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.elharit...@gmail.com>:
> thank you
>
> 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.
>
> 2015-03-20 17:15 GMT+00:00 Trey Hilyard <kct...@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.elharit...@gmail.com> wrote:
>>
>>> 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,28)
>>> -- 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.
>>>
>>>
>>> --
>>> _____________________________________________________________________
>>> -- 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
>>
>
>
--
_____________________________________________________________________
-- 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
