Outbound Calls

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Asterisk Users 5 Comments

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_MONITOR26869820.301”) in new stack
— Executing [s@macro-user-callerid:2] Set(“SIP/101-00000103”,
“AMPUSER1”) 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(REALCALLERIDNUM1)”) in new stack
— Executing [s@macro-user-callerid:5] Set(“SIP/101-00000103”,
“AMPUSER1”) 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”,
“AMPUSERCIDNAME1”) 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”,
“AMPUSERCID1”) 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)1)”) 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)1”) in new stack
— Executing [s@macro-user-callerid:29] Set(“SIP/101-00000103”,
“CALLERID(name)1”) in new stack
— Executing [s@macro-user-callerid:30] Set(“SIP/101-00000103”,
“CDR(cnum)1”) in new stack
— Executing [s@macro-user-callerid:31] Set(“SIP/101-00000103”,
“CDR(cnam)1”) 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

5 thoughts on - Outbound Calls

  • 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.

  • 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.

  • 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.

  • 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 :

  • 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 :