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Problem with OCS -> PBX & PSTN Calls


i3Q Systems
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Hi,

 

I've followed the guide in the WIKI to commission pbxnsipwith OCS 2007. My environment includes the OCS mediation server running on a virtual machine on the PBXNSIP box:

 

OCS Server = 10.150.100.206

OCS Mediation Server = 10.150.100.208

PBXNSIP = 10.150.100.207

 

So far I can place calls from pbxnsip to the PSTN and to OCS 2007 users. However, when I try to place a call from OCS to a pbxnsip user the OCS phone gives a ringing signal and nothing happens at the pbxnsip handset. At the point when I end the OCS call the pbxnsip handset briefly rings (1 second) and shows the CLI for the call. Similarly, if an OCS user places a call to the outside world the destination phone doesn't ring but the OCS handset continues to indicate that the call is in progress.

 

Log file below:

 

9] 2008/10/19 20:30:02: SIP Rx tcp:10.150.100.208:1882:

INVITE sip:5959@10.150.100.207;user=phone SIP/2.0

FROM: <sip:+448444125901@ocs2007-b.i3q.local;user=phone>;epid=92DD4034CF;tag=787dddd41e

TO: <sip:5959@10.150.100.207;user=phone>

CSEQ: 3 INVITE

CALL-ID: aaa78f53-bd6b-408a-9ed7-c817b74c209b

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 10.150.100.208:1882;branch=z9hG4bKb9bef14c

CONTACT: <sip:ocs2007-b.i3q.local:5060;transport=Tcp;maddr=10.150.100.208;ms-opaque=7e364f61608d9cf6>

CONTENT-LENGTH: 308

SUPPORTED: 100rel

USER-AGENT: RTCC/3.0.0.0 MediationServer

CONTENT-TYPE: application/sdp; charset=utf-8

ALLOW: UPDATE

ALLOW: Ack, Cancel, Bye,Invite

 

v=0

o=- 0 0 IN IP4 10.150.100.208

s=session

c=IN IP4 10.150.100.208

b=CT:1000

t=0 0

m=audio 60192 RTP/AVP 97 101 0 8

c=IN IP4 10.150.100.208

a=rtcp:60193

a=label:Audio

a=rtpmap:97 RED/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=ptime:20

 

[9] 2008/10/19 20:30:02: UDP: Opening socket on port 49992

[9] 2008/10/19 20:30:02: UDP: Opening socket on port 49993

[8] 2008/10/19 20:30:02: Could not find a trunk (2 trunks)

[9] 2008/10/19 20:30:02: Resolve 3899: tcp 10.150.100.208 1882

[9] 2008/10/19 20:30:02: SIP Tx tcp:10.150.100.208:1882:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 10.150.100.208:1882;branch=z9hG4bKb9bef14c

From: <sip:+448444125901@ocs2007-b.i3q.local;user=phone>;tag=787dddd41e;epid=92DD4034CF

To: <sip:5959@10.150.100.207;user=phone>;tag=a5c8bccdb1

Call-ID: aaa78f53-bd6b-408a-9ed7-c817b74c209b

CSeq: 3 INVITE

Content-Length: 0

 

 

[9] 2008/10/19 20:30:02: Resolve 3900: tcp 10.150.100.208 1882

[9] 2008/10/19 20:30:02: SIP Tx tcp:10.150.100.208:1882:

SIP/2.0 401 Authentication Required

Via: SIP/2.0/TCP 10.150.100.208:1882;branch=z9hG4bKb9bef14c

From: <sip:+448444125901@ocs2007-b.i3q.local;user=phone>;tag=787dddd41e;epid=92DD4034CF

To: <sip:5959@10.150.100.207;user=phone>;tag=a5c8bccdb1

Call-ID: aaa78f53-bd6b-408a-9ed7-c817b74c209b

CSeq: 3 INVITE

User-Agent: pbxnsip-PBX/3.0.1.3023

WWW-Authenticate: Digest realm="ocs2007-b.i3q.local",nonce="2d6b402435b608391b2aaead32e24530",domain="sip:5959@10.150.100.207;user=phone",algorithm=MD5

Content-Length: 0

 

 

[9] 2008/10/19 20:30:07: SIP Rx tcp:10.150.100.208:1882:

