I have a remote Polycom 501 that I cannot get to answer inbound calls. The phone will register and I can make outbound calls. When I dial the Ext it does not ever show ringing on the line status page. I think I have all of the required ports open, 5060 9000-9015.
Any ideas would be great as this is driving me carazy!
Here is the log of a call to the ext 106 which is online and ready/
Message
09:28:26.322 Call::Terminate [CM503008]: Call(5): Call is terminated
09:28:19.111 MediaServerReporting::SetRemoteParty [MS210004] C:5.2:Offer provided. Connection(proxy mode): 71.29.222.28:9004(9005)
09:28:18.761 CallCtrl:
nSelectRouteReq [CM503004]: Call(5): Calling: Ext:106@[Dev:sip:[email protected]:5060]
09:28:18.761 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected];user=phone]
09:28:18.751 MediaServerReporting::SetRemoteParty [MS210000] C:5.1:Offer received. RTP connection: 192.168.1.100:2228(2229)
09:28:18.751 CallLeg::setRemoteSdp Remote SDP is set for legC:5.1
09:28:18.751 Extension:rintEndpointInfo [CM505001]: Ext.100: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084] Transport: [sip:192.168.1.104:5060]
09:28:18.721 CallCtrl:nIncomingCall [CM503001]: Call(5): Incoming call from Ext.100 to [sip:[email protected];user=phone]
09:28:18.460 CallLeg:nNewCall [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK8341724F92B8B3
Max-Forwards: 70
Contact: [sip:[email protected]:5060]
To: [sip:[email protected];user=phone]
From: "100"[sip:[email protected]];tag=58CD8CF5-9B9DE92C
Call-ID: [email protected]
CSeq: 2 INVITE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Proxy-Authorization: Digest username="100",realm="3CXPhoneSystem",nonce="12852307697:87795fff08742192d1adc42ea51ea06a",uri="sip:[email protected]:5060;user=phone",response="053c26b1b50f9d76b41095100d22cce8",algorithm=MD5
Supported: 100rel, replaces
User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084
Allow-Events: talk, hold, conference
Content-Length: 0
Thanks!
Any ideas would be great as this is driving me carazy!
Here is the log of a call to the ext 106 which is online and ready/
Message
09:28:26.322 Call::Terminate [CM503008]: Call(5): Call is terminated
09:28:19.111 MediaServerReporting::SetRemoteParty [MS210004] C:5.2:Offer provided. Connection(proxy mode): 71.29.222.28:9004(9005)
09:28:18.761 CallCtrl:
nSelectRouteReq [CM503004]: Call(5): Calling: Ext:106@[Dev:sip:[email protected]:5060] 09:28:18.761 CallCtrl:
nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected];user=phone] 09:28:18.751 MediaServerReporting::SetRemoteParty [MS210000] C:5.1:Offer received. RTP connection: 192