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SIP Trunk Not Working

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jcallen03

jcallen03

Programmer
Sep 16, 2011
120
US
I am trying to set up a Nexvortex SIP trunk.

Here are the messages I am seeing with the monitor.

695464mS SIP Tx: UDP 192.168.42.2:5060 -> 66.23.129.253:5060
INVITE sip:mad:192.168.42.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.42.2:5060;rport;branch=z9hG4bK66febafa5b39e8e2f4bad834d3167158
From: "Membership" <sip:4143769110@nexvortex.com>;tag=ac90fe895a3fe640
To: <sip:mad:192.168.42.1>
Call-ID: ad01a39c824cab8e851e4dd45a4d110a@192.168.42.2
CSeq: 151821899 INVITE
Contact: "Membership" <sip:4142713656@192.168.42.2:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
Content-Length: 218

v=0
o=UserA 3991637103 830104317 IN IP4 192.168.42.2
s=Session SDP
c=IN IP4 192.168.42.2
t=0 0
m=audio 49154 RTP/AVP 0 8 18
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
695535mS SIP Rx: UDP 66.23.129.253:5060 -> 192.168.42.2:5060
SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL)
Via: SIP/2.0/UDP 192.168.42.2:5060;rport=5060;branch=z9hG4bK66febafa5b39e8e2f4bad834d3167158;received=67.53.0.87
From: "Membership" <sip:4143769110@nexvortex.com>;tag=ac90fe895a3fe640
To: <sip:mad:192.168.42.1>;tag=9c6a9fdfd4d16ebaa52f34c4c528cbe5.3129
Call-ID: ad01a39c824cab8e851e4dd45a4d110a@192.168.42.2
CSeq: 151821899 INVITE
Server: nVSIP 12.02.01
Content-Length: 0

696044mS SIP Tx: UDP 192.168.42.2:5060 -> 66.23.129.253:5060
INVITE sip:mad:192.168.42.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.42.2:5060;rport;branch=z9hG4bKc6029f4a14b2f309b099a97a91c2569b
From: "Membership" <sip:4143769110@nexvortex.com>;tag=b202340936270599
To: <sip:mad:192.168.42.1>
Call-ID: 3cc032be7078dd084641b7c358305832@192.168.42.2
CSeq: 1492389436 INVITE
Contact: "Membership" <sip:4142713656@192.168.42.2:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
Content-Length: 219

v=0
o=UserA 3884592705 2372130808 IN IP4 192.168.42.2
s=Session SDP
c=IN IP4 192.168.42.2
t=0 0
m=audio 49154 RTP/AVP 0 8 18
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
696136mS SIP Rx: UDP 66.23.129.253:5060 -> 192.168.42.2:5060
SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL)
Via: SIP/2.0/UDP 192.168.42.2:5060;rport=5060;branch=z9hG4bKc6029f4a14b2f309b099a97a91c2569b;received=67.53.0.87
From: "Membership" <sip:4143769110@nexvortex.com>;tag=b202340936270599
To: <sip:mad:192.168.42.1>;tag=9c6a9fdfd4d16ebaa52f34c4c528cbe5.26ed
Call-ID: 3cc032be7078dd084641b7c358305832@192.168.42.2
CSeq: 1492389436 INVITE
Server: nVSIP 12.02.01
Content-Length: 0

697053mS CMExtnEvt: Membership: Recover Timer reason=CMTRWrapUp
697053mS CMExtnEvt: v=1 State, new=Idle old=PortRecoverDelay,0,0,Membership
697054mS CMExtnTx: v=210, p1=0
CMVoiceMailStatus
Line: type=IPLine 250 Call: lid=0 id=-1 in=0
Called[Membership Msgs=0 Old=0 Sav=0] Type=Default (100) Reason=CMDRdirect Calling[00000000] Type=Default Plan=Default
Display [Membership Msgs=0]
Timed: 24/01/12 10:07
697781mS RES: Tue 24/1/2012 10:07:34 FreeMem=67941728(2) CMMsg=5 (6) Buff=5200 960 999 7463 5 Links=7992
697781mS RES2: IP 500 V2 7.0(27) Tasks=40 RTEngine=0 CMRTEngine=0 ExRTEngine=0 Timer=53 Poll=0 Ready=0 CMReady=0 CMQueue=0 VPNNQueue=0 Monitor=1 SSA=1 ASC=1 SYS=MNTD OPT=UMNT SDSPD=2034
704044mS SIP Tx: UDP 192.168.42.2:5060 -> 66.23.129.253:5060
INVITE sip:mad:192.168.42.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.42.2:5060;rport;branch=z9hG4bKc6029f4a14b2f309b099a97a91c2569b
From: "Membership" <sip:4143769110@nexvortex.com>;tag=b202340936270599
To: <sip:mad:192.168.42.1>
Call-ID: 3cc032be7078dd084641b7c358305832@192.168.42.2
CSeq: 1492389436 INVITE
Contact: "Membership" <sip:4142713656@192.168.42.2:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
Content-Length: 219

