[ klikton @ 27.07.2012. 15:49 ] @
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
[ optix @ 27.07.2012. 18:29 ] @
A jesi probao sa registracijom na port 4060, posto se ocigledno to sad koristi (nekad je bio 5064) ?

Code:

username:[email protected]:4060/username
[ klikton @ 27.07.2012. 18:39 ] @
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
[ optix @ 27.07.2012. 22:01 ] @
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.
[ klikton @ 27.07.2012. 22:14 ] @
Elastix 1.8.7.0

Pokusao sam ranije taj string... I varijacije sa /username

Rezlutat je isti.

Najgore je sto pozivi prodju od mene ka Orionu. ali dalje nece
[ optix @ 27.07.2012. 22:26 ] @
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.


[ klikton @ 27.07.2012. 22:44 ] @
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

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.
[ optix @ 27.07.2012. 23:09 ] @
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?
[ klikton @ 27.07.2012. 23:32 ] @
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
[ klikton @ 27.07.2012. 23:44 ] @
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...





[Ovu poruku je menjao klikton dana 28.07.2012. u 00:54 GMT+1]

[Ovu poruku je menjao klikton dana 28.07.2012. u 00:58 GMT+1]

[Ovu poruku je menjao klikton dana 28.07.2012. u 01:04 GMT+1]
[ Dennis @ 26.03.2013. 05:20 ] @
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.