search for: dial_exec

...39;Zap/1-1' Feb 3 12:11:17 NOTICE[6424]: chan_zap.c:5363 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial("Zap/1-1", "SIP/2001&SIP/2002&IAX2/2003&IAX2/2101&SIP/2102&SIP/2103|10") in new stack Feb 3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' Feb 3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' Feb 3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create channel of type 'IAX2' -- Called 2101 Feb 3 12:11:17 NOTIC...

...> sip show registry Host Username Refresh State 192.168.7.16:5060 7062210 105 Registered -- Executing Dial("SIP/192.168.7.14-085a4790", "Zap/1/67742142") in new stack Nov 30 01:41:52 NOTICE[507921]: app_dial.c:742 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time -- Executing Dial("SIP/192.168.7.14-085a4790", "Zap/1/61002663") in new stack Nov 30 01:41:52 NOTICE[507921]: app_dial.c:742 dial_exec: Unable to create channel of type 'Zap...

...u ok, second concurrent call fails with: Aug 6 11:52:30 DEBUG[737292]: chan_h323.c:1038 setup_incoming_call: Sending to context [ip2pri] -- Executing Dial("H323/ip$192.168.32.25:60271/984", "Zap/1/9541163107100") in new stack Aug 6 11:52:30 NOTICE[753677]: app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' Aug 6 11:52:30 NOTICE[753677]: app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time Aug 6 11:52:40 WARNING[753677]: pbx.c:1836 ast_pbx_run: Timeout, but no rule 't' in context 'ip2...

...irst and second one works, but on the third connection this happens: -- Executing Dial("SIP/gunnar-26ea", "CAPI/7501:7986:bBYEXTENSION") in new stack chan_capi.c:1147 capi_request: didn't find capi device with outgoing msn = 7501. you should check your config! app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI' == Everyone is busy at this time -- Executing Dial("SIP/gunnar-26ea", "CAPI/7502:7986:bBYEXTENSION") in new stack chan_capi.c:1147 capi_request: didn't find capi device with outgoing msn = 7502. you should check your config! a...

Hi, I would like to use a digum card to call an external number through my PSTN. I think that I have a problem in the configuration. Asterisk returns me app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' I use Fedora core 3. I installed libpri, zaptel and asterisk. I plugged my line on the FXS module (green part). I make modprobe zaptel && modprobe wctdm without error return. Configuration files : /etc/zaptel.conf fxsks=1 loadzone = it defa...

asterisk-users Any help will be appreciated. This card did not connect with E1 line how to loop E400P card to test ? now I loop the card. span 1 ---span2 RJ45 pins 1--4 2--5 but show : When calling ,showing error: app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' Asterisk Ready. *CLI> -- Registered SIP '2002' at 192.168.139.59 port 3289 expires 120 -- Executing Dial("SIP/2002-a4af", "Zap/1") in new stack Jul 28 15:51:25 NOTICE[1910]: app_dial.c:764 dial_exec: Unable to...

....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

...card to work with my PRI. When I attempt to make an outbound call via: exten => 1004,1,Dial(Zap/g1/NPANXXXXXX) I see the following on the asterisk console: -- Executing Dial("SIP/sbruton-b8ce", "Zap/g1/NPANXXXXXX") in new stack Apr 21 08:18:48 NOTICE[16401]: app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time both the zaptel and wct1xxp modules are loaded and ztcfg reports no errors: # ztcfg -v Zaptel Configuration ====================== SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) 24 channels configured....

..., tv_usec = 0}) at channel.c:3260 #9 0x080655fd in ast_channel_bridge (c0=0xb659fcd0, c1=0x9455ca0, config=0xb6c4feb0, fo=0xb6c4f988, rc=0xb6c4f98c) at channel.c:3524 #10 0xb78fad29 in ast_bridge_call (chan=0xb659fcd0, peer=0x9455ca0, config=0xb6c4feb0) at res_features.c:1319 #11 0xb7099301 in dial_exec_full (chan=0xb659fcd0, data=0xb6c4feb0, peerflags=0xb6c50568) at app_dial.c:1577 #12 0xb7097dc5 in dial_exec (chan=0xb7ed1900, data=0xb7ed1900) at app_dial.c:1619 #13 0x0808e445 in pbx_extension_helper (c=0xb659fcd0, con=0xb7ed1900, context=0xb659fe20 "op05_x", exten=0xb659ff14 "0011...