v=0
o=UserA 3884592705 2372130808 IN IP4 192.168.42.2
s=Session SDP
c=IN IP4 192.168.42.2
t=0 0
m=audio 49154 RTP/AVP 0 8 18
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
704140mS SIP Rx: UDP 66.23.129.253:5060 -> 192.168.42.2:5060
SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL)
Via: SIP/2.0/UDP 192.168.42.2:5060;rport=5060;branch=z9hG4bKc6029f4a14b2f309b099a97a91c2569b;received=67.53.0.87
From: "Membership" <sip:4143769110@nexvortex.com>;tag=b202340936270599
To: <sip:mad:192.168.42.1>;tag=9c6a9fdfd4d16ebaa52f34c4c528cbe5.26ed
Call-ID: 3cc032be7078dd084641b7c358305832@192.168.42.2
CSeq: 1492389436 INVITE
Server: nVSIP 12.02.01
Content-Length: 0


Does anyone know what is wrong?
 
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SIP 500 usually means that you are not recognized by the provider. Are you using registration or static public IP?

Kyle Holladay / IPOfficeHelp.com
ACSS/ACIS/APSS Avaya SME Communications
APDS Avaya Data
MCP/MCTS Exchange 2007/2010
Adtran ATSA, Aruba ACMA

"Thinking is the hardest work there is, which is the probable reason why so few engage in it." - Henry Ford
 
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Nexvortex is saying my SIP messages have the internal IP instead of the external IP. How do I setup the IP Office to use the external IP? The IP Office is connected to a Linksys RV042 router. Internet IP is dynamic.
 
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@jcallen03 Goto;

Line > SIP Line > Transport > Use Network Topology > Lan 1 or 2, use the one where the internet router is on.

Avaya_Red.gif

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool )
______________________________________
 
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@jcallen03 / HSM, I've never used a Stun server on any IPO i've ever installed. I always put it on 0.0.0.1 to turn it off. Make sure the 69.x.x.x is gone.

Avaya_Red.gif

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool )
______________________________________
 
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I think the STUN server helped. The IP Office did not know my public IP. I put a public STUN server in and my public IP information filled in. I have a dynamic IP so I could not fill it in myself.

Now I can make outbound calls. Caller ID says unavailable.

Inbound calls are still not working.
 
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This is what the monitor shows for incoming calls.

3339091mS SIP Tx: UDP 192.168.42.2:5060 -> 192.168.32.2:5060
OPTIONS sip:Unknown@192.168.32.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.42.2:5060;rport;branch=z9hG4bK886c1809eb9fbc7b193bfacb46d7f53a
From: <sip:Unknown@192.168.32.2>;tag=1b32d2611d6800b1
To: <sip:Unknown@192.168.32.2>
Call-ID: 5d7353252c80fb27aa39b3a3b0e440e2@192.168.42.2
CSeq: 2007623209 OPTIONS
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Length: 0

3343091mS SIP Tx: UDP 192.168.42.2:5060 -> 192.168.32.2:5060
OPTIONS sip:Unknown@192.168.32.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.42.2:5060;rport;branch=z9hG4bK886c1809eb9fbc7b193bfacb46d7f53a
From: <sip:Unknown@192.168.32.2>;tag=1b32d2611d6800b1
To: <sip:Unknown@192.168.32.2>
Call-ID: 5d7353252c80fb27aa39b3a3b0e440e2@192.168.42.2
CSeq: 2007623209 OPTIONS
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Length: 0

 
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Bas - you must use ITSPs that implement SBC, but there are many out there, including some larger ones here in the UK that still need STUN to work.

ACSS - SME
General Geek

CallUsOn.png


1832163.png
 
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@HSM, ever tried to put the ip to 0.0.0.0 or 0.0.0.1 ?
90% of the providers here have a SBC.
You need to fill in the public IP in that stun tab, that way it will use the public IP instead of the internal.

@jcallen03 add a URI and on all 3 pulldown menu's fill in a *

Avaya_Red.gif

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool )
______________________________________
 
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other method with registered trunks is URI set to use Authentication name in the drop down boxes and set an incoming line group ID that's unique and that you can set in the ICR to put the call to the hunt group you require. kinda turns it into a multi channel analogue trunk ay. and you can set short codes to use the relevant outgoing ID for particular dialing codes then.

The * URI entry, and the "Use Internal Data" entry work well on a direct IP SIP trunk,
* routes all incoming requests to ICR.
Use Internal Data seems to populate the "From" field info for outbound calls correctly.

Is that fairly accurate Bas /HSM?
 
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