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I have created a sip trunk between a bcm 400 pbx and a openuc (eZuce) ip pbx server. The polycom phones are connected to the openuc server. They are able to call the bcm 400 phones, however, when the phones  on the bcm 400 call the polycom phones, you hear a busy tone (instead of a dialing tone) and the polycom phones continue to ring.

 

 

While observing the sip invite we noticed that there is no a=rtpmap. Here is a copy of the SIP INVITE:

 

Time: 2012-08-07T16:00:55.884461Z
Frame: 467639
Source: xxx.xxx.corp-SipXProxy
Dest: 10.10.128.20:5060

INVITE sip:5006@10.10.128.20;x-sipX-nonat SIP/2.0
Record-Route: <sip:10.10.0.202:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7ENDA0Y2NkNjgtY2EwYTAwMGEtMTNjNC01MDIxM2JiOC02ODlhOWUzMC01MDIxM2JiOA%60%60.900_ntap%2Aid%7ENzExMC0xOA%60%60%2184bf9b8c85160397ec2992b629f54396;sipXecs-CallDest=UNK%2CINT>
From: "XXX 3550"<sip:xxxxxx0040;phone-context=unknown@xxx.corp:5060;user=phone>;tag=404ccd68-ca0a000a-13c4-50213bb8-689a9e30-50213bb8
To: <sip:5006;phone-context=@xxx.corp:5060;user=phone>
Call-Id: 875dea0-ca0a000a-13c4-50213bb8-5f9c83ab-50213bb8@xxx.corp
Cseq: 1 INVITE
Via: SIP/2.0/TCP 10.10.0.202;branch=z9hG4bK-XX-3ee9CE8QzlmInQNO9o8osjIk4A
Via: SIP/2.0/UDP 10.10.0.202;branch=z9hG4bK-XX-3ee6_EfXX8_OCAg`3qx2sgZl1Q~vRe_tWUkxwfpw5LiM949`g;id=7110-18
Via: SIP/2.0/UDP 10.10.0.202;branch=z9hG4bK-XX-3ee0syyqwlJ`hzEC3MKMBK0vUw~xRN`JP7MkuC5xq9kOBvC4A
Via: SIP/2.0/UDP 10.10.0.202;branch=z9hG4bK-XX-3edbrPA8op2Zbc63_H6hIFKMXg~IYgc29cyhGYWC0nBRb1ctQ
Via: SIP/2.0/UDP 10.0.10.202:5060;branch=z9hG4bK-50213bb8-1d149cd-70ee60f9
Max-Forwards: 14
Supported: 100rel,sipvc
User-Agent: Nortel Networks BCM VoIP Gateway release_41 version_41.500.0.33
Privacy: None
X-Nt-Corr-Id: 875dea0-ca0a000a-13c4-50213bb8-5f9c83ab-50213bb8@xxx.corp
Contact: <sip:xxxxxx0040;phone-context=unknown@xxx.corp:5060;maddr=10.0.10.202;transport=udp;user=phone;x-sipX-nonat>
Allow: INVITE,UPDATE,INFO,ACK,OPTIONS,CANCEL,BYE,NOTIFY,PRACK
Content-Type: application/SDP
Content-Length: 168
Date: Tue, 07 Aug 2012 16:00:55 GMT
Expires: 20
X-Sipx-Handled: X10.10.0.202-10.10.0.202

v=0
o=- 1344355256 1344355256 IN IP4 10.0.10.67
s=-
t=0 0
m=audio 51006 RTP/AVP 0 18 8 4
c=IN IP4 10.0.10.67
a=fmtp:18 annexb=no
a=fmtp:4 annexa=no
a=ptime:30

1 REPLY 1
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Hello Delroy,

welcome to the Polycom Community.

Would this post not be more suitable in a Nortel Community?

 

Your post states that the Invite does not contain the attribute and the trace below shows the User-agent as the Nortel Device.

 

I would recommend to check with Nortel / Avaya and/or openuc.


Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

Polycom Global Services

------------------------------------------------
Notice: I am an HP Poly employee but all replies within the community are done as a volunteer outside of my day role. This community forum is not an official HP Poly support resource, thus responses from HP Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge.
If you need immediate and/or official assistance for former Poly\Plantronics\Polycom please open a service ticket through your support channels
For HP products please check HP Support.

Please also ensure you always check the General VoIP , Video Endpoint , UC Platform (Microsoft) , PSTN
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