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[salah.elharit200 at gm...]

hello list

i have an issue related to outbound calls i can contact all the number except on number given by our provider in trunk 


the issue just when i configure my trunk in our server but when i configure the trunk directly in x-lite i can contact this number without issue


below the cli


  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [0149xxxxxx@from-internal:1] Macro("SIP/101-00000103", "user-callerid,LIMIT,EXTERNAL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/101-00000103", "TOUCH_MONITOR=1426869820.301") in new stack
    -- Executing [s@macro-user-callerid:2] Set("SIP/101-00000103", "AMPUSER=101") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("SIP/101-00000103", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("SIP/101-00000103", "1?Set(REALCALLERIDNUM=101)") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/101-00000103", "AMPUSER=101") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/101-00000103", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/101-00000103", "AMPUSERCIDNAME=101") in new stack
    -- Executing [s@macro-user-callerid:8] GotoIf("SIP/101-00000103", "0?report") in new stack
    -- Executing [s@macro-user-callerid:9] Set("SIP/101-00000103", "AMPUSERCID=101") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/101-00000103", "__DIAL_OPTIONS=tr") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/101-00000103", "CALLERID(all)="101" <101>") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("SIP/101-00000103", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:13] ExecIf("SIP/101-00000103", "1?Set(GROUP(concurrency_limit)=101)") in new stack
    -- Executing [s@macro-user-callerid:14] ExecIf("SIP/101-00000103", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:15] GotoIf("SIP/101-00000103", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,2
    -- Executing [s@macro-user-callerid:28] Set("SIP/101-00000103", "CALLERID(number)=101") in new stack
    -- Executing [s@macro-user-callerid:29] Set("SIP/101-00000103", "CALLERID(name)=101") in new stack
    -- Executing [s@macro-user-callerid:30] Set("SIP/101-00000103", "CDR(cnum)=101") in new stack
    -- Executing [s@macro-user-callerid:31] Set("SIP/101-00000103", "CDR(cnam)=101") in new stack
    -- Executing [s@macro-user-callerid:32] Set("SIP/101-00000103", "CHANNEL(language)=en") in new stack
    -- Executing [0149xxxxxx@from-internal:2] Set("SIP/101-00000103", "MOHCLASS=default") in new stack
    -- Executing [0149xxxxxx@from-internal:3] Set("SIP/101-00000103", "_NODEST=") in new stack
    -- Executing [0149xxxxxx@from-internal:4] Gosub("SIP/101-00000103", "sub-record-check,s,1(out,0149xxxxxx,)") in new stack
    -- Executing [s@sub-record-check:1] Set("SIP/101-00000103", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [s@sub-record-check:2] GotoIf("SIP/101-00000103", "1?check") in new stack
    -- Goto (sub-record-check,s,7)
    -- Executing [s@sub-record-check:7] Set("SIP/101-00000103", "__MON_FMT=wav") in new stack
    -- Executing [s@sub-record-check:8] GotoIf("SIP/101-00000103", "1?next") in new stack
    -- Goto (sub-record-check,s,11)
    -- Executing [s@sub-record-check:11] ExecIf("SIP/101-00000103", "0?Return()") in new stack
    -- Executing [s@sub-record-check:12] ExecIf("SIP/101-00000103", "0?Set(__REC_POLICY_MODE=)") in new stack
    -- Executing [s@sub-record-check:13] GotoIf("SIP/101-00000103", "0?out,1") in new stack
    -- Executing [s@sub-record-check:14] Set("SIP/101-00000103", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s@sub-record-check:15] Set("SIP/101-00000103", "NOW=1426869820") in new stack
    -- Executing [s@sub-record-check:16] Set("SIP/101-00000103", "__DAY=20") in new stack
    -- Executing [s@sub-record-check:17] Set("SIP/101-00000103", "__MONTH=03") in new stack
    -- Executing [s@sub-record-check:18] Set("SIP/101-00000103", "__YEAR=2015") in new stack
    -- Executing [s@sub-record-check:19] Set("SIP/101-00000103", "__TIMESTR=20150320-164340") in new stack