CANCEL sip:5959@10.150.100.207;user=phone SIP/2.0

FROM: <sip:+448444125901@ocs2007-b.i3q.local;user=phone>;tag=787dddd41e;epid=92DD4034CF

TO: <sip:5959@10.150.100.207;user=phone>

CSEQ: 3 CANCEL

CALL-ID: aaa78f53-bd6b-408a-9ed7-c817b74c209b

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 10.150.100.208:1882;branch=z9hG4bKb9bef14c

CONTENT-LENGTH: 0

 

 

[9] 2008/10/19 20:30:07: Resolve 3901: tcp 10.150.100.208 1882

[9] 2008/10/19 20:30:07: SIP Tx tcp:10.150.100.208:1882:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 10.150.100.208:1882;branch=z9hG4bKb9bef14c

From: <sip:+448444125901@ocs2007-b.i3q.local;user=phone>;tag=787dddd41e;epid=92DD4034CF

To: <sip:5959@10.150.100.207;user=phone>;tag=a5c8bccdb1

Call-ID: aaa78f53-bd6b-408a-9ed7-c817b74c209b

CSeq: 3 CANCEL

Content-Length: 0

 

 

[7] 2008/10/19 20:30:07: Attendant: Calling extension 5959

[7] 2008/10/19 20:30:07: Cannot convert number 5901 into global format

[8] 2008/10/19 20:30:07: Play audio_moh/noise.wav

[9] 2008/10/19 20:30:07: UDP: Opening socket on port 49690

[9] 2008/10/19 20:30:07: UDP: Opening socket on port 49691

[9] 2008/10/19 20:30:07: Using outbound proxy sip:10.151.1.110:2060;transport=udp because of flow-label

[9] 2008/10/19 20:30:07: Resolve 3902: url sip:10.151.1.110:2060;transport=udp

[9] 2008/10/19 20:30:07: Resolve 3902: a udp 10.151.1.110 2060

[9] 2008/10/19 20:30:07: Resolve 3902: udp 10.151.1.110 2060

[9] 2008/10/19 20:30:07: SIP Tx udp:10.151.1.110:2060:

INVITE sip:5959@10.151.1.110:2060;line=uj2nq50p SIP/2.0

Via: SIP/2.0/UDP 10.150.100.207:5060;branch=z9hG4bK-3769dbfa325b98892300e17758ccb78e;rport

From: "Steven Gee" <sip:5901@ocs2007-b.i3q.local>;tag=33095

To: "Steven Gee" <sip:5959@ocs2007-b.i3q.local>

Call-ID: aa266aad@pbx

CSeq: 1577 INVITE

Max-Forwards: 70

Contact: <sip:5959@10.150.100.207:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.1.3023

Alert-Info: <http://127.0.0.1/Bellcore-dr2>

Content-Type: application/sdp

Content-Length: 294

 

v=0

o=- 23991 23991 IN IP4 10.150.100.207

s=-

c=IN IP4 10.150.100.207

t=0 0

m=audio 49690 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[9] 2008/10/19 20:30:07: Resolve 3903: tcp 10.150.100.208 1882

[9] 2008/10/19 20:30:07: SIP Tx tcp:10.150.100.208:1882:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 10.150.100.208:1882;branch=z9hG4bKb9bef14c

From: <sip:+448444125901@ocs2007-b.i3q.local;user=phone>;tag=787dddd41e;epid=92DD4034CF

To: <sip:5959@10.150.100.207;user=phone>;tag=a5c8bccdb1

Call-ID: aaa78f53-bd6b-408a-9ed7-c817b74c209b

CSeq: 3 CANCEL

Contact: <sip:+448444125901@10.150.100.207:5060;transport=tcp>

User-Agent: pbxnsip-PBX/3.0.1.3023

Content-Length: 0

 

 

[3] 2008/10/19 20:30:07: Via is empty, cannot send reply

[6] 2008/10/19 20:30:07: No valid source address for sending email to user

[9] 2008/10/19 20:30:07: Resolve 3905: udp 10.151.1.110 2060

[9] 2008/10/19 20:30:07: SIP Tx udp:10.151.1.110:2060:

