<div dir="ltr">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 .<div>any help</div><div>thanks and regards</div></div><div class="gmail_extra"><br><div class="gmail_quote">2015-03-25 12:59 GMT+00:00 Matthew Jordan <span dir="ltr"><<a href="mailto:mjordan@digium.com" target="_blank">mjordan@digium.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class="HOEnZb"><div class="h5">On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit<br>
<<a href="mailto:salah.elharit200@gmail.com">salah.elharit200@gmail.com</a>> wrote:<br>
> hello list,<br>
><br>
> i have asterisk 11.15.0 and i have some trunks sip from my provider<br>
><br>
> we have some ip phone astra 6731i<br>
><br>
> each Ip-phone is configured with trunk and we call<br>
><br>
> no ihave configured another trunk from the same provider in my asterisk<br>
><br>
> i can call all numbers just the numbers are configured in thses ip phones.<br>
><br>
> but when i configured the same trunk in x-lite i can call theses ip-phones<br>
> without issue<br>
> the problem just when i configure the trunk in my server and i use<br>
> extension<br>
><br>
> all the ip-phone and x-lite and server asterisk in the same network<br>
> 192.168.1.x<br>
><br>
> == Using SIP RTP TOS bits 184<br>
> == Using SIP RTP CoS mark 5<br>
> -- Called SIP/FD/0033149XXXXXX<br>
> -- SIP/FD-000000b9 is making progress passing it to SIP/306-000000b8<br>
> > 0x2afec424c430 -- Probation passed - setting RTP source address to<br>
> <a href="http://192.168.1.212:57592" target="_blank">192.168.1.212:57592</a><br>
> > 0xc5922b0 -- Probation passed - setting RTP source address to<br>
> 217.195.xx.xxx:29674<br>
> -- Got SIP response 556 "No address found" back from 217.195.XX.XXX:5060<br>
> == Everyone is busy/congested at this time (1:0/1/0)<br>
> -- Executing [s@macro-dialout-trunk:23] NoOp("SIP/306-000000b8", "Dial<br>
> failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34")<br>
> in new stack<br>
> -- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/306-000000b8",<br>
> "0?continue,1:s-CONGESTION,1") in new stack<br>
> -- Goto (macro-dialout-trunk,s-CONGESTION,1)<br>
> -- Executing [s-CONGESTION@macro-dialout-trunk:1]<br>
> Set("SIP/306-000000b8", "RC=34") in new stack<br>
> -- Executing [s-CONGESTION@macro-dialout-trunk:2]<br>
> Goto("SIP/306-000000b8", "34,1") in new stack<br>
> -- Goto (macro-dialout-trunk,34,1)<br>
> -- Executing [34@macro-dialout-trunk:1] Goto("SIP/306-000000b8",<br>
> "continue,1") in new stack<br>
> -- Goto (macro-dialout-trunk,continue,1)<br>
> -- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/306-000000b8",<br>
> "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to<br>
> other trunks") in new stack<br>
> -- Executing [continue@macro-dialout-trunk:2] Set("SIP/306-000000b8",<br>
> "CALLERID(number)=306") in new stack<br>
> -- Executing [0149XXXXXX@from-internal:7] Macro("SIP/306-000000b8",<br>
> "outisbusy,") in new stack<br>
> -- Executing [s@macro-outisbusy:1] Progress("SIP/306-000000b8", "") in<br>
> new stack<br>
> -- Executing [s@macro-outisbusy:2] GotoIf("SIP/306-000000b8",<br>
> "0?emergency,1") in new stack<br>
> -- Executing [s@macro-outisbusy:3] GotoIf("SIP/306-000000b8",<br>
> "0?intracompany,1") in new stack<br>
> -- Executing [s@macro-outisbusy:4] Playback("SIP/306-000000b8",<br>
> "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack<br>
> [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: file.c:701<br>
> ast_openstream_full: File all-circuits-busy-now does not exist in any format<br>
> [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: file.c:1017<br>
> ast_streamfile: Unable to open all-circuits-busy-now (format (ulaw)): No<br>
> such file or directory<br>
> [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: app_playback.c:484<br>
> playback_exec: ast_streamfile failed on SIP/306-000000b8 for<br>
> all-circuits-busy-now&pls-try-call-later, noanswer<br>
> [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: file.c:701<br>
> ast_openstream_full: File pls-try-call-later does not exist in any format<br>
> [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: file.c:1017<br>
> ast_streamfile: Unable to open pls-try-call-later (format (ulaw)): No such<br>
> file or directory<br>
> [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: app_playback.c:484<br>
> playback_exec: ast_streamfile failed on SIP/306-000000b8 for<br>
> all-circuits-busy-now&pls-try-call-later, noanswer<br>
> -- Executing [s@macro-outisbusy:5] Congestion("SIP/306-000000b8", "20")<br>
> in new stack<br>
> [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: channel.c:4862 ast_prod:<br>
> Prodding channel 'SIP/306-000000b8' failed<br>
> == Spawn extension (macro-outisbusy, s, 5) exited non-zero on<br>
> 'SIP/306-000000b8' in macro 'outisbusy'<br>
> == Spawn extension (from-internal, 0149XXXXXX, 7) exited non-zero on<br>
> 'SIP/306-000000b8'<br>
> -- Executing [h@from-internal:1] Hangup("SIP/306-000000b8", "") in new<br>
> stack<br>
> == Spawn extension (from-internal, h, 1) exited non-zero on<br>
> 'SIP/306-000000b8'<br>
> == MixMonitor close filestream (mixed)<br>
> == End MixMonitor Recording SIP/306-000000b8<br>
><br>
<br>
</div></div>The verbose output states why your call is congested:<br>
<span class=""><br>
-- Got SIP response 556 "No address found" back from 217.195.XX.XXX:5060<br>
<br>
</span>The far end came back with a 556 response to the outbound INVITE<br>
request. It doesn't think that whatever you dialled exists.<br>
<span class="HOEnZb"><font color="#888888"><br>
--<br>
Matthew Jordan<br>
Digium, Inc. | Director of Technology<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> & <a href="http://asterisk.org" target="_blank">http://asterisk.org</a><br>
<br>
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</font></span></blockquote></div><br></div>