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Problem sa Orion trunkom i odlazecim pozivima

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Problem sa Orion trunkom i odlazecim pozivima 27.07.2012. u 16:49 · pre 173 meseca #1
Pozdrav svima,

Mucim se da postavim Orionov Sip trunk vec neko vreme i postalo je nepodnosljivo :)

Problem je sledeci

Uspevam da dobijem pozive,ali ne mogu da ostvarim odlazeci poziv

Postavljam Vam logove,pa ako neko ima neku sugestiju, puno hvala

Code:
sip show peers


Code:
Name/username              Host                                    Dyn Forcerport ACL Port     Status
101/101                    192.168.3.200                            D   N   A  22960    OK (102 ms)
OrionTelekom/username      xxx.xxx.xxx.xxx                              N       4060    OK (11 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]


Code:
sip show registry


Code:
Host                                    dnsmgr   Username       Refresh State                Reg.Time
xxx.xxx.xxx.xxx:5060                      N      username        120 Request Sent
1 SIP registrations.



Kao sto se vidi, vidljiv je ali nije se registovao


Code:
Peer details


Code:
username=username
type=friend
secret=password
rtpstart=5004
rtpend=5014
qualify=yes
outboundproxy=xxx.xxx.xxx.xxx
insecure=invite,port
port=4060
host=xxx.xxx.xxx.xxx
fromdomain=xxx.xxx.xxx.xxx
dtmfmode=rfc2833
disallow=all
allow=alaw&ulaw
context=from-trunk
canreinvite=no
[email protected]
auth=md5
nat=yes



Register String

username:[email protected]


probao sam i sa
Code:
username:[email protected]/username


dalje


Code:
sip set debug on


Code:
Supported: replaces, timer
Contact: <sip:192.168.3.147:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: OPTIONS)
Retransmitting #5 (no NAT) to xxx.xxx.xxx:5060:
REGISTER sip:xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.3.147:5060;branch=z9hG4bK5b772af7
Max-Forwards: 70
From: <sip:[email protected]>;tag=as5a27e2b6
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 108 REGISTER
User-Agent: FPBX-2.8.1(1.8.7.0)
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0


---
Retransmitting #6 (no NAT) to xxx.xxx.xxx:5060:
REGISTER sip:xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.3.147:5060;branch=z9hG4bK5b772af7
Max-Forwards: 70
From: <sip:[email protected]>;tag=as5a27e2b6
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 108 REGISTER
User-Agent: FPBX-2.8.1(1.8.7.0)
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0


---
Retransmitting #7 (no NAT) to xxx.xxx.xxx:5060:
REGISTER sip:xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.3.147:5060;branch=z9hG4bK5b772af7
Max-Forwards: 70
From: <sip:[email protected]>;tag=as5a27e2b6
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 108 REGISTER
User-Agent: FPBX-2.8.1(1.8.7.0)
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0


---
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to xxx.xxx.xxx:5060:
REGISTER sip:xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.3.147:5060;branch=z9hG4bK47d92ace
Max-Forwards: 70
From: <sip:[email protected]>;tag=as7c6ec84a
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 109 REGISTER
User-Agent: FPBX-2.8.1(1.8.7.0)
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0


---
Really destroying SIP dialog '[email protected]' Method: REGISTER
Retransmitting #1 (no NAT) to xxx.xxx.xxx:5060:
REGISTER sip:xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.3.147:5060;branch=z9hG4bK47d92ace
Max-Forwards: 70
From: <sip:[email protected]>;tag=as7c6ec84a
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 109 REGISTER
User-Agent: FPBX-2.8.1(1.8.7.0)
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0


---
Retransmitting #2 (no NAT) to xxx.xxx.xxx:5060:
REGISTER sip:xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.3.147:5060;branch=z9hG4bK47d92ace
Max-Forwards: 70
From: <sip:[email protected]>;tag=as7c6ec84a
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 109 REGISTER
User-Agent: FPBX-2.8.1(1.8.7.0)
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0


---
Retransmitting #3 (no NAT) to xxx.xxx.xxx:5060:
REGISTER sip:xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.3.147:5060;branch=z9hG4bK47d92ace
Max-Forwards: 70
From: <sip:[email protected]>;tag=as7c6ec84a
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 109 REGISTER
User-Agent: FPBX-2.8.1(1.8.7.0)
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0


---
Retransmitting #4 (no NAT) to xxx.xxx.xxx:5060:
REGISTER sip:xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.3.147:5060;branch=z9hG4bK47d92ace
Max-Forwards: 70
From: <sip:[email protected]>;tag=as7c6ec84a
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 109 REGISTER
User-Agent: FPBX-2.8.1(1.8.7.0)
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0


