On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit <salah.elharit...@gmail.com> 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