..., class 'default', on Zap/3-1 -- Attempting native bridge of Zap/2-1 and Zap/3-1 -- Executing Macro("Zap/2-2", "dial-FXO-only|7351056") in new stack -- Executing Dial("Zap/2-2", "ZAP/3/7351056") in new stack Sep 2 16:14:16 NOTICE[2652]: app_dial.c:764 dial_exec: Unable to create channel of type 'ZAP' == Everyone is busy/congested at this time -- Executing Dial("Zap/2-2", "ZAP/4/7351056") in new stack -- Called 4/7351056 -- Zap/4-1 answered Zap/2-2 -- Attempting native bridge of Zap/2-2 and Zap/4-1 -- Building conference on cal...

...g 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 256) to 4 Oct 6 15:04:08 NOTICE[-168981584]: app_dial.c:742 dial_exec: Unable to create channel of type 'IAX2' -- SIP/34...

...uestions posed here. I am wanting to pass the Zap channel that the original call came from to my macro embedded in the Dial command. I've tried to add arguments to the macro by using the syntax M(x,arg1), and I always get the following errors. Dec 2 15:57:53 WARNING[96234416]: app_dial.c:629 dial_exec: Could not find macro to which we should jump. Dec 2 15:57:53 WARNING[96234416]: app_dial.c:636 dial_exec: Macro flag set without trailing ') Is there a patch that would allow me to do this? Or is there a way to get the Zap channel name that requested the Dial on the outbound trunk from with...

...seTimeout("Zap/4-1", "10") in new stack -- Set Response Timeout to 10 -- Executing Dial("Zap/4-1", "Zap/1&SIP/3014&SIP/3016&SIP/3017|35|tr") in new stack -- Called 1 -- Called 3014 Mar 27 17:27:15 NOTICE[10890]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' -- Called 3017 -- SIP/3017-ba4c is ringing -- Zap/1-1 is ringing -- SIP/3014-556e is ringing -- Zap/1-1 answered Zap/4-1 -- Attempting native bridge of Zap/4-1 and Zap/1-1 -- Attempting native bridge of Zap/4-1 and Z...

...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=G723.1 context=from-sip [2000] type=friend username=2000 secret=xxxxxxxx host=d...

...es, everything is fine. Should the connection go down, and then come back again, those behind a NAT are still able to make calls, but are unable to receive calls. -- Executing Dial("SIP/1239-ba74", "SIP/1242|60|t") in new stack Jan 12 23:45:19 NOTICE[21576]: app_dial.c:803 dial_exec: Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time (1:0/1/0) However, extension 1242 is still able to call 1239? Is this a configuration problem in Asterisk or in the phones? Please help Regards David Norton -------------- next part ---...

...API/0721111:b${EXTEN:1}) When I make a call I receive : -- Executing Dial("SIP/2111-9940", "CAPI/0721111:b335111111") in new stack -- data =0721111:b335111111 -- capi request omsn =0721111 == found capi with omsn =0721111 May 28 10:36:56 NOTICE[180241]: app_dial.c:655 dial_exec: Unable to create channel of type 'CAPI' == Everyone is busy at this time Someone can help me ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040528/9fc6d9a9/attachment.htm

...decide(s) to go on permanent busy. Dialing that phone gives: -- Executing Macro("SIP/24567-7856", "dialphone|SIP/27654") in new stack -- Executing Dial("SIP/24567-7856", "SIP/27654|10|tr") in new stack Jun 15 14:23:41 NOTICE[1343506]: app_dial.c:536 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy at this time But dialing in the other direction (from the busy phone out) gives normal (good) results: -- Executing Macro("SIP/27654-6e2b", "dialphone|SIP/24567") in new stack -- Executing Dial(&quo...

I recently moved my Asterisk configuration to a new server and re-built Asterisk from CVS. Now, I'm experiencing the following issue with SIP: Executing Dial("Zap/1-1", "SIP/100|20") in new stack NOTICE[-1232077904]: File app_dial.c, Line 518 (dial_exec): Unable to create channel of type 'SIP' == Everyone is busy at this time Has anyone seen this issue before? Thanks.

...rnel module, and tried to use it to make an outgoing call (via a phone connected to an ATA-186). However, I just get a reorder tone and see this on the console: -- Executing Dial("SIP/ata1-4409", "Zap/1/5551212") in new stack NOTICE[1200825920]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type 'Zap' == Everyone is busy at this time -- Executing Congestion("SIP/ata1-4409", "") in new stack My zapata.conf file is unchanged from the default, which seemed to be appropriate for the X100P in this usage. It's not clear...

...k -- Goto (default,100,1) -- Executing AGI("IAX2[callfwd2@misc1]/16386", "exm") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/exm -- AGI Script Executing Application: (DIAL) Options: (Zap/23|12|m) Jun 7 13:28:30 NOTICE[730128]: app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time -- AGI Script exm completed, returning 0 -- Hungup 'IAX2[callfwd2@misc1]/16386' At this time, the caller goes to the start (s) extension and keep trying to call somebody up to hangup. I'm cu...