A kada se poziva sa lokala dobija se sledece

Code:
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [9715****@from-internal:1] Macro("SIP/101-00000008", "user-callerid,SKIPTTL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/101-00000008", "AMPUSER=101") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/101-00000008", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/101-00000008", "1?Set(REALCALLERIDNUM=101)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/101-00000008", "AMPUSER=101") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/101-00000008", "AMPUSERCIDNAME=101") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/101-00000008", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/101-00000008", "AMPUSERCID=101") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/101-00000008", "CALLERID(all)="101" <101>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/101-00000008", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/101-00000008", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] Set("SIP/101-00000008", "CALLERID(number)=101") in new stack
    -- Executing [s@macro-user-callerid:20] Set("SIP/101-00000008", "CALLERID(name)=101") in new stack
    -- Executing [s@macro-user-callerid:21] NoOp("SIP/101-00000008", "Using CallerID "101" <101>") in new stack
    -- Executing [9715****@from-internal:2] NoOp("SIP/101-00000008", "Calling Out Route: 9_outside") in new stack
    -- Executing [9715****@from-internal:3] Set("SIP/101-00000008", "MOHCLASS=default") in new stack
    -- Executing [9715****@from-internal:4] Set("SIP/101-00000008", "_NODEST=") in new stack
    -- Executing [9715****@from-internal:5] Macro("SIP/101-00000008", "record-enable,101,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/101-00000008", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/101-00000008", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/101-00000008", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,15)
    -- Executing [s@macro-record-enable:15] GotoIf("SIP/101-00000008", "0?IN") in new stack
    -- Executing [s@macro-record-enable:16] ExecIf("SIP/101-00000008", "1?MacroExit()") in new stack
    -- Executing [9715****@from-internal:6] Macro("SIP/101-00000008", "dialout-trunk,2,715****,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/101-00000008", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/101-00000008", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/101-00000008", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/101-00000008", "DIAL_NUMBER=715****") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/101-00000008", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/101-00000008", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/101-00000008", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/101-00000008", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/101-00000008", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/101-00000008", "outbound-callerid,2") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/101-00000008", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/101-00000008", "0?Set(REALCALLERIDNUM=101)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/101-00000008", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/101-00000008", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/101-00000008", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/101-00000008", "TRUNKOUTCID=3811141*****") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/101-00000008", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/101-00000008", "1?Set(CALLERID(all)=3811141*****)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/101-00000008", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/101-00000008", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/101-00000008", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/101-00000008", "1?sub-flp-2,s,1") in new stack
    -- Executing [s@sub-flp-2:1] ExecIf("SIP/101-00000008", "0?Set(TARGET_FLP_2=15****)") in new stack
    -- Executing [s@sub-flp-2:2] GotoIf("SIP/101-00000008", "0?match") in new stack
    -- Executing [s@sub-flp-2:3] Return("SIP/101-00000008", "") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/101-00000008", "OUTNUM=715****") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/101-00000008", "custom=SIP/OrionTelekom") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/101-00000008", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/101-00000008", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/101-00000008", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/101-00000008", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/101-00000008", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/101-00000008", "SIP/OrionTelekom/715****,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/OrionTelekom/715****
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/101-00000008", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 111") in new stack
    -- Executing [s@macro-dialout-trunk:21] Goto("SIP/101-00000008", "s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/101-00000008", "RC=111") in new stack
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/101-00000008", "111,1") in new stack
    -- Goto (macro-dialout-trunk,111,1)
    -- Executing [111@macro-dialout-trunk:1] Goto("SIP/101-00000008", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/101-00000008", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,continue,3)
    -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/101-00000008", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 111 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:4] Set("SIP/101-00000008", "CALLERID(number)=101") in new stack
    -- Executing [9715****@from-internal:7] Macro("SIP/101-00000008", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("SIP/101-00000008", "") in new stack
    -- Executing [s@macro-outisbusy:2] GotoIf("SIP/101-00000008", "0?emergency,1") in new stack
    -- Executing [s@macro-outisbusy:3] GotoIf("SIP/101-00000008", "0?intracompany,1") in new stack
    -- Executing [s@macro-outisbusy:4] Playback("SIP/101-00000008", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
    -- <SIP/101-00000008> Playing 'all-circuits-busy-now.gsm' (language 'en')
    -- <SIP/101-00000008> Playing 'pls-try-call-later.gsm' (language 'en')
  == Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/101-00000008' in macro 'outisbusy'
  == Spawn extension (from-internal, 9715****, 7) exited non-zero on 'SIP/101-00000008'
    -- Executing [h@from-internal:1] Macro("SIP/101-00000008", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/101-00000008", "1?endmixmoncheck") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] NoOp("SIP/101-00000008", "End of MIXMON check") in new stack
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/101-00000008", "1?nomeetmemon") in new stack
    -- Goto (macro-hangupcall,s,15)
    -- Executing [s@macro-hangupcall:15] NoOp("SIP/101-00000008", "MEETME_RECORDINGFILE=") in new stack
    -- Executing [s@macro-hangupcall:16] GotoIf("SIP/101-00000008", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,18)
    -- Executing [s@macro-hangupcall:18] NoOp("SIP/101-00000008", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:19] GotoIf("SIP/101-00000008", "1?noautomon2") in new stack
    -- Goto (macro-hangupcall,s,25)
    -- Executing [s@macro-hangupcall:25] NoOp("SIP/101-00000008", "MONITOR_FILENAME=") in new stack
    -- Executing [s@macro-hangupcall:26] GotoIf("SIP/101-00000008", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,29)
    -- Executing [s@macro-hangupcall:29] GotoIf("SIP/101-00000008", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,32)
    -- Executing [s@macro-hangupcall:32] GotoIf("SIP/101-00000008", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,34)
    -- Executing [s@macro-hangupcall:34] Hangup("SIP/101-00000008", "") in new stack
  == Spawn extension (macro-hangupcall, s, 34) exited non-zero on 'SIP/101-00000008' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-00000008'



Kao sto rekoh, svaka pomoc je dobrodosla

Inace, port 4060 sam od njih dobio da prosledim.Nije greska
Re: Problem sa Orion trunkom i odlazecim pozivima 27.07.2012. u 19:29 · pre 173 meseca #2
A jesi probao sa registracijom na port 4060, posto se ocigledno to sad koristi (nekad je bio 5064) ?

Code:

username:[email protected]:4060/username
Potpis
"99% of your thought process is protecting your self-conceptions,
and 98% of that is wrong."
Re: Problem sa Orion trunkom i odlazecim pozivima 27.07.2012. u 19:39 · pre 173 meseca #3
Nisam probao

Sada jesam i dobijam


Code:
localhost*CLI> sip show registry
Host                                    dnsmgr      Username       Refresh State                Reg.Time
xxx.xxx.xxx.xxx:4060                      N      username        120     No Authentication
1 SIP registrations.



probao sam i bez /username sintakse

Hvala na odgovoru svakako
Re: Problem sa Orion trunkom i odlazecim pozivima 27.07.2012. u 23:01 · pre 173 meseca #4
Ako je to 1.8 verzija asteriska, onda register string treba da izgleda:

authid:password:authid@domen

Umesto domena moze da bude i IP adresa naravno.
Potpis
"99% of your thought process is protecting your self-conceptions,
and 98% of that is wrong."
Re: Problem sa Orion trunkom i odlazecim pozivima 27.07.2012. u 23:26 · pre 173 meseca #6
Ok, to je distro verzija, ali koja je verzija samog asterisk-a na njemu?

asterisk -V

Ako je 1.8 onda se koristi gore pomenuti string, ako je 1.4 i sl, onda onaj stari username:password@domen:port/username

Dalje, ne razumem te najbolje koji deo poziva ti ne radi. Na osnovu onog log-a, reklo bi se da ti odlazni pozivi ne rade, sad kazes da prolaze od tebe ka Orion-u (a to je odlazni smer), u prvoj poruci si rekao suprotno...?

Trece, jesi siguran za one RTP portove gore koji su ti u definiciji trunk-a? Telekom za BBTF koristi te, ali koliko znam Orion nije nikada forsirao neke nestandardne rtp portove... Iskomentarisi te dve linije pa probaj opet.


Potpis
"99% of your thought process is protecting your self-conceptions,
and 98% of that is wrong."
Re: Problem sa Orion trunkom i odlazecim pozivima 27.07.2012. u 23:44 · pre 173 meseca #7
1.8 asterisk

Obrnuo sam sorry
Odlazni ne rade.. Dolaz radi...

Nisam siguran.. Stavio sam ih na kraju jer nisam vise znao sta da probam... Sto se kaze iz ocaja :)

Sve sto sam dobio od njih je na usmenom nivou.. javni ip, user/pass i port 4060 :D

probao sam i sa rtp i bez.. Uvek je isti rezultat. Nazalost, ne mogu zvati podrsku do ponedeljka jer je podrska malobrojna...

Svakako ti hvala na sugestijama

Probacu na x-lite da nakacim trunk pa da vidim sta ce biti.
Re: Problem sa Orion trunkom i odlazecim pozivima 28.07.2012. u 00:09 · pre 173 meseca #8
Ajde sada okaci sta dobijas kada ukljucis debug za taj IP, posto si promenio da ide na port 4060..

I koji tacno je register string sada za 1.8?
Potpis
"99% of your thought process is protecting your self-conceptions,
and 98% of that is wrong."
Re: Problem sa Orion trunkom i odlazecim pozivima 28.07.2012. u 00:32 · pre 173 meseca #9
Code:
<--- SIP read from UDP:xxx.xxx.xxx.xxx:4060 --->
OPTIONS sip:192.168.3.147:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:4060;branch=z9hG4bK1936335d;rport
From: "asterisk" <sip:[email protected]:4060>;tag=as4abc4d80
To: <sip:192.168.3.147:5060>
Contact: <sip:[email protected]:4060>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: FaxServer PBX
Max-Forwards: 70
Date: Fri, 27 Jul 2012 22:33:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Looking for s in from-sip-external (domain 192.168.3.147:5060)

<--- Transmitting (no NAT) to xxx.xxx.xxx.xxx:4060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:4060;branch=z9hG4bK1936335d;rport;received=xxx.xxx.xxx.xxx
From: "asterisk" <sip:[email protected]:4060>;tag=as4abc4d80
To: <sip:192.168.3.147:5060>;tag=as7d1c3da2
Call-ID: [email protected]
CSeq: 102 OPTIONS
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:192.168.3.147:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '[email protected]' Method: OPTIONS
Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:4060:
OPTIONS sip:xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.3.147:5060;branch=z9hG4bK076c1dc1;rport
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as6cea550c
To: <sip:xxx.xxx.xxx.xxx>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.7.0)
Date: Fri, 27 Jul 2012 22:36:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:xxx.xxx.xxx.xxx:4060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.3.147:5060;branch=z9hG4bK076c1dc1;received=192.168.3.147;rport=5060
From: "Unknown" <sip:[email protected]:5060>;tag=as6cea550c
To: <sip:xxx.xxx.xxx.xxx>;tag=as003e7f9a
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FaxServer PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:xxx.xxx.xxx.xxx:4060 --->
OPTIONS sip:192.168.3.147:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:4060;branch=z9hG4bK19090df6;rport
From: "asterisk" <sip:[email protected]:4060>;tag=as321baf0a
To: <sip:192.168.3.147:5060>
Contact: <sip:[email protected]:4060>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: FaxServer PBX
Max-Forwards: 70
Date: Fri, 27 Jul 2012 22:34:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Looking for s in from-sip-external (domain 192.168.3.147:5060)

<--- Transmitting (no NAT) to xxx.xxx.xxx.xxx:4060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:4060;branch=z9hG4bK19090df6;rport;received=xxx.xxx.xxx.xxx
From: "asterisk" <sip:[email protected]:4060>;tag=as321baf0a
To: <sip:192.168.3.147:5060>;tag=as118e4fd5
Call-ID: [email protected]
CSeq: 102 OPTIONS
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:192.168.3.147:5060>
Accept: application/sdp
Content-Length: 0



FaxServer?? pa sta su pustili po ovome?


string

username:pass:username@domain:4060
Re: Problem sa Orion trunkom i odlazecim pozivima 28.07.2012. u 00:44 · pre 173 meseca
izmenjeno 3 puta, poslednji put klikton 28.07.2012. u 01:04
Izmene poruke
  1. klikton28.07.2012. u 01:04zapis iz teksta poruke
  2. klikton28.07.2012. u 00:58zapis iz teksta poruke
  3. klikton28.07.2012. u 00:54zapis iz teksta poruke
#10
He.. radi... ne znam

sad cu sve confige moguce da backupujem i da veceras prelupam celu centralu

Ne moze tek tako da proradi.... ne razumem sad nista

Promenio sam reg string, port..

Mogu i da primam pozive i da zovem

a na sip registry pise no authen...
Re: Problem sa Orion trunkom i odlazecim pozivima 26.03.2013. u 06:20 · pre 165 meseci #11
Pozdrav,

Jesi li uspeo da rešiš? Koja je usluga u pitanju tačno? Jel to za fizička ili pravna lica? Jesi li probao CallerID da setujes, jer te možda puštaju napolje na osnovu njega.
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