search for: ast_request

I'm trying to produce some enhancements to one of the applications, and am trying to use ast_request("zap", format, "pseudo") to create a new channel on /dev/zap/pseudo, which I can then bind to a zaptel conference and play a stream to it. I've been using as inspiration the Radio Repeater app, app_rpt.c, which uses this technique to play idents and announcements. Unfortun...

...n is the extention/device that calls go out on via voiptalk... (my call provider)... If I include the c/ in the TRUNK line I get... -- Executing Dial("IAX2/08700688nnn@217.14.132.nnn:4569-1", "c/07961106nnn|20|r") in new stack May 18 10:37:40 WARNING[24416]: channel.c:1957 ast_request: No channel type registered for 'c' May 18 10:37:40 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to create channel of type 'c' (cause 66) Asterisk shows this from the moment the sip channel is considered not to have answered (1 sec)... -- Nobody picked up in 1000 ms...

...t host=dynamic callerid="some name" <300> auth=md5 Then in my extensions.conf I have: exten => 300,1,Dial(IAX/${EXTEN},20) exten => 300,2,Hangup I can dial from iaxComm (a soft IAX client) and that works fine. But when I try to dial 300 get: WARNING[22077]: channel.c1970 ast_request: No channel type registered for 'IAX' NOTICE[22077]: app_dial.c:777 dial_exec: Unable to create channel of type 'IAX' I have restarted Asterisk after config change. What have I not done. I am just testing the iaxComm program. Angus

...s 10 seconds before hanging up the calling Zap channel, even though autofallthrough is set to yes. Here is the verbose output: -- Accepting call from '' to '2001' on channel 0/1, span 3 -- Executing Dial("Zap/63-1", "IAX2/sipeer/2001") in new stack == ast_request: Original IAX2 capabilities=65535, fmt=72 == ast_request: Negotiated IAX2 capabilities=64, fmt=64 -- Call accepted by 192.168.0.2 (format alaw) -- Format for call is alaw -- Called sipeer/2001 -- IAX2/sipeer-9504 is busy -- Hungup 'IAX2/sipeer-9504' == Everyone is bu...

...he meetme aplication, and just dont work.. my meetme.conf is: [rooms] conf = 700 i calling from a sip phone, the extension number is 600. there is the error: Executing [700@numberplan-custom-1:1] MeetMe("SIP/600-09111e58", "700|MI") in new stack WARNING[20055]: channel.c:3024 ast_request: No channel type registered for 'zap' WARNING[20055]: app_meetme.c:755 build_conf: Unable to open pseudo channel - trying device WARNING[20055]: app_meetme.c:758 build_conf: Unable to open pseudo device <SIP/600-09111e58> Playing 'conf-invalid' (language 'es') Spawn ex...

I think I have it configured properly. ztcfg -vv shows it as channel 1 and zttool shows it as OK. But I can't dial out. When I try, it shows it arrive in teh right stack, but then issues the following errors: channel.c:1676 ast_request: No channel type registered for '{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}' = = Everyone is busy at this time My config files are below: sip.conf [general] port = 5060 bindaddr = 0.0.0.0 disallow=all allow=gsm allow=ulaw allow=alaw allow=G723...

Hi, I have the following problem that when asterisk receives SIP response 302 it cannot forward the call I get such debug: [Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel type registered for 'Local' [Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to create local channel for call forward to 'Local/poczta at routing-sip' (cause = 66) What could be the reason for this? Thank You in advance bests -tomasz <--- SI...

Hi List, I am from Peru, I have installed an asterisk server in my company with digium card E1 TE120P, I am having issues when i make calls, here the error from my server [Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No channel type registered for 'Zap' [Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) My sources are: libpri-1.4.1.tar.gz zaptel-1.4...

...m with codecs, I guess, but I don't understand what or why. When trying to use an IAX connection to get to another office, I get: -- Executing Dial("SIP/68-4ab6", "IAX2/ast33:pass@192.168.1.130/08@from-sip") in new stack Feb 11 14:16:58 WARNING[5653]: channel.c:1898 ast_request: No translator path exists for channel type IAX2 (native 0) to 4 Feb 11 14:16:58 NOTICE[5653]: app_dial.c:800 dial_exec: Unable to create channel of type 'IAX2' (cause 0) This is particularly confounding because I have all codecs disabled except ulaw (all over, sip devices included). I...

Trying to use IAX2 extension as call center agent but getting this on the CLI May 24 20:34:20 WARNING[1209214400]: channel.c:1783 ast_request: No channel type registered for 'IAX2[2001@2001]' using AddQueueMember as the login mechanism and that seems to work but * will not deliver to the IAX2 extension. any ideas? Jason Kawakami -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium...

...hannel => 1 I am using an SNOM200 SIP phone and a TDM01B (1-port FXO bundle). When I run asterisk and dial from the SIP phone I get this error: *CLI> -- Executing Dial("SIP/555-83ee", "ZAP/1/92262802") in new stack Jul 23 13:50:24 WARNING[-267056208]: channel.c:1860 ast_request: No channel type registered for 'ZAP' Jul 23 13:50:24 NOTICE[-267056208]: app_dial.c:696 dial_exec: Unable to create channel of type 'ZAP' == Everyone is busy/congested at this time Here's my channel map: [root@pbx asterisk]# /sbin/ztcfg -vv Zaptel Configuration ====...

Hello list When i'm trying to dial into our pstn the following errors occure: -- Executing Dial("SIP/snomsip-dbd0", "/2100") in new stack Sep 9 10:02:22 WARNING[59409]: channel.c:1901 ast_request: No channel type registered for '' Sep 9 10:02:22 NOTICE[59409]: app_dial.c:715 dial_exec: Unable to create channel of type '' == Everyone is busy/congested at this time -- Executing Congestion("SIP/snomsip-dbd0", "") in new stack == Spawn extension (def...

...translate IAX2 to SIP" How can I make my IAX device communicate with a SIP device (and vice-versa)? Here's what the log says: -- Executing Dial("IAX2/myuser@myuser/15", "SIP/3422|27|tr") in new stack -- Called 3422 Oct 6 15:04:08 WARNING[-168981584]: channel.c:1879 ast_request: No translator path exists for channel type IAX2 (native 256) to 4 Oct 6 15:04:08 NOTICE[-168981584]: app_dial.c:742 dial_exec: Unable to create channel of type 'IAX2' Oct 6 15:04:08 WARNING[-168981584]: channel.c:1879 ast_request: No translator path exists for channel type IAX2 (native 2...

...L.co asterisk 1.0.1 pwlib_1.5.2 openh323_1.12.2 asterisk-oh323-0.6.3b Calling from Asterisk (2004) to the H.323phone (61-8004) gives me the following error -- Executing Dial("SIP/2004-8350", "H323/192.168.204.130") in new stack Dec 7 13:45:19 WARNING[1032209]: channel.c:1901 ast_request: No channel type registered for 'H323' Dec 7 13:45:19 NOTICE[1032209]: app_dial.c:742 dial_exec: Unable to create channel of type 'H323' == Everyone is busy/congested at this time Dec 7 13:45:29 WARNING[1032209]: pbx.c:1933 ast_pbx_run: Timeout, but no rule 't' in contex...

hi all, I have a problem with @ 1.2.4 on debian kernel 2.6.8-2-386.: -- Executing Dial("SIP/2003-bbae", "zap/2/03460816149|30|t") in new stack Feb 14 17:25:25 WARNING[4246]: channel.c:2535 ast_request: No channel type registered for 'zap' Feb 14 17:25:25 NOTICE[4246]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'zap' (cause 66 - Channel not implemented) I have a TDM400 card. with 2 channels. it seems thera are no zap channels! even in the CLI , there are...

...uhci_hcd 26256 0 rtc 10164 1 ztdummy usbcore 84740 4 ohci_hcd,ehci_hcd,uhci_hcd However, when I enter a meetme conference, I get this: -- Playing 'conf-getconfno' (language 'en') Mar 3 15:27:26 WARNING[23657]: channel.c:2535 ast_request: No channel type registered for 'zap' Mar 3 15:27:26 WARNING[23657]: app_meetme.c:461 build_conf: Unable to open pseudo channel - trying device -- Created MeetMe conference 1023 for conference '123' Uhm.... WHY? If I didn't have ztdummy installed, Asterisk would complain t...

...c/bristuff-0.3.0-PRE-1l/asterisk-1.2.4//include -I. -c AstAPIBase.c AstAPIBase.c: In function `asterisk_recordfile': AstAPIBase.c:435: warning: ISO C90 forbids mixed declarations and code AstAPIBase.c: In function `asterisk_request_and_dial': AstAPIBase.c:813: warning: passing arg 6 of `ast_request_and_dial' makes integer from pointer without a cast AstAPIBase.c:813: error: too few arguments to function `ast_request_and_dial' AstAPIBase.c: In function `asterisk_request': AstAPIBase.c:880: error: too few arguments to function `ast_request' make: *** [AstAPIBase.o] Error 1...

...0, &outstate, cid_num, cid_name); } static struct ast_channel *callback_request_and_dial(struct ast_channel *caller, const char *type, int format, void *data, int timeout, int *outstate, const char *cid_num, const char *cid_name) { int cause = 0; struct ast_channel *chan; if ((chan = ast_request(type, format, data, &cause))) { ast_set_callerid(chan, cid_num, cid_name, cid_num); ast_channel_inherit_variables(caller, chan); printfl("\n\n In if ((chan = ast_request(type, format, data, &cause)))"); if (!ast_call(chan, data, timeout)) { dosomething; } dosom...

Has anyone been able to successfully solve the following issue: WARNING[21725]: channel.c:3024 ast_request: No channel type registered for 'Zap' [Mar 11 01:26:53] WARNING[21725]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) Since we updated asterisk from 1.2.13 to asterisk 1.2.16 the module went away so we updated to 1.4....

...ing and installing pwlib and openh323 , the asterisk, give the folloing error. please tell me where the problem is ? Best Mani *CLI> -- Executing Dial("SIP/2.2.2.2-086f5ac0", "H323/652#150388590962@1.1.1.1|60") in new stack Mar 28 14:17:23 WARNING[11985]: channel.c:2576 ast_request: No translator path exists for channel type H323 (native 4) to 256 Mar 28 14:17:23 NOTICE[11985]: app_dial.c:1059 dial_exec_full: Unable to create channel of type 'H323' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/2.2.2....