CANCEL sip:5959@10.151.1.110:2060;line=uj2nq50p SIP/2.0

Via: SIP/2.0/UDP 10.150.100.207:5060;branch=z9hG4bK-3769dbfa325b98892300e17758ccb78e;rport

From: "Steven Gee" <sip:5901@ocs2007-b.i3q.local>;tag=33095

To: "Steven Gee" <sip:5959@ocs2007-b.i3q.local>

Call-ID: aa266aad@pbx

CSeq: 1577 CANCEL

Max-Forwards: 70

Content-Length: 0

 

 

[3] 2008/10/19 20:30:07: Via is empty, cannot send reply

[9] 2008/10/19 20:30:07: SIP Rx udp:10.151.1.110:2060:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 10.150.100.207:5060;branch=z9hG4bK-3769dbfa325b98892300e17758ccb78e;rport=5060

From: "Steven Gee" <sip:5901@ocs2007-b.i3q.local>;tag=33095

To: "Steven Gee" <sip:5959@ocs2007-b.i3q.local>;tag=i6odvrl6ys

Call-ID: aa266aad@pbx

CSeq: 1577 INVITE

Contact: <sip:5959@10.151.1.110:2060;line=uj2nq50p>;flow-id=1

Require: 100rel

RSeq: 1

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Content-Length: 0

 

 

[5] 2008/10/19 20:30:07: Call aa266aad@pbx#33095: Last request not finished

[9] 2008/10/19 20:30:07: Resolve 3907: url sip:10.151.1.110:2060;transport=udp

[9] 2008/10/19 20:30:07: Resolve 3907: a udp 10.151.1.110 2060

[9] 2008/10/19 20:30:07: Resolve 3907: udp 10.151.1.110 2060

[9] 2008/10/19 20:30:07: SIP Tx udp:10.151.1.110:2060:

PRACK sip:5959@10.151.1.110:2060;line=uj2nq50p SIP/2.0

Via: SIP/2.0/UDP 10.150.100.207:5060;branch=z9hG4bK-6512016dd53e4e86ea9d51dcc913519b;rport

From: "Steven Gee" <sip:5901@ocs2007-b.i3q.local>;tag=33095

To: "Steven Gee" <sip:5959@ocs2007-b.i3q.local>;tag=i6odvrl6ys

Call-ID: aa266aad@pbx

CSeq: 1578 PRACK

Max-Forwards: 70

Contact: <sip:5959@10.150.100.207:5060;transport=udp>

RAck: 1 1577 INVITE

Content-Length: 0

 

 

[9] 2008/10/19 20:30:07: SIP Tr udp:10.151.1.110:2060:

CANCEL sip:5959@10.151.1.110:2060;line=uj2nq50p SIP/2.0

Via: SIP/2.0/UDP 10.150.100.207:5060;branch=z9hG4bK-3769dbfa325b98892300e17758ccb78e;rport

From: "Steven Gee" <sip:5901@ocs2007-b.i3q.local>;tag=33095

To: "Steven Gee" <sip:5959@ocs2007-b.i3q.local>

Call-ID: aa266aad@pbx

CSeq: 1577 CANCEL

Max-Forwards: 70

Content-Length: 0

 

 

[9] 2008/10/19 20:30:07: SIP Rx udp:10.151.1.110:2060:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.150.100.207:5060;branch=z9hG4bK-3769dbfa325b98892300e17758ccb78e;rport=5060

From: "Steven Gee" <sip:5901@ocs2007-b.i3q.local>;tag=33095

To: "Steven Gee" <sip:5959@ocs2007-b.i3q.local>;tag=i6odvrl6ys

Call-ID: aa266aad@pbx

CSeq: 1577 CANCEL

Content-Length: 0

 

 

[8] 2008/10/19 20:30:07: Call aa266aad@pbx#33095: Response does not correspond to open request

[9] 2008/10/19 20:30:07: SIP Rx udp:10.151.1.110:2060:

SIP/2.0 487 Request Terminated

Via: SIP/2.0/UDP 10.150.100.207:5060;branch=z9hG4bK-3769dbfa325b98892300e17758ccb78e;rport=5060