    -- Executing [s@sub-record-check:20] Set("SIP/101-00000103", "__FROMEXTEN=101") in new stack
    -- Executing [s@sub-record-check:21] Set("SIP/101-00000103", "__CALLFILENAME=out-0149xxxxxx-101-20150320-164340-1426869820.301") in new stack
    -- Executing [s@sub-record-check:22] Goto("SIP/101-00000103", "out,1") in new stack
    -- Goto (sub-record-check,out,1)
    -- Executing [out@sub-record-check:1] ExecIf("SIP/101-00000103", "1?Set(__REC_POLICY_MODE=always)") in new stack
    -- Executing [out@sub-record-check:2] GosubIf("SIP/101-00000103", "1?record,1(exten,0149xxxxxx,101)") in new stack
    -- Executing [record@sub-record-check:1] Set("SIP/101-00000103", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
    -- Executing [record@sub-record-check:2] MixMonitor("SIP/101-00000103", "2015/03/20/out-0149xxxxxx-101-20150320-164340-1426869820.301.wav,,") in new stack
    -- Executing [record@sub-record-check:3] Set("SIP/101-00000103", "__REC_STATUS=RECORDING") in new stack
  == Begin MixMonitor Recording SIP/101-00000103
    -- Executing [record@sub-record-check:4] Set("SIP/101-00000103", "CDR(recordingfile)=out-0149xxxxxx-101-20150320-164340-1426869820.301.wav") in new stack
    -- Executing [record@sub-record-check:5] Return("SIP/101-00000103", "") in new stack
    -- Executing [out@sub-record-check:3] Return("SIP/101-00000103", "") in new stack
    -- Executing [0149xxxxxx@from-internal:5] Macro("SIP/101-00000103", "dialout-trunk,5,0033149xxxxxx,,off") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/101-00000103", "DIAL_TRUNK=5") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/101-00000103", "0?sub-pincheck,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/101-00000103", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/101-00000103", "DIAL_NUMBER=0033149xxxxxx") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/101-00000103", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/101-00000103", "OUTBOUND_GROUP=OUT_5") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/101-00000103", "0?nomax") in new stack
    -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/101-00000103", "0?chanfull") in new stack
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/101-00000103", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/101-00000103", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/101-00000103", "outbound-callerid,5") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/101-00000103", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/101-00000103", "0?Set(REALCALLERIDNUM=101)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/101-00000103", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/101-00000103", "USEROUTCID=0176xxxxxx") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/101-00000103", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/101-00000103", "TRUNKOUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/101-00000103", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,14)
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/101-00000103", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/101-00000103", "1?Set(CALLERID(all)=0176xxxxxx)") in new stack
    -- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/101-00000103", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/101-00000103", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-outbound-callerid:18] Set("SIP/101-00000103", "CDR(outbound_cnum)=0176215694") in new stack
    -- Executing [s@macro-outbound-callerid:19] Set("SIP/101-00000103", "CDR(outbound_cnam)=") in new stack
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/101-00000103", "1?sub-flp-5,s,1()") in new stack
    -- Executing [s@sub-flp-5:1] ExecIf("SIP/101-00000103", "0?Set(TARGET_FLP_5=0033149xxxxxx)") in new stack
    -- Executing [s@sub-flp-5:2] GotoIf("SIP/101-00000103", "0?match") in new stack
