search for: __ast_request_and_dial

...AGI("Local/50015308467418@default-ca2e,2", "call_log.agi|50015308467418") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi -- Zap/22-1 is proceeding passing it to Local/50015308467410@default-fb03,2 Mar 26 08:20:26 NOTICE[21367]: channel.c:2421 __ast_request_and_dial: Don't know what to do with control frame 15 -- AGI Script call_log.agi completed, returning 0 -- Executing AGI("Zap/11-1", "agi-VDADtransfer.agi|8365") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi -- AGI Script call_log....

...etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 Feb 2 14:53:09 NOTICE[18377]: chan_local.c:526 local_alloc: No such extension/c ontext 059*162*178*122*78600051 at default creating local channel Feb 2 14:53:09 NOTICE[18377]: channel.c:2514 __ast_request_and_dial: Unable to request channel Local/059*162*178*122*78600051 at default == Parsing '/etc/asterisk/manager.conf': Found ____________________________________________________________________________________________ When I call my DID, it get answered at my end but at other end , customer...

...il account is enough to use the talking bit, or do I have to register in the googletalk software again? Oh btw. Here's my error: bach >> [Oct 25 21:18:11] ERROR[28847]: chan_gtalk.c:908 gtalk_alloc: no gtalk capable clients to talk to. [Oct 25 21:18:11] NOTICE[28847]: channel.c:3243 __ast_request_and_dial: Unable to request channel gtalk/gtalk_account/my_buddy at gmail.com Kindest regards Julien -------- Music was my first love and it will be my last (John Miles) ======== FIND MY WEB-PROJECT AT: ======== http://ltsb.sourceforge.net the Linux TextBased Studio guide ======= AND MY PER...

...times out. Looking at app_parkandannounce.c /* Now place the call to the extention */ snprintf(buf, sizeof(buf), "%d", lot); memset(&oh, 0, sizeof(oh)); oh.parent_channel = chan; oh.vars = ast_variable_new("_PARKEDAT", buf); dchan = __ast_request_and_dial(dialtech, AST_FORMAT_SLINEAR, dialstr,30000, &outstate, chan->cid.cid_num, chan->cid.cid_name, &oh); I assume (I hope not incorrectly) that I have to modify the variable chan->cid.cid_name Could one of the Asterisk gurus point me in the right direction as to how to do this? Tha...

...om-conf Extension: 1000 I get the following in the CLI : [Mar 22 14:40:26] -- Attempting call on SIP/test3&SIP/test1 for 1000 at from-conf:1 (Retry 1) [Mar 22 14:40:26] WARNING[29908]: chan_sip.c:2994 create_addr: No such host: test3&SIP [Mar 22 14:40:26] NOTICE[29908]: channel.c:3046 __ast_request_and_dial: Unable to request channel SIP/test3&SIP/test1 [Mar 22 14:40:26] NOTICE[29908]: pbx_spool.c:356 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) So how can I simultaneously call different SIP-accounts from a call-file...

...lOut/12365533643|30|otT") in new stack -- Called CallOut/12365533643 -- Call accepted by 12.11.11.11 (format ulaw) -- Format for call is ulaw -- IAX2/10.11.240.110:4569-3 is proceeding passing it to Local/912365533643@default-f348,2 Jan 6 13:20:41 NOTICE[26911]: channel.c:2416 __ast_request_and_dial: Don't know what to do with control frame 15 -- IAX2/10.11.240.110:4569-3 is circuit-busy -- Hungup 'IAX2/12.11.11.11:4569-3' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Goto("Local/912365533643@default-f348,2", "s-CONGESTION|1") i...

...the following callfile: -- Attempting call on h323/101014037402962@65.17.207.253 for 9999@meetme:1 (Retry 1) Aug 19 09:18:30 WARNING[-1273607248]: channel.c:1659 ast_request: No translator path exists for channel type h323 (native 257) to 64 Aug 19 09:18:30 NOTICE[-1273607248]: channel.c:1597 __ast_request_and_dial: Unable to request channel h323/101014037402962@65.17.207.253 Aug 19 09:18:30 NOTICE[-1273607248]: pbx_spool.c:235 attempt_thread: Call failed to go through, reason 0 Callfile: MaxRetries: 2 extension: 9999 Channel: h323/101014037402962@65.17.207.253 CallerID: LAKEVIEW 4037422000 This is my h32...

...WaitTime: 600 Context: principal_in Extension: 014433988789 Priority: 1 ---------------------- I get this messages -------------------------------------- Aug 4 11:46:06 WARNING[9420]: chan_unicall.c:1240 unicall_call: Make Call failed - Blocked Aug 4 11:46:06 NOTICE[9420]: channel.c:1827 __ast_request_and_dial: Unable to request channel UniCall/g1/1 -- Hungup 'UniCall/11-1' Aug 4 11:46:06 NOTICE[9420]: pbx_spool.c:229 attempt_thread: Call failed to go through, reason 0 -------------------------------------- So i can see Unicall channels are configured but blocked (as UC show c...

...find proper notation for channel, trying unicall/1, unicall/1/1001, unicall/g1, unicall/g1/1000 and still having no luck. klaudia*CLI> !cp call /var/spool/asterisk/outgoing -- Attempting call on Unicall/1001 for application Dial(363) (Retry 1) Mar 25 09:29:34 NOTICE[19920]: channel.c:2429 __ast_request_and_dial: Unable to request channel Unicall/1001 Mar 25 09:29:34 NOTICE[19920]: pbx_spool.c:270 attempt_thread: Call failed to go through, reason 0 I use: Asterisk 1.2.4, with folowing form soft-switch: -rw-r--r-- 1 root root 314K Feb 5 17:01 libdtmfr2-20060205.tar.gz -rw-r--r-- 1 root root 346K Feb...

...doing a make and make install (and ignoring errors ;-), then firing up "asterisk -vvvc" and copying one of my test*.call files to the outgoing directory, I got: *CLI> -- Attempting call on Zap/1/12223334444 for 800@callme:2 (Retry 1) Apr 1 16:29:08 NOTICE[17424]: channel.c:1563 __ast_request_and_dial: Unable to request channel Zap/1/12223334444 Apr 1 16:29:08 NOTICE[17424]: pbx_spool.c:199 attempt_thread: Call failed to go through, reason 0 (I replaced my home/cell number with 2223334444 for illustration's sake.) Anyway, I haven't yet succeeded in digging explanations out of www.voip-...

...> Context: local > Exten: 4080915 > Priority: 1 > Callerid: testtest > Channel: OSS/dsp > Then I get: > Response: Error > Message: Originate failed and in the console: > channel.c:1828 ast_request: No channel type registered for 'OSS' > channel.c:1744 __ast_request_and_dial: Unable to request channel > OSS/dsp What went wrong? Thanks for any hints! Roger.

...isn't receiving anything in return? Ignoring this (assuming for some wacko reason, this is OKAY), I put a sample.call file in asterisk outgoing spool (Zap/g1) and asterisk very quickly traverses all channels and then gives the following error: Aug 12 03:13:20 NOTICE[245776]: channel.c:1597 __ast_request_and_dial: Unable to request channel Zap/g1/[PHONENUMBERREMOVED] Aug 12 03:13:20 NOTICE[245776]: pbx_spool.c:235 attempt_thread: Call failed to go through, reason 0 Now the question is: whatchu gonna do, when they come for you? Cheers folks, al. _________________________________________________________...

