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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 -- La...

...t starts this works. In fact it works for some time. Then it just stops with this error on the CLI. [Apr 4 15:10:21] NOTICE[4357]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_mediaport105' to extension '1105' rejected because extension not found in context 'smvoice-mediaport'. When doing the "dialplan show" it clearly in the context. [ Context 'smvoice-mediaport' created by 'pbx_config' ] '1105' => 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] Its telling me it cannot find it. Its there - t...

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

I am using asterisk 1.8.3. I am getting this error: [Mar 15 09:49:12] NOTICE[1049]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_vizioconfrm104' to extension '1104' rejected because extension not found in context 'smvoice-mediaport'. "dialplan show" gives me that the context is present: [ Context 'smvoice-mediaport' created by 'pbx_config' ] '1104' => 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] 'mediaport_direct' => 1. Goto(smvoice-m...

...(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...

...ere for me to dial. However, I get the INV when I dial. Any ideas on why the phone is displaying invalid and what to do about it??? Thanks, jerry sip.conf ------------------------ [201] type=friend dtmfmode=rfc2833 username=201 secret=201 disallow=all allow=ulaw allow=alaw host=dynamic context=smvoice-sip callerid="Media Assistant" <201> [202] type=friend dtmfmode=rfc2833 username=202 secret=202 disallow=all allow=ulaw allow=alaw host=dynamic context=smvoice-sip callerid="Media Assistant" <201> [203] type=friend dtmfmode=rfc2833 username=203 secret=203 disallow=a...

...IP that - and just let it do the Dial() - I stopped everything - got it running again. - and then the Dial() hangs on the second call. So both ChanIsAvail() or Dial() both hang on the second call in. So only 1 call in will work. Below is the CLI report of the call that works. This is my context [smvoice-mediacontroller-public-address] exten => s,1,ChanIsAvail(Console/default) exten => s,n,GotoIf($["${AVAILCHAN}" = ""]?smvoice-busy,s,1) exten => s,n,Playback(beep) exten => s,n,Dial(Console/default) exten => s,n,Hangup Now what ??? Jerry onnected to Asterisk 1...

I am upgrading a box from 1.4 to 1.6 and my console/dsp stopped working. I am getting a SIP/401 Unauthorized error and then a SIP/404 error. I changed nothing in the configs. Is there a particular parameter needed for 1.6 that 1.4 did not care about? If I drop back to 1.4 it starts working again. Thanks Jerry

...ALSA??? some traces below Jerry == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4.26.1 currently running on ebox3850 (pid = 3902) Verbosity was 0 and is now 5 -- Executing [mediaport_audio_visual at smvoice-mediaport:1] Goto("SIP/devcentos5x64_to_ebox3850-0837c170", "smvoice-mediaport-audio-visual|s|1") in new stack -- Goto (smvoice-mediaport-audio-visual,s,1) -- Executing [s at smvoice-mediaport-audio-visual:1] ChanIsAvail("SIP/devcentos5x64_to_ebox3850-0837c170&q...

...1]: chan_sip.c:16416 handle_request_invite: > > / Call from 'devcentos5x64_to_ebox4300' to extension > > 'mediaport_audio_visual' rejected because extension not found. > > > > Jerry-- > > > > from the console, type "dialplan show smvoice-mediaport", and > > let's verify for certain that it's in there. > > > > I'll try to reproduce your problem in my test system here. > > > > murf Jerry-- I think you've found a bug! I put in an smvoice-mediaport context just like the one you des...

ok switching to "Console/default" does show the text --- <("<) --- Call to device 'default' on console from 'default' <2564286000> --- (>")> --- --- <("<) --- Auto-answered --- (>")> --- However I don't hear any audio. Thanks Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have the following dialplan. Everything seems good except for one thing. If the background message is playing and the user hangs up and does not press a digit how do I run an agi on that event. I tried an exten => h,1,agi(smvoice,-digium_failed) but that was never called. I am using 1.4.10 thanks, Jerry --------------------------- [smvoice-analog] exten => s,1,Wait(1) exten => s,2,Set(TIMEOUT(absolute)=30) exten => s,3,Background(SM_PRESS_ONE_TO_HEAR_MESSAGE) exten => s,4,Goto(s,3) ; Accept any digit to cont...

1.4 was working fine. I thought I would try 1.6 beta 9 from my asteirsk 1.4 server to my asterisk client 1.6beta it wont accept the call. [Jul 18 20:34:55] NOTICE[966]: chan_sip.c:16416 handle_request_invite: Call from 'JJ' to extension 'jj_audio' rejected because extension not found. I changed nothing in the config files. I tried setting debug level to 5 and verbose to 5 all

...strange error: [Nov 3 08:32:27] NOTICE[8022]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user "404" <sip:404 at 192.168.1.8>;tag=547521CB-DB0D6130 This is the only line that prints on the console... Typically I get a few lines like: -- Executing [33 at smvoice-sip:1] Dial("SIP/404-18afe560", "SIP/bt610tmm/1044") in new stack -- Called bt610tmm/1044 -- SIP/bt610tmm-b4046c70 answered SIP/404-18afe560 -- Packet2Packet bridging SIP/404-18afe560 and SIP/bt610tmm-b4046c70 == Spawn extension (smvoice-sip, 33, 1) exited non-ze...

...ing asterisk 1.4.41 and the AMI I am trying to execute a command over AMI, specifically "core show channels concise" "sometimes" I get this back: asterisk_command_show_channels() execute failed. 'Response: Follows[CR ][LF ]Privilege: Command[CR ][LF ]OutgoingSpoolFailed!smvoice-dialout!failed!1!Down!AGI!smvoice|-digium_failed!!!3!0!(None)[LF ]' I'm not expecting to see that... My manager.conf file section is: [JJJJ] secret=YES permit=127.0.0.1/255.255.255.0 read = system,call,command,agent,user write = system,call,command,agent,user,originate ;read = system,ca...

...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...

...oes not ring? I also tried changing the canreinvite for no to yes but that made no difference after restarting. Very simple network. server, linksys router and 2 phones. 192.168.1.X for everything. Any ideas? Jerry [522] type=friend username=522 secret=522 dtmfmode=RFC2833 host=dynamic context=smvoice-sip callerid="522 522" <522> qualify=no canreinvite=no nat=no disallow=all allow=ulaw allow=alaw allow=gsm [532] type=friend username=532 secret=532 dtmfmode=RFC2833 host=dynamic context=smvoice-sip callerid=532 qualify=no canreinvite=no nat=no disallow=all allow=ulaw allow=alaw...

...h machines installed mpg123. I have a windows machine behind the firewall that plays the audio stream so firewall is not the issue. I see no errors on the CLI, I see the MusicOnHold startup like it should - just no audio. I took the setup directly from the internet setup that worked. -- Goto (smvoice-audio-streaming,s,1) -- Executing [s at smvoice-audio-streaming:1] Set("SIP/401-00000000", "CHANNEL(MUSICCLASS)=easyonhold") in new stack -- Executing [s at smvoice-audio-streaming:2] MusicOnHold("SIP/401-00000000", "") in new stack -- Started mu...

hi I am getting this error: -- Executing [50 at smvoice-sip:1] Answer("SIP/440-0856dd70", "") in new stack -- Executing [50 at smvoice-sip:2] rtsp("SIP/440-0856dd70", "rtsp://192.168.1.175/img/video.sav") in new stack [Apr 30 11:22:48] WARNING[8031]: app_rtsp.c:1037 rtsp_play: >rtsp play [Apr 30 11:22:48...

I have two calls (626 and 542) coming into the same phone (524). SIP/524-000005b5!smvoice-sip!!1!Up!AppDial!(Outgoing Line)!_2XX!!3!9!SIP/542-000005b4 SIP/542-000005b4!smvoice-sip!_2XX!8!Up!Dial!SIP/524|30|!542!!3!9!SIP/524-000005b5 SIP/524-000005b3!smvoice-sip!!1!Up!AppDial!(Outgoing Line)!_2XX!!3!40!SIP/526-000005b2 SIP/526-000005b2!smvoice-sip!_2XX!8!Up!Dial!SIP/524|30|!209!!3!40!...