SIP Trunk to OC

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Nov 23, 2010
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#1
hi mate
i create sip trunk for makeing call to OCS R2
Actually i follow this link
http://blogs.breezetraining.com.au/mick ... TNPBX.aspx
unfortunately i couldn't make successful call for example from xlite to OC
you know when i call from xlite to OC ,you can hear tone in xlite but nothing happen in OCS side
(OC user no +7001 , elastix user ext is 1212 )
here is debug :

<------------->
--- (12 headers 15 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 172.18.1.191 : 53657 (NAT)
Using INVITE request as basis request - e44eca4c626e6641MWEzNjViNTk1ZDViNDdlYWM2NjY0Yzg3YzM0NmQ1MDA.
Found peer '1212' for '1212' from 172.18.1.191:53657
Found RTP audio format 107
Found RTP audio format 119
Found RTP audio format 0
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format BV32 for ID 107
Found audio description format BV32-FEC for ID 119
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.18.1.191:36868
Looking for 7001 in from-internal (domain 172.18.1.60)
list_route: hop: <sip:1212@172.18.1.191:53657>

<--- Transmitting (NAT) to 172.18.1.191:53657 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.18.1.191:53657;branch=z9hG4bK-d87543-5266110f611fd62e-1--d87543-;received=172.18.1.191;rport=53657
From: "ali"<sip:1212@172.18.1.60>;tag=d67dfd0c
To: "7001"<sip:7001@172.18.1.60>
Call-ID: e44eca4c626e6641MWEzNjViNTk1ZDViNDdlYWM2NjY0Yzg3YzM0NmQ1MDA.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:7001@172.18.1.60>
Content-Length: 0


<------------>
-- Executing [7001@from-internal:1] Macro("SIP/1212-00000091", "exten-vm,novm,7001") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/1212-00000091", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/1212-00000091", "AMPUSER=1212") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/1212-00000091", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/1212-00000091", "1?Set(REALCALLERIDNUM=1212)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/1212-00000091", "AMPUSER=1212") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/1212-00000091", "AMPUSERCIDNAME=lisha") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/1212-00000091", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/1212-00000091", "AMPUSERCID=1212") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/1212-00000091", "CALLERID(all)="lisha" <1212>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/1212-00000091", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/1212-00000091", "0?continue") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/1212-00000091", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/1212-00000091", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("SIP/1212-00000091", "Using CallerID "lisha" <1212>") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/1212-00000091", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/1212-00000091", "VMBOX=novm") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/1212-00000091", "EXTTOCALL=7001") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/1212-00000091", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/1212-00000091", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/1212-00000091", "RT=""") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/1212-00000091", "record-enable,7001,IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/1212-00000091", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/1212-00000091", "0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/1212-00000091", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] GotoIf("SIP/1212-00000091", "1?IN") in new stack
-- Goto (macro-record-enable,s,20)
-- Executing [s@macro-record-enable:20] ExecIf("SIP/1212-00000091", "1?MacroExit()") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/1212-00000091", "dial,,tr,7001") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/1212-00000091", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/1212-00000091", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is 'lisha' number is '1212'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 7001 to extension map
-- dialparties.agi: Extension 7001 cf is disabled
-- dialparties.agi: Extension 7001 do not disturb is disabled
dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
dialparties.agi: Extension 7001 has ExtensionState: 0
-- dialparties.agi: Checking CW and CFB status for extension 7001
-- dialparties.agi: dbset CALLTRACE/7001 to 1212
-- dialparties.agi: Filtered ARG3: 7001
-- <SIP/1212-00000091>AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/1212-00000091", "SIP/+7001@from-msuc,,tr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 172.18.1.60 port 17320
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 172.18.1.67:5060:
INVITE sip:+7001@172.18.1.67:5060 SIP/2.0
Via: SIP/2.0/TCP 172.18.1.60:5060;branch=z9hG4bK2c0d29da;rport
Max-Forwards: 70
From: "lisha" <sip:1212@MD.dorb.com>;tag=as6b1ca8f0
To: <sip:+7001@172.18.1.67:5060>
Contact: <sip:1212@172.18.1.60;transport=TCP>
Call-ID: 6ba31ea9388fdbc36e2d21c7688a1789@MD.dorb.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.10
Date: Thu, 09 Dec 2010 13:10:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 176410072 176410072 IN IP4 172.18.1.60
s=Asterisk PBX 1.6.2.10
c=IN IP4 172.18.1.60
t=0 0
m=audio 17320 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
-- Called +7001@from-msuc

<--- Transmitting (NAT) to 172.18.1.191:53657 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.18.1.191:53657;branch=z9hG4bK-d87543-5266110f611fd62e-1--d87543-;received=172.18.1.191;rport=53657
From: "ali"<sip:1212@172.18.1.60>;tag=d67dfd0c
To: "7001"<sip:7001@172.18.1.60>;tag=as45906fd1
Call-ID: e44eca4c626e6641MWEzNjViNTk1ZDViNDdlYWM2NjY0Yzg3YzM0NmQ1MDA.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:7001@172.18.1.60>
Content-Length: 0


<------------>

<--- SIP read from TCP:172.18.1.67:5060 --->
SIP/2.0 100 Trying
FROM: "lisha"<sip:1212@MD.dorb.com>;tag=as6b1ca8f0
TO: <sip:+7001@172.18.1.67:5060>
CSEQ: 102 INVITE
CALL-ID: 6ba31ea9388fdbc36e2d21c7688a1789@MD.dorb.com
VIA: SIP/2.0/TCP 172.18.1.60:5060;branch=z9hG4bK2c0d29da;rport
CONTENT-LENGTH: 0


<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:172.18.1.191:53657 --->



<------------->
Reliably Transmitting (NAT) to 172.18.1.191:53657:
OPTIONS sip:1212@172.18.1.191:53657;rinstance=c8d29d8c6e6c2b5f SIP/2.0
Via: SIP/2.0/UDP 172.18.1.60:5060;branch=z9hG4bK337ea8dc;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@172.18.1.60>;tag=as615be746
To: <sip:1212@172.18.1.191:53657;rinstance=c8d29d8c6e6c2b5f>
Contact: <sip:Unknown@172.18.1.60>
Call-ID: 49ea37c10506e78520d0f7c667c2fb8b@172.18.1.60
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Thu, 09 Dec 2010 13:10:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:172.18.1.191:53657 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.1.60:5060;branch=z9hG4bK337ea8dc;rport=5060
Contact: <sip:172.18.1.191:53657>
To: <sip:1212@172.18.1.191:53657;rinstance=c8d29d8c6e6c2b5f>;tag=253a1558
From: "Unknown"<sip:Unknown@172.18.1.60>;tag=as615be746
Call-ID: 49ea37c10506e78520d0f7c667c2fb8b@172.18.1.60
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1002tx stamp 29712
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '49ea37c10506e78520d0f7c667c2fb8b@172.18.1.60' Method: OPTIONS
ivr*CLI>


i couldn't make call from OC to Elastix users also

would you plz help me to figure it out ? what should i do ?
 

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