SIP trunk incoming call problem! 2

[abdullamahmud]

I have two IP Office 500V2 in two different location. One location (Location A) can make and receive call but the other location (Location B) can make call but unable to receive call with same configuration and same SIP provider. When i try to call the number from my cell phone it says the number is not valid. My PBX received the invite from remote and then said "404 Not Found". I take out the monitor log for both case and i saw the invite message is bit different between them. If any one have any idea about this please let me know. see the following for monitor details,

Successful log (INVITE and TRYING and RINGING): Location A

1287628872mS SIP Rx: UDP 69.26.45.13:5060 -> 10.12.1.200:5060
INVITE sip:17213206317@10.12.1.200:5060 SIP/2.0
Via: SIP/2.0/UDP 69.26.45.13:5060;branch=z9hG4bK1sansay1273302923rdb13371
Record-Route: <sip:sansay1273302923rdb13371@69.26.45.13:5060;lr;transport=udp>
To: <sip:17213206317@10.12.1.200>
From: AbdullaM <sip:17212057173@69.26.45.13>;tag=sansay1273302923rdb13371
Call-ID: 377224797-1-1890338634@69.26.45.12
CSeq: 1 INVITE
Contact: <sip:17212057173@69.26.45.13:5060>
Cisco-Guid: 1355039610-1868220491-563076731-59834452
Supported: timer,100rel
Session-Expires: 36000;refresher=uac
Min-SE: 90
P-Asserted-Identity: AbdullaM <sip:17212057173@10.0.131.228>
Remote-Party-ID: AbdullaM <sip:17212057173@10.0.131.228>;party=calling
Expires: 300
h323-conf-id: 1355039610-1868220491-563076731-59834452
Content-disposition: session
User-Agent: Sippy
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 299

v=0
o=Sansay-VSXi 188 1 IN IP4 69.26.45.13
s=Session Controller
c=IN IP4 69.26.45.13
t=0 0
m=audio 32918 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
2013-08-15T15:55:04 1287628876mS CMCallEvt: 0.27712.0 -1 BaseEP: NEW CMEndpoint f4d93800 TOTAL NOW=9 CALL_LIST=4
2013-08-15T15:55:04 1287628876mS CH : CHand::AllocateIPConnectionPoint cp[645]b0r0[TOTAL CP:141]
2013-08-15T15:55:04 1287628877mS VOIP: (Out) 0.27712.0 -1 SIPTrunk Endpoint: SetOutboundSecInfo, size 2
2013-08-15T15:55:04 1287628879mS SIP Tx: UDP 10.12.1.200:5060 -> 69.26.45.13:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 69.26.45.13:5060;branch=z9hG4bK1sansay1273302923rdb13371
Record-Route: <sip:sansay1273302923rdb13371@69.26.45.13:5060;lr;transport=udp>
From: AbdullaM <sip:17212057173@69.26.45.13>;tag=sansay1273302923rdb13371
To: <sip:17213206317@10.12.1.200>;tag=bfb9565df16a95e5
Call-ID: 377224797-1-1890338634@69.26.45.12
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Server: IP Office 8.1 (67)
Content-Length: 0
2013-08-15T15:55:04 1287628901mS SIP Tx: UDP 10.12.1.200:5060 -> 69.26.45.13:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 69.26.45.13:5060;branch=z9hG4bK1sansay1273302923rdb13371
Record-Route: <sip:sansay1273302923rdb13371@69.26.45.13:5060;lr;transport=udp>
From: AbdullaM <sip:17212057173@69.26.45.13>;tag=sansay1273302923rdb13371
To: <sip:17213206317@10.12.1.200>;tag=bfb9565df16a95e5
Call-ID: 377224797-1-1890338634@69.26.45.12
CSeq: 1 INVITE
Contact: "Extn4427" <sip:17213206317@10.12.1.200:5060;transport=udp>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Server: IP Office 8.1 (67)
Content-Length: 0

unSuccessful log (INVITE and TRYING and RINGING): Location B

4018843826mS SIP Rx: UDP 69.26.45.13:5060 -> 10.10.1.200:5060
INVITE sip:17217352619@12.219.224.27:5060 SIP/2.0
Via: SIP/2.0/UDP 69.26.45.13:5060;branch=z9hG4bK1sansay1273637994rdb10667
Record-Route: <sip:sansay1273637994rdb10667@69.26.45.13:5060;lr;transport=udp>
To: <sip:17217352619@12.219.224.27>
From: "STL COMMUNICAT" <sip:7212057173@69.26.45.13>;tag=sansay1273637994rdb10667
Call-ID: 377312587-0-1891759044@69.26.45.12
CSeq: 1 INVITE
Contact: <sip:7212057173@69.26.45.13:5060>
Cisco-Guid: 2821155162-1698270372-3854612192-3776580839
Supported: timer,100rel
Session-Expires: 36000;refresher=uac
Min-SE: 90
P-Asserted-Identity: "STL COMMUNICAT" <sip:7212057173@10.0.131.228>
Remote-Party-ID: "STL COMMUNICAT" <sip:7212057173@10.0.131.228>;party=calling
Expires: 300
h323-conf-id: 2821155162-1698270372-3854612192-3776580839
Content-disposition: session
User-Agent: Sippy
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 275