From: "Steven Gee" <sip:5901@ocs2007-b.i3q.local>;tag=33095

To: "Steven Gee" <sip:5959@ocs2007-b.i3q.local>;tag=i6odvrl6ys

Call-ID: aa266aad@pbx

CSeq: 1577 INVITE

Contact: <sip:5959@10.151.1.110:2060;line=uj2nq50p>;flow-id=1

Content-Length: 0

 

 

[7] 2008/10/19 20:30:07: Call aa266aad@pbx#33095: Clear last INVITE

[9] 2008/10/19 20:30:07: Resolve 3908: url sip:10.151.1.110:2060;transport=udp

[9] 2008/10/19 20:30:07: Resolve 3908: a udp 10.151.1.110 2060

[9] 2008/10/19 20:30:07: Resolve 3908: udp 10.151.1.110 2060

[9] 2008/10/19 20:30:07: SIP Tx udp:10.151.1.110:2060:

ACK sip:5959@10.151.1.110:2060;line=uj2nq50p SIP/2.0

Via: SIP/2.0/UDP 10.150.100.207:5060;branch=z9hG4bK-3769dbfa325b98892300e17758ccb78e;rport

From: "Steven Gee" <sip:5901@ocs2007-b.i3q.local>;tag=33095

To: "Steven Gee" <sip:5959@ocs2007-b.i3q.local>;tag=i6odvrl6ys

Call-ID: aa266aad@pbx

CSeq: 1577 ACK

Max-Forwards: 70

Contact: <sip:5959@10.150.100.207:5060;transport=udp>

Content-Length: 0

 

 

[5] 2008/10/19 20:30:07: INVITE Response: Terminate aa266aad@pbx

[7] 2008/10/19 20:30:07: Other Ports: 1

[7] 2008/10/19 20:30:07: Call Port: aaa78f53-bd6b-408a-9ed7-c817b74c209b#a5c8bccdb1

[9] 2008/10/19 20:30:08: SIP Tr udp:10.151.1.110:2060:

PRACK sip:5959@10.151.1.110:2060;line=uj2nq50p SIP/2.0

Via: SIP/2.0/UDP 10.150.100.207:5060;branch=z9hG4bK-6512016dd53e4e86ea9d51dcc913519b;rport

From: "Steven Gee" <sip:5901@ocs2007-b.i3q.local>;tag=33095

To: "Steven Gee" <sip:5959@ocs2007-b.i3q.local>;tag=i6odvrl6ys

Call-ID: aa266aad@pbx

CSeq: 1578 PRACK

Max-Forwards: 70

Contact: <sip:5959@10.150.100.207:5060;transport=udp>

RAck: 1 1577 INVITE

Content-Length: 0

 

 

[9] 2008/10/19 20:30:08: SIP Rx udp:10.151.1.110:2060:

SIP/2.0 481 Call Leg/Transaction Does Not Exist

Via: SIP/2.0/UDP 10.150.100.207:5060;branch=z9hG4bK-6512016dd53e4e86ea9d51dcc913519b;rport=5060

From: "Steven Gee" <sip:5901@ocs2007-b.i3q.local>;tag=33095

To: "Steven Gee" <sip:5959@ocs2007-b.i3q.local>;tag=i6odvrl6ys

Call-ID: aa266aad@pbx

CSeq: 1578 PRACK

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, callerid

Content-Length: 0

 

 

[9] 2008/10/19 20:30:08: SIP Rx udp:10.151.1.110:2060:

SIP/2.0 487 Request Terminated

Via: SIP/2.0/UDP 10.150.100.207:5060;branch=z9hG4bK-3769dbfa325b98892300e17758ccb78e;rport=5060

From: "Steven Gee" <sip:5901@ocs2007-b.i3q.local>;tag=33095

To: "Steven Gee" <sip:5959@ocs2007-b.i3q.local>;tag=i6odvrl6ys

Call-ID: aa266aad@pbx

CSeq: 1577 INVITE

Contact: <sip:5959@10.151.1.110:2060;line=uj2nq50p>;flow-id=1

Content-Length: 0

 

 

[9] 2008/10/19 20:30:08: SIP Tm udp:10.151.1.110:2060:

ACK sip:5959@10.151.1.110:2060;line=uj2nq50p SIP/2.0

Via: SIP/2.0/UDP 10.150.100.207:5060;branch=z9hG4bK-3769dbfa325b98892300e17758ccb78e;rport