    -- Executing [s@sub-flp-5:3] Return("SIP/101-00000103", "") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/101-00000103", "OUTNUM=0033149xxxxxx") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/101-00000103", "custom=SIP/FD") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/101-00000103", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
    -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/101-00000103", "0?Set(DIAL_TRUNK_OPTIONS=M(confirm))") in new stack
    -- Executing [s@macro-dialout-trunk:17] Macro("SIP/101-00000103", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/101-00000103", "") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/101-00000103", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/101-00000103", "1?Set(CONNECTEDLINE(num,i)=0033149xxxxxx)") in new stack
    -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/101-00000103", "1?Set(CONNECTEDLINE(name,i)=CID:0176xxxxxx)") in new stack
    -- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/101-00000103", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:22] Dial("SIP/101-00000103", "SIP/FD/0033149xxxxxx,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/FD/0033149xxxxxx
    -- SIP/FD-00000104 is making progress passing it to SIP/101-00000103
       > 0x1d47d800 -- Probation passed - setting RTP source address to 192.168.1.138:54690
       > 0x1d4faf90 -- Probation passed - setting RTP source address to 217.195.xx.xxx:36928
    -- 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/101-00000103", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34") in new stack
    -- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/101-00000103", "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/101-00000103", "RC=34") in new stack
    -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/101-00000103", "34,1") in new stack
    -- Goto (macro-dialout-trunk,34,1)
    -- Executing [34@macro-dialout-trunk:1] Goto("SIP/101-00000103", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/101-00000103", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:2] Set("SIP/101-00000103", "CALLERID(number)=101") in new stack
    -- Executing [0149xxxxxx@from-internal:6] Macro("SIP/101-00000103", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("SIP/101-00000103", "") in new stack
    -- Executing [s@macro-outisbusy:2] GotoIf("SIP/101-00000103", "0?emergency,1") in new stack
    -- Executing [s@macro-outisbusy:3] GotoIf("SIP/101-00000103", "0?intracompany,1") in new stack
    -- Executing [s@macro-outisbusy:4] Playback("SIP/101-00000103", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
[2015-03-20 16:43:40] WARNING[2826][C-00000089]: file.c:701 ast_openstream_full: File all-circuits-busy-now does not exist in any format
[2015-03-20 16:43:40] WARNING[2826][C-00000089]: file.c:1017 ast_streamfile: Unable to open all-circuits-busy-now (format (ulaw)): No such file or directory
[2015-03-20 16:43:40] WARNING[2826][C-00000089]: app_playback.c:484 playback_exec: ast_streamfile failed on SIP/101-00000103 for all-circuits-busy-now&pls-try-call-later, noanswer
[2015-03-20 16:43:40] WARNING[2826][C-00000089]: file.c:701 ast_openstream_full: File pls-try-call-later does not exist in any format
[2015-03-20 16:43:40] WARNING[2826][C-00000089]: file.c:1017 ast_streamfile: Unable to open pls-try-call-later (format (ulaw)): No such file or directory
[2015-03-20 16:43:40] WARNING[2826][C-00000089]: app_playback.c:484 playback_exec: ast_streamfile failed on SIP/101-00000103 for all-circuits-busy-now&pls-try-call-later, noanswer
    -- Executing [s@macro-outisbusy:5] Congestion("SIP/101-00000103", "20") in new stack
[2015-03-20 16:43:40] WARNING[2826][C-00000089]: channel.c:4862 ast_prod: Prodding channel 'SIP/101-00000103' failed
  == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/101-00000103' in macro 'outisbusy'
  == Spawn extension (from-internal, 0149xxxxxx, 6) exited non-zero on 'SIP/101-00000103'
    -- Executing [h@from-internal:1] Hangup("SIP/101-00000103", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-00000103'
  == MixMonitor close filestream (mixed)
  == End MixMonitor Recording SIP/101-00000103