v=0
o=Sansay-VSXi 188 1 IN IP4 69.26.45.13
s=Session Controller
c=IN IP4 69.26.45.13
t=0 0
m=audio 19626 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
2013-08-15T15:18:47 4018843830mS CMCallEvt: 0.733887.0 -1 BaseEP: NEW CMEndpoint f4b1b770 TOTAL NOW=39 CALL_LIST=19
2013-08-15T15:18:47 4018843831mS CH : CHand::AllocateIPConnectionPoint cp[460]b0r0[TOTAL CP:320]
2013-08-15T15:18:47 4018843831mS VOIP: (In) 0.733887.0 -1 SIPTrunk Endpoint: SetOutboundSecInfo, size 2
2013-08-15T15:18:47 4018843834mS SIP Tx: UDP 10.10.1.200:5060 -> 69.26.45.13:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 69.26.45.13:5060;branch=z9hG4bK1sansay1273637994rdb10667
Record-Route: <sip:sansay1273637994rdb10667@69.26.45.13:5060;lr;transport=udp>
From: "STL COMMUNICAT" <sip:7212057173@69.26.45.13>;tag=sansay1273637994rdb10667
To: <sip:17217352619@12.219.224.27>;tag=bb203e9071506576
Call-ID: 377312587-0-1891759044@69.26.45.12
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Server: IP Office 8.1 (67)
Content-Length: 0

2013-08-15T15:18:47 4018843836mS SIP Tx: UDP 10.10.1.200:5060 -> 69.26.45.13:5060
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 69.26.45.13:5060;branch=z9hG4bK1sansay1273637994rdb10667
Record-Route: <sip:sansay1273637994rdb10667@69.26.45.13:5060;lr;transport=udp>
From: "STL COMMUNICAT" <sip:7122057173@69.26.45.13>;tag=sansay1273637994rdb10667
To: <sip:17217352619@12.219.224.27>;tag=bb203e9071506576
Call-ID: 377312587-0-1891759044@69.26.45.12
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Server: IP Office 8.1 (67)
Content-Length: 0

My question is how and why i have received two different SIP header unsuccessful call(INVITE sip:17217352619@12.219.224.27:5060 SIP/2.0) and successful call (INVITE sip:17213206317@10.12.1.200:5060 SIP/2.0)?

Thanks
AB

Abdulla Mahmud
ACSS, APSS, ACIS, MCP, dCAA
BS.c(CS), MS.c(IS)

 

[intrigrant]

Not much to go on but the successfull one uses the local IP address whil the unsuccesfull uses the WAN IP address.
I assume the unsuccessfull IP Office is not connected to the internet directly is it?

 

[abdullamahmud]

Both are connected to internet directly and both can ping SIP provider IP. None of them are using LAN2 or WAN port or public IP. I am looking for an answer how the SIP INVITE get changed for Location B.

Thanks
Abdulla Mahmud

Abdulla Mahmud
ACSS, APSS, ACIS, MCP, dCAA
BS.c(CS), MS.c(IS)

 

[intrigrant]

Don not be harsh as I first have to understand the setup which is necessery to understand the problem.
If you want a direct solution then call a expert and do not come here.

Your system 2 use the WAN ip address in the INVITE and your system 1 use the IP Office ip address, that is different and there is your problem.
That is my final reply, good luck!

 

[abdullamahmud]

Do not get me wrong. I was not harsh but unfortunately i have poor writing impression. I am really sorry for that. Thanks for your reply i will try to find out the problem.

Thanks
AB

Abdulla Mahmud
ACSS, APSS, ACIS, MCP, dCAA
BS.c(CS), MS.c(IS)

 

[intrigrant]

Thinking about it a 404 error means that the incoming number is not recognised, in your URI enter in the "Local URI" a * and make a Incoming Call route without a DID number and enter the same line ID as set in the URI, target it to a extension and see what happens.

 

[abdullamahmud]

Salute you.

It worked. Thank you very much for the solution. I should buy a lunch for you.

Thanks
AB

Abdulla Mahmud
ACSS, APSS, ACIS, MCP, dCAA
BS.c(CS), MS.c(IS)

 

[intrigrant]

Well, I think, according to your name, a lunch would become quite expensive so a star will do.

And glad for you it worked out and the language problem at the start happens more often so no problem there either...

have a nice day