...risk/outgoing"/$(date +%Y%mNaVI%M%S)-$1 fi Unfortunately, I keep getting these errors on every voicemail that I leave: Oct 7 14:36:16 NOTICE[17889]: chan_local.c:455 local_alloc: No such extension/context 2817001@default creating local channel Oct 7 14:36:16 NOTICE[17889]: channel.c:2098 __ast_request_and_dial: Unable to request channel Local/2817001 Oct 7 14:36:16 NOTICE[17889]: pbx_spool.c:243 attempt_thread: Call failed to go through, reason 0 I know this is probably fairly easy to fix but I'm not exactly sure how the outbound call files in /var/spool/asterisk/outgoing work. Any ideas as what I...

hello, *CLI> debug level 99 /tmp/aqq Debugging level set to 99, file '/tmp/aqq' *CLI> *CLI> == Manager 'admin' logged on from 127.0.0.1 -- Requested transfer capability: 0x00 - SPEECH May 25 22:27:41 NOTICE[27407]: channel.c:2422 __ast_request_and_dial: Don't know what to do with control frame 15 > Channel Zap/1-1 was answered. > Launching txfax(/var/spool/asterisk/fax/QWWWxx61763.tif|caller) on Zap/1-1 -- Channel 0/1, span 1 got hangup request *** glibc detected *** double free or corruption (!prev): 0x081d1768 *...

...uot;Channel: Local/0123456789\@context \n"); $tn->print("Exten: 221\n"); $tn->print("Priority: 1 \n\n"); .... i get this error from asterisk cli : chan_local.c:498 local_alloc: No such extension/context 0123456789 at context creating local channel channel.c:2491 __ast_request_and_dial: Unable to request channel Local/0123456789 at context and the call doesnt start. What can i do to resolve this problem? Thanks

...ing is wrong when i use originate command to call my phone (Asterisk1.4.22 + xp100 card). Actually, i have two problems. The first one: If i fire a outgoing call using originate command directly, after my pc startup, i will receive below error message: [Nov 26 07:58:53] NOTICE[6559]: channel.c:2898 __ast_request_and_dial: Unable to request channel Zap/1/13xxxxxxxxx but i can call the FXO using my phone, everything seems perfect! After the incomming call, i fire outgoing call using originate again, it works now, my phone can ring, i also can pick up it(I seems originate did not create a new Zap channel,just used an...

...;peer not found": [Nov 19 20:01:23] ERROR[7827]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("sdf", "(null)", ...): Name or service not known [Nov 19 20:01:23] WARNING[7827]: chan_sip.c:5041 create_addr: No such host: sdf [Nov 19 20:01:23] NOTICE[7827]: channel.c:5106 __ast_request_and_dial: Unable to request channel SIP/sdf I didn't find any bug report regarding this issue. Is there any setting in sip.conf to disable host resolving in case of undefined peer name? -- Best regards, Grigoriy Puzankin

...ound_callbackservice] exten => s,1,Wait(1) exten => s,2,Playback(themessage) Asterisk logging: =============== -- Attempting call on Zap/4/0612345678 for application SetCallerPres(prohib) (Retry 1) -- Requested transfer capability: 0x00 - SPEECH Sep 2 01:37:09 NOTICE[2349]: channel.c:1865 __ast_request_and_dial: Don't know what to do with control frame 15 ---------------------------- All attempts still show the calling number to the called party. I hope anybody has some good idea because I am lost .. Kind regards, Michel -------------- next part -------------- An HTML attachment was scrubbed... UR...

Hello, I am trying to get MWI working after integrating Asterisk with CCM.I have followed the instructions in http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Voicemail+IntegrationMy problem is that I don't see externnotify's script being called at all in the logs, and not sure if I miss something here! In Voicemail general I addedpollmailboxes =

Hello I'm using an AGI script in Lua to make a callback through Zaptel. For this to work, I must wait until the channel is idle, or I get this kind of error, even after waiting over 10 seconds after the remote end rings once and hangs up: ============== channel.c:2863 __ast_request_and_dial: Unable to request channel Zap/1/123456 pbx_spool.c:341 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) ============== According to this article, CHANNEL STATUS=0 means that the line is available: www.voip-info.org/wiki/...