[es] - Problemi sa konfiguracijom SIP Trunka prema VoIP provajderu

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Re: Problemi sa konfiguracijom SIP Trunka prema VoIP provajderu

^{25.05.2011. u 20:07 - pre 119 meseci}

Danas mi nikako ne ide 'vragu....uspio sam se djelimicno nakaciti na Callwithus ali nekim slucajem ne mogu uspostaviti poziv a i ringing dial tone je uzasno los, a nakon zvonjenja javi se digitalna sekretarica i jos mi obracuna poziv.....trenutno sam odustao od njega.

Drugi problem mi je sad trunkovanje prema localphone.com provajderu. Trebaju mi samo odlazni pozivi
@Ivane pomagaj....

pod SIP trunkom i outgoing settings sam definisao

Code:

username=xxxxx
type=peer
secret=xxxxx
authuser=xxxxx
host=localphone.com
fromdomain=localphone.com
dtmfmode=rfc2833
disallow=all
canreinvite=no
canredirect=no
insecure=very
nat=no
allow=alaw&ulaw&gsm

incoming pozivi mi nisu ni potrebni. Dial rules sam postavio

Code:
0|.

Ovo mi je CLI output:

Code:

localhost*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
localphone.com:5060 N 1684538 105 Registered Wed, 25 May 2011 21:02:04
1 SIP registrations.

No medjutim poziv ne mogu uspostaviti.....evo CLI outputa kada pokusam da preko xlite uspostavim poziv

Code:

== Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [[email protected]:1] Macro("SIP/201-0000071a", "user-callerid,SKIPTTL,") in new stack
    -- Executing [[email protected]:1] Set("SIP/201-0000071a", "AMPUSER=201") in new stack
    -- Executing [[email protected]:2] GotoIf("SIP/201-0000071a", "0?report") in new stack
    -- Executing [[email protected]:3] ExecIf("SIP/201-0000071a", "1?Set(REALCALLERIDNUM=201)") in new stack
    -- Executing [[email protected]:4] Set("SIP/201-0000071a", "AMPUSER=201") in new stack
    -- Executing [[email protected]:5] Set("SIP/201-0000071a", "AMPUSERCIDNAME=201") in new stack
    -- Executing [[email protected]:6] GotoIf("SIP/201-0000071a", "0?report") in new stack
    -- Executing [[email protected]:7] Set("SIP/201-0000071a", "AMPUSERCID=201") in new stack
    -- Executing [[email protected]:8] Set("SIP/201-0000071a", "CALLERID(all)="201" <201>") in new stack
    -- Executing [[email protected]:9] ExecIf("SIP/201-0000071a", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [[email protected]o-user-callerid:10] GotoIf("SIP/201-0000071a", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [[email protected]:19] NoOp("SIP/201-0000071a", "Using CallerID "201" <201>") in new stack
    -- Executing [[email protected]:2] Set("SIP/201-0000071a", "_NODEST=") in new stack
    -- Executing [[email protected]:3] Macro("SIP/201-0000071a", "record-enable,201,OUT,") in new stack
    -- Executing [[email protected]:1] GotoIf("SIP/201-0000071a", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [[email protected]:4] ExecIf("SIP/201-0000071a", "0?MacroExit()") in new stack
    -- Executing [[email protected]:5] GotoIf("SIP/201-0000071a", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,15)
    -- Executing [[email protected]:15] GotoIf("SIP/201-0000071a", "0?IN") in new stack
    -- Executing [[email protected]:16] ExecIf("SIP/201-0000071a", "1?MacroExit()") in new stack
    -- Executing [[email protected]:4] Macro("SIP/201-0000071a", "dialout-trunk,2,038762723254,,") in new stack
    -- Executing [[email protected]:1] Set("SIP/201-0000071a", "DIAL_TRUNK=2") in new stack
    -- Executing [[email protected]:2] GosubIf("SIP/201-0000071a", "0?sub-pincheck,s,1") in new stack
    -- Executing [[email protected]:3] GotoIf("SIP/201-0000071a", "1?disabletrunk,1") in new stack
    -- Goto (macro-dialout-trunk,disabletrunk,1)
    -- Executing [[email protected]:1] NoOp("SIP/201-0000071a", "TRUNK: SIP/JustVoip DISABLED - falling through to next trunk") in new stack
    -- Executing [[email protected]:5] Macro("SIP/201-0000071a", "outisbusy,") in new stack
    -- Executing [[email protected]:1] Progress("SIP/201-0000071a", "") in new stack
    -- Executing [[email protected]:2] GotoIf("SIP/201-0000071a", "0?emergency,1") in new stack
    -- Executing [[email protected]:3] GotoIf("SIP/201-0000071a", "0?intracompany,1") in new stack
    -- Executing [[email protected]:4] Playback("SIP/201-0000071a", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
    -- <SIP/201-0000071a> Playing 'all-circuits-busy-now.gsm' (language 'en')
    -- <SIP/201-0000071a> Playing 'pls-try-call-later.gsm' (language 'en')
    -- Executing [[email protected]:5] Congestion("SIP/201-0000071a", "20") in new stack
  == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/201-0000071a' in macro 'outisbusy'
  == Spawn extension (from-internal, 0038762723254, 5) exited non-zero on 'SIP/201-0000071a'
    -- Executing [[email protected]:1] Macro("SIP/201-0000071a", "hangupcall") in new stack
    -- Executing [[email protected]:1] GotoIf("SIP/201-0000071a", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [[email protected]:3] NoOp("SIP/201-0000071a", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [[email protected]:4] GotoIf("SIP/201-0000071a", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [[email protected]:7] GotoIf("SIP/201-0000071a", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,10)
    -- Executing [[email protected]:10] GotoIf("SIP/201-0000071a", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,12)
    -- Executing [[email protected]:12] Hangup("SIP/201-0000071a", "") in new stack
  == Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/201-0000071a' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/201-0000071a'

Odgovor na temu

Ivan Vujisic
Ivan Vujisic
Bgd

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Re: Problemi sa konfiguracijom SIP Trunka prema VoIP provajderu

^{25.05.2011. u 20:47 - pre 119 meseci}

Prvo:

sip show registry ti pokazuje ekstenzije registrovane kod provajdera za dolazne pozive. Ubi register => u sip.conf ili u sta ja znam sip_additional_mother_fucker.conf ili iz GUIja

Drugo:

CLI output je ka-ta-str-ofa! Eto razloga zasto nikad necu pristati da koristim Trixbox, Elastix etc. Neces verovati koliko CLI output jednostavnije izgleda ako sam definises dialplan. Veruj mi da to ne vodi nicemu. Moja skromna procena citanjem CLI loga je da nisi dobro definisao trunk. Evo obrati paznju na ovo

-- Executing [[email protected]:1] NoOp("SIP/201-0000071a", "TRUNK: SIP/JustVoip DISABLED - falling through to next trunk") in new stack
-- Executing [[email protected]:5] Macro("SIP/201-0000071a", "outisbusy,") in new stack
-- Executing [[email protected]:1] Progress("SIP/201-0000071a", "") in new stack
-- Executing [[email protected]:2] GotoIf("SIP/201-0000071a", "0?emergency,1") in new stack
-- Executing [[email protected]:3] GotoIf("SIP/201-0000071a", "0?intracompany,1") in new stack
-- Executing [[email protected]:4] Playback("SIP/201-0000071a", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack

OK, disable-ovao si JustVoIP trunk, ali ti posle toga ne ide na drugi trunk.

Odgovor na temu