search for: smvoice_callprogress

...(4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096) Under 1.4.41 I get an error and hang up doing the exact same thing. All I am doing Is calling a cell phone over the PRI then dialing my SIP/524 extension. This is from 1.4.35 > Channel DAHDI/18-1 was answered. -- Executing [smvoice_callprogress at smvoice-dialout:1] GotoIf("DAHDI/18-1", "1?smvoice_callprogress|3:smvoice_callprogress|2") in new stack -- Goto (smvoice-dialout,smvoice_callprogress,3) -- Executing [smvoice_callprogress at smvoice-dialout:3] AGI("DAHDI/18-1", "smvoice) in new stack...

I am getting this error from call manager (4.0) and asterisk 1.2.4 I have canreinvite=yes on the call manager setup. I can call into the asterisk box from call manager. THat seems to work. When I am calling out of the box using a call file I see this entry from call manager... What might be the problem with my setup? THanks, JErry ---------------- <Date>03/06/2006

...release_4.0 version_sse-4.00.55 Content-Length: 0 <-------------> ?--- (10 headers 0 lines) --- ? [Khfemsrv*CLI> Really destroying SIP dialog '3ee92dbe77f51a1748f736be4593719d@161.49.142.250' Method: OPTIONS ? [Khfemsrv*CLI> -- Attempting call on SIP/QuadNortel/7113 for smvoice_callprogress@smvoice-dialout:1 (Retry 1) ? [Khfemsrv*CLI> Audio is at 161.49.142.250 port 10000 ? [Khfemsrv*CLI> Adding codec 0x4 (ulaw) to SDP ?Adding codec 0x8 (alaw) to SDP ? [Khfemsrv*CLI> Reliably Transmitting (no NAT) to 192.168.45.129:5060: INVITE sip:7113@192.168.45.129 SIP/2.0 Via: SIP/2.0/...

I am using asterisk 1.4.21 with outgoing call files. I am call a line that is busy as you can see below. How can my AGI ask what the status of the last call was so I can tell if there was NO ANSWER or it was BUSY? Thanks, Jerry -- Attempting call on SIP/401 for smvoice_callprogress at smvoice-dialout:1 (Retry 1) -- Got SIP response 486 "Busy" back from 192.168.1.161 > Channel SIP/401-15aa5ab0 was never answered. -- Executing [failed at smvoice-dialout:1] AGI("OutgoingSpoolFailed", "smvoice") in new stack -- Launched AGI Scr...

-- Attempting call on DAHDI/1wwwwww for smvoice_callprogress at smvoice-dialout:1 (Retry 1) [Oct 16 14:36:42] WARNING[16408]: chan_dahdi.c:8132 dahdi_request: Unknown option 'w' in '1wwwwww' [Oct 16 14:36:43] WARNING[16408]: chan_dahdi.c:1481 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) Does DAHDI not k...

...am seeing the above error on DAHDI 2.1.0, asterisk 1.4.22 and libpri 1.4.7 I am using a TE120P card. I am also getting this VERY frequently: -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'DAHDI/0-1' Versus a normal hangup: == Spawn extension (smvoice-dialout, smvoice_callprogress, 4) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' system.conf: loadzone=us defaultzone=us span=1,1,0,esf,b8zs bchan=1-3 dchan=24 chan_dahdi.conf: [channels] switchtype=national signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asre...