do not move to my in-viatalk context (they stay in the default context.) Has anyone else managed to get this to work? My user entry looks like: [viatalk-in] username=1407965XXXX context=viatalk-in type=user host=965.407.1.switch.vtnoc.net I've also tried username=+1407965XXXX, host=67.15.74.73, host=67.15.74.73:5060, and host=dynamic. SIP debug from an incoming call shows: <-- SIP read from 67.15.74.73:5060: INVITE sip:s@10.1.42.254 SIP/2.0 Via: SIP/2.0/UDP 67.15.74.73:5060;branch=z9hG4bK6169ed4e;rport From: "We...