in var/log/asterisk 




[2015-03-20 16:52:24] WARNING[2897][C-0000008c] file.c: File all-circuits-busy-now does not exist in any format
[2015-03-20 16:52:24] WARNING[2897][C-0000008c] file.c: Unable to open all-circuits-busy-now (format (ulaw)): No such file or directory
[2015-03-20 16:52:24] WARNING[2897][C-0000008c] app_playback.c: ast_streamfile failed on SIP/101-00000109 for all-circuits-busy-now&pls-try-call-later, noanswer
[2015-03-20 16:52:24] WARNING[2897][C-0000008c] file.c: File pls-try-call-later does not exist in any format
[2015-03-20 16:52:24] WARNING[2897][C-0000008c] file.c: Unable to open pls-try-call-later (format (ulaw)): No such file or directory
[2015-03-20 16:52:24] WARNING[2897][C-0000008c] app_playback.c: ast_streamfile failed on SIP/101-00000109 for all-circuits-busy-now&pls-try-call-later, noanswer
[2015-03-20 16:52:24] WARNING[2897][C-0000008c] channel.c: Prodding channel 'SIP/101-00000109' failed



any help please


thanks and regards.

[kctrey at gmail.com]

I am making some assumptions, but assuming the 217.195.xx.xxx is your provider, you are getting this back from them:

"Got SIP response 556 "No address found" back from 217.195.xx.xxx:5060"



Are you sure that "0033149xxxxxx" is the format the provider is expecting? You might try enabling SIP debug on the 217.195.xx.xx IP and seeing what the INVITE looks like, but normally a 556 indicates that your provider didn't have routing for either the R-URI or they didn't recognize that is was coming from you. You might compare the SIP INVITE coming from Asterisk to the one from Z-Lite and see where the differences are.






On Fri, Mar 20, 2015 at 12:03 PM Salaheddine Elharit <salah.elharit200@gmail.com (salah.elharit200@gmail.com)> wrote:

[salah.elharit200 at gm...]

i noticed that when i active the voicemail in the IP-phone where the number 0033149xxxxxx is configured i can call this number without issue


Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/FD/0033149xxxxxx == Begin MixMonitor Recording SIP/101-0000010d
    -- SIP/FD-0000010e is making progress passing it to SIP/101-0000010d
       > 0x2b393cfc2610 -- Probation passed - setting RTP source address to 192.                                                                                        168.1.138:55542
       > 0x1d08efa0 -- Probation passed - setting RTP source address to  217.195.xx.xx:46346
    -- SIP/FD-0000010e answered SIP/101-0000010d
       > 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346

thanks and regards.

[kctrey at gmail.com]

So you are saying that it resolved the issue to activate voicemail on the device that sits past your trunk provider? That confuses me a little, but if your calls are working, that's great news.


On Fri, Mar 20, 2015 at 1:44 PM Salaheddine Elharit <salah.elharit200@gmail.com (salah.elharit200@gmail.com)> wrote:

[salah.elharit200 at gm...]

thanks for your response

i noticed that when i active the voicemail in the IP-phone where the number 0033149xxxxxx is configured i can call this number without issue


the server asterisk and the ip-phone where the number is configured are in the same network 192.168.1.X


Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/FD/0033149xxxxxx
  == Begin MixMonitor Recording SIP/101-0000010d
    -- SIP/FD-0000010e is making progress passing it to SIP/101-0000010d
       > 0x2b393cfc2610 -- Probation passed - setting RTP source address to 192.                                                                                        168.1.138:55542
       > 0x1d08efa0 -- Probation passed - setting RTP source address to  217.195.xx.xx:46346
    -- SIP/FD-0000010e answered SIP/101-0000010d
       > 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346

thanks and regards.


2015-03-20 18:39 GMT+00:00 Salaheddine Elharit <salah.elharit200@gmail.com (salah.elharit200@gmail.com)>:

[salah.elharit200 at gm...]

hi 
 
the issue still the same i have 2 trunks whe i configure the first in x-lite and the second in my server or my ip-phone snom320 directly 
 
from x-lite i can call my trunk without issue but when i try ti call from snom320 to x-lite or from my server asterisk using extension in x-lite the call all time is failed 
 
any help please
 
thanks and regards

2015-03-20 19:28 GMT+00:00 Trey Hilyard <kctrey@gmail.com (kctrey@gmail.com)>: