tnaks for your response but the number dialed exist and i can call this number when i configure the trunk directly in x-lite and i call call also this number from my cell phone . any help thanks and regards
2015-03-25 12:59 GMT+00:00 Matthew Jordan <[email protected]>: > On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit > <[email protected]> wrote: > > hello list, > > > > i have asterisk 11.15.0 and i have some trunks sip from my provider > > > > we have some ip phone astra 6731i > > > > each Ip-phone is configured with trunk and we call > > > > no ihave configured another trunk from the same provider in my asterisk > > > > i can call all numbers just the numbers are configured in thses ip > phones. > > > > but when i configured the same trunk in x-lite i can call theses > ip-phones > > without issue > > the problem just when i configure the trunk in my server and i use > > extension > > > > all the ip-phone and x-lite and server asterisk in the same network > > 192.168.1.x > > > > == Using SIP RTP TOS bits 184 > > == Using SIP RTP CoS mark 5 > > -- Called SIP/FD/0033149XXXXXX > > -- SIP/FD-000000b9 is making progress passing it to SIP/306-000000b8 > > > 0x2afec424c430 -- Probation passed - setting RTP source address > to > > 192.168.1.212:57592 > > > 0xc5922b0 -- Probation passed - setting RTP source address to > > 217.195.xx.xxx:29674 > > -- Got SIP response 556 "No address found" back from > 217.195.XX.XXX:5060 > > == Everyone is busy/congested at this time (1:0/1/0) > > -- Executing [s@macro-dialout-trunk:23] NoOp("SIP/306-000000b8", > "Dial > > failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = > 34") > > in new stack > > -- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/306-000000b8", > > "0?continue,1:s-CONGESTION,1") in new stack > > -- Goto (macro-dialout-trunk,s-CONGESTION,1) > > -- Executing [s-CONGESTION@macro-dialout-trunk:1] > > Set("SIP/306-000000b8", "RC=34") in new stack > > -- Executing [s-CONGESTION@macro-dialout-trunk:2] > > Goto("SIP/306-000000b8", "34,1") in new stack > > -- Goto (macro-dialout-trunk,34,1) > > -- Executing [34@macro-dialout-trunk:1] Goto("SIP/306-000000b8", > > "continue,1") in new stack > > -- Goto (macro-dialout-trunk,continue,1) > > -- Executing [continue@macro-dialout-trunk:1] > NoOp("SIP/306-000000b8", > > "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to > > other trunks") in new stack > > -- Executing [continue@macro-dialout-trunk:2] > Set("SIP/306-000000b8", > > "CALLERID(number)=306") in new stack > > -- Executing [0149XXXXXX@from-internal:7] Macro("SIP/306-000000b8", > > "outisbusy,") in new stack > > -- Executing [s@macro-outisbusy:1] Progress("SIP/306-000000b8", "") > in > > new stack > > -- Executing [s@macro-outisbusy:2] GotoIf("SIP/306-000000b8", > > "0?emergency,1") in new stack > > -- Executing [s@macro-outisbusy:3] GotoIf("SIP/306-000000b8", > > "0?intracompany,1") in new stack > > -- Executing [s@macro-outisbusy:4] Playback("SIP/306-000000b8", > > "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack > > [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: file.c:701 > > ast_openstream_full: File all-circuits-busy-now does not exist in any > format > > [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: file.c:1017 > > ast_streamfile: Unable to open all-circuits-busy-now (format (ulaw)): No > > such file or directory > > [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: app_playback.c:484 > > playback_exec: ast_streamfile failed on SIP/306-000000b8 for > > all-circuits-busy-now&pls-try-call-later, noanswer > > [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: file.c:701 > > ast_openstream_full: File pls-try-call-later does not exist in any format > > [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: file.c:1017 > > ast_streamfile: Unable to open pls-try-call-later (format (ulaw)): No > such > > file or directory > > [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: app_playback.c:484 > > playback_exec: ast_streamfile failed on SIP/306-000000b8 for > > all-circuits-busy-now&pls-try-call-later, noanswer > > -- Executing [s@macro-outisbusy:5] Congestion("SIP/306-000000b8", > "20") > > in new stack > > [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: channel.c:4862 > ast_prod: > > Prodding channel 'SIP/306-000000b8' failed > > == Spawn extension (macro-outisbusy, s, 5) exited non-zero on > > 'SIP/306-000000b8' in macro 'outisbusy' > > == Spawn extension (from-internal, 0149XXXXXX, 7) exited non-zero on > > 'SIP/306-000000b8' > > -- Executing [h@from-internal:1] Hangup("SIP/306-000000b8", "") in > new > > stack > > == Spawn extension (from-internal, h, 1) exited non-zero on > > 'SIP/306-000000b8' > > == MixMonitor close filestream (mixed) > > == End MixMonitor Recording SIP/306-000000b8 > > > > The verbose output states why your call is congested: > > -- Got SIP response 556 "No address found" back from > 217.195.XX.XXX:5060 > > The far end came back with a 556 response to the outbound INVITE > request. It doesn't think that whatever you dialled exists. > > -- > Matthew Jordan > Digium, Inc. | Director of Technology > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com & http://asterisk.org > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
