search for: attempt_thread

...CLI SNIPPET------------// -- Attempting call on DAHDI/g1/918005551212 for 12564286000@ xs-globx-ds3:1 (Retry 1) -- Requested transfer capability: 0x00 - SPEECH -- PROGRESS with cause code 31 received -- Hungup 'DAHDI/1-1' [2011-06-15 01:35:14] NOTICE[27176]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (3) Remote end Ringing -- Attempting call on DAHDI/g1/918005551212 for 12564286000 at xs-globx-ds3:1 (Retry 2) -- Requested transfer capability: 0x00 - SPEECH -- PROGRESS with cause code 31 received -- Hungup 'DAHDI/1-1' [2011-06-15 01:35:2...

...at when I force a reload using 'amportal a reload' I see this loop start in 'asterisk -rvvvvvvvvvv': > Channel Local/s at tc-maint-000002a4;1 was answered. > Launching NoCDR() on Local/s at tc-maint-000002a4;1 [2013-06-24 15:22:32] NOTICE[32678]: pbx_spool.c:402 attempt_thread: Call completed to Local/s at tc-maint [2013-06-24 15:22:32] NOTICE[32678]: pbx_spool.c:402 attempt_thread: Call completed to Local/s at tc-maint == Spawn extension (tc-maint, s, 5) exited non-zero on 'Local/s at tc-maint-000002a4;2' -- Attempting call on Local/s at tc-maint for appli...

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

...273607248]: 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 h323.conf file: ; The NuFone Network's ; Open H.323 driver configuration ; [general] port = 1720 bindaddr = 0.0.0.0 disallow=all...

...ion: $CALLERID Callerid: <my home phone number> Context: out WaitTime: 15 MaxRetries: 9999 RetryTime: 20 Priority: 1 EOF mv $TEMP /var/spool/asterisk/outgoing/ -----8<------ Some cut log output to show the timing: -----8<------ [2020-04-23 13:05:38.104] NOTICE[23290]: pbx_spool.c:450 attempt_thread: Call failed to go through, reason (5) Remote end is Busy -- Attempting call on xxxxxx at out:1 (Retry 327) -- Called xxxxxxx -- Call accepted by 1.2.3.4:4569 (format alaw) -- Format for call is (alaw) -- xxxxxxxxxxxxxxxxxx is making progress -- xxxxxxxxxxxxxxxxxx is busy...

...mvoice-dialout:1] AGI("OutgoingSpoolFailed", "smvoice") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice == Spawn extension (smvoice-dialout, failed, 1) exited non-zero on 'OutgoingSpoolFailed' [Aug 7 09:20:56] NOTICE[11272]: pbx_spool.c:341 attempt_thread: Call failed to go through, reason (5) Remote end is Busy

...> Launching SMS(0) on mISDN/2-u11 -- SMS RX 93 00 6D -- SMS TX 91 1C 01 0B 0C 81 70 21 41 43 58 03 00 F1 11 C4 74 79 0E 4A CF E9 A0 72 DA 0D A2 96 E7 74... -- SMS RX 95 09 01 00 70 20 82 12 55 70 40 38 -- SMS TX 94 00 6C Feb 28 21:55:09 NOTICE[1963]: pbx_spool.c:279 attempt_thread: Call completed to mISDN/g:extern/0193010 Situation 2 =========== -- Attempting call on mISDN/g:extern/0193010 for application SMS(0) (Retry 1) > Channel mISDN/2-u10 was answered. > Launching SMS(0) on mISDN/2-u10 -- SMS RX 93 00 6D -- SMS TX 91 1C 01 0A 0C...

...(0) on CAPI/ISDN4#02/0809101000-1 -- SMS RX 93 00 6D -- SMS TX 91 0D 01 01 0A 81 60 70 93 66 66 00 F1 01 58 5C -- SMS RX 96 0A 01 A0 00 70 90 31 51 54 45 80 24 -- SMS TX 94 00 6C == ISDN4#02: CAPI Hangingup for PLCI=0x104 in state 2 [Sep 13 15:45:58] NOTICE[23584]: pbx_spool.c:371 attempt_thread: Call completed to CAPI/g1/0809101000 > ISDN4#02: CAPI INFO 0x3495: Call rejected As I understand it the "RX 96" is FT NAK'ing my "TX 91" "DELIVER". Any idea what I'm doing wrong?

...I taken when I place a call in the spool directory looks like this: -- Attempting call on SIP/MTN-NEW/my-number for application MP3Player(/myfile) (Retry 1) == Using SIP RTP CoS mark 5 > Channel SIP/MTN-NEW-00000005 was never answered. [Jul 20 10:52:11] NOTICE[14580]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (8) Congestion (circuits busy) My sip.conf looks like this: [MTN-NEW] host=192.168.34.1 disallow=all allow=ilbc allow=gsm allow=g729 allow=g723 allow=ulaw allow=g729 type=peer My SIP provider says that no traffic is picked up at their SBC or on the WAN gatewa...

...st1 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 ?? Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com...