From: "Steven Gee" <sip:5901@ocs2007-b.i3q.local>;tag=33095

To: "Steven Gee" <sip:5959@ocs2007-b.i3q.local>;tag=i6odvrl6ys

Call-ID: aa266aad@pbx

CSeq: 1577 ACK

Max-Forwards: 70

Contact: <sip:5959@10.150.100.207:5060;transport=udp>

Content-Length: 0

 

 

[9] 2008/10/19 20:30:08: Message repetition, packet dropped

[9] 2008/10/19 20:30:08: SIP Rx udp:10.151.1.110:2060:

SIP/2.0 481 Call Leg/Transaction Does Not Exist

Via: SIP/2.0/UDP 10.150.100.207:5060;branch=z9hG4bK-6512016dd53e4e86ea9d51dcc913519b;rport=5060

From: "Steven Gee" <sip:5901@ocs2007-b.i3q.local>;tag=33095

To: "Steven Gee" <sip:5959@ocs2007-b.i3q.local>;tag=i6odvrl6ys

Call-ID: aa266aad@pbx

CSeq: 1578 PRACK

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, callerid

Content-Length: 0

 

 

[9] 2008/10/19 20:30:08: Message repetition, packet dropped

 

 

Regards

Steven

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That concerns me. If the PBX cannot identify the call comes from a trunk then it will not work. What is the outbound proxy? Did you include the transport=tcp parameter in the outbound proxy.

 

Sorry, do you mean the trunk configuration in pbxnsip? If so I've set-up outbound proxy to be ocs2007-b.i3q.local (our OCS mediation server). In OCS Mediation server the next hop PSTN is specified as the IP address of the pbxnsip box 10.150.100.207 on port 5060.

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Hi i3Q Systems,

 

what kind of virtualization solution you are using for the OCS Mediation Server virtual machine? How is it connected to the rest of your enviroment? Virtual internal Network or a real Card from the host system?

 

It is really strange the pbxnsip "Could not find a trunk (2 trunks)" identify's two associated trunks, when a call from OCS Mediation Server is coming in.

 

Please re-check also your DNS-Records including Reverse LookUp's.

 

Did you configured your trunks (one to a VoIP-Provider/Gatway and one to OCS mediation Server) with IP-Adresses or DNS-names? Please also re-check if the trunks are correctly configured.

 

Best regards,

 

Jan

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Hi i3Q Systems,

 

what kind of virtualization solution you are using for the OCS Mediation Server virtual machine? How is it connected to the rest of your enviroment? Virtual internal Network or a real Card from the host system?

 

It is really strange the pbxnsip "Could not find a trunk (2 trunks)" identify's two associated trunks, when a call from OCS Mediation Server is coming in.

 

Please re-check also your DNS-Records including Reverse LookUp's.

 

Did you configured your trunks (one to a VoIP-Provider/Gatway and one to OCS mediation Server) with IP-Adresses or DNS-names? Please also re-check if the trunks are correctly configured.

 

Best regards,

 

Jan

Hi,

 

I'm using Microsoft Virtual Server to host the mediation server and it's using a virtual network interface. I configured the trunks using DNS names within pbxnsip and have two - one to OCS and one to thid party service provider. I think the trunks are configured correctly.

 

Any other thoughts please?

 

Thanks

Steven

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Sorry, do you mean the trunk configuration in pbxnsip? If so I've set-up outbound proxy to be ocs2007-b.i3q.local (our OCS mediation server). In OCS Mediation server the next hop PSTN is specified as the IP address of the pbxnsip box 10.150.100.207 on port 5060.

 

Yes, the trunk config in pbxnsip. That message above says "I got a request and I have no idea where it comes from". If the PBX cannot identify the trunk then there may be something wrong with the trunk outbound proxy. For example a famous problem is that the outbound proxy is set to the IP address of the host; when the request comes in the IP address of the request is 127.0.0.1, which does not match that IP address.

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Hi

 

I am still not sure what is causing the problems, but I think a test is worth trying. If possible in your enviroment, copy the OCS mediation server VHD-file to another VirtualServer Host-System (separate to pbxnsip). Stop the OCS mediation server @pbxnsip-VS-Host and start the Copy. Please try calling inside, outside and internal again.

 

Best regards,

Jan

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