...m("DAHDI/1-1", "env echo -e "Channel:DAHDI/g0/...........\\nContext:fax-confirm-tx\\nExtension: s\\nPriority: 1\\n" >/var/spool/asterisk/outgoing/call-1254077933.390") in new stack -- Hungup 'DAHDI/1-1' [Sep 27 14:59:24] NOTICE[8413]: pbx_spool.c:357 attempt_thread: Call completed to DAHDI/g0/.............. -- Attempting call on DAHDI/g0/............. for s at fax-confirm-tx:1 (Retry 1) -- Requested transfer capability: 0x00 - SPEECH > Channel DAHDI/1-1 was answered. -- Executing [s at fax-confirm-tx:1] SendFAX("DAHDI/1-1&quot...

...> Channel Zap/1-1 was answered. == Starting Zap/1-1 at callbacktest,start,1 failed so falling back to exten 's' == Starting Zap/1-1 at callbacktest,s,1 still failed so falling back to context 'default' -- Hungup 'Zap/1-1' [Mar 15 16:22:11] NOTICE[368]: pbx_spool.c:351 attempt_thread: Call completed to Zap/1/5551234 ====== I followed this tutorial, and don't understand why Asterisk tries to jump to extension "s": www.voip-info.org/wiki/view/Asterisk+auto-dial+out Thank you.

...00... -- SMS RX 92 01 FF 6E 00 00... -- SMS TX 92 01 FF 6E 00 00... -- Executing Hangup("Zap/3-1", "") in new stack == Spawn extension (smsdial, s, 3) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' Dec 2 09:46:45 NOTICE[9188]: pbx_spool.c:242 attempt_thread: Call completed to Zap/g1/9,5375345 And this is logdata from /var/log/asterisk/event_log: Dec 2 09:45:46 asterisk[8554]: Received to /var/spool/asterisk/sms/sc-me.5375321/2004-12-02_09:45:46-00 Dec 2 09:46:45 asterisk[8554]: No data, hanging up Dec 2 09:46:45 asterisk[855...

...person says they can connect a software as extension 4452 and it works just fine. I have my register: register => 4452 at X.X.X.X/4452 [4452] type=friend username=4452 host=X.X.X.X allow=all dtmfmode=inband When I try to make the call I get : [Jan 28 08:05:57] NOTICE[44761]: pbx_spool.c:447 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) [Jan 28 08:05:57] NOTICE[44761]: pbx_spool.c:450 attempt_thread: Queued call to SIP/4452/7XXXXXXXXX expired without completion after 0 attempts (where 7 is the outside line). The exten 4...

...t I see on the console: -- Executing [failed at Dialout:1] NoOp("OutgoingSpoolFailed", "======== Call to 155555555 failed ") in new stack == Auto fallthrough, channel 'OutgoingSpoolFailed' status is 'UNKNOWN' [Oct 9 14:16:50] NOTICE[29172]: pbx_spool.c:341 attempt_thread: Call failed to go through, reason 0 [Oct 9 14:16:50] WARNING[15708]: pbx_spool.c:245 apply_outgoing: At least one of app or extension must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/t.call Does this just not work? Or am I missing something? Thanks in ad...

...g [fax@incomingfax:1] Set("SIP/xxx.xxx.xx.xx-08237608", "FAXFILE=/tmp/recievedfax.tif") in new stack -- Executing [fax@incomingfax:2] RxFAX("SIP/xxx.xxx.xx.xx-08237608", "/tmp/recievedfax.tif") in new stack [Nov 20 14:00:57] NOTICE[19864]: pbx_spool.c:341 attempt_thread: Call failed to go through, reason 0 -- Attempting call on SIP/sipfaxline/(asteriskfaxnumber) for out_fax@outgoingfax:1 (Retry 2) -- Executing [(asteriskfaxnumber)@dids-inbound:1] Goto("SIP/xxx.xxx.xx.xx-0824ddf8", "incomingfax|fax|1") in new stack -- Goto (incomingf...

...-- 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 channel). There is not much info about unicall so i require your advice, what can i do? Where do i look? Also i constantly receive messages Aug...

...ing 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 5 16:57 libmfcr2-20060205.tar.gz -rw-r--r-- 1 root root 300K Feb 5 16:54 libsupertone-20060205.tar.gz...

...se_invite: Re-invite to non-existing call leg on other UA. SIP dialog '240370c953d10a75430c0e2e0d4764a6 at sip.neophonex.hu'. Giving up. == Spawn extension (internal, 95206301111111, 3) exited non-zero on 'SIP/neophonex-081ab240' [Jun 25 16:57:10] NOTICE[18440]: pbx_spool.c:351 attempt_thread: Call completed to SIP/06202222222 at neophonex Please help me to figure out why the calls are hung up. Thanks Adam

..."Forbidden" from '"Edward Jones" <sip:3145152000 at 192.168.122.57>;tag=as4eecf94f' Scheduling destruction of SIP dialog '504b8ce74a81e0f90ba457e77e8c9e60 at 192.168.122.57:5060' in 32000 ms (Method: INVITE) [Apr 23 08:20:59] NOTICE[19157]: pbx_spool.c:389 attempt_thread: Call failed to go through, reason (1) Hangup [Apr 23 08:20:59] NOTICE[19157]: pbx_spool.c:392 attempt_thread: Queued call to SIP/SMtrunk/913145152244 expired without completion after 0 attempts Thanks, Scott Haley IS Voice Projects Team Edward Jones Investments Phone: 314-515-2244 Email: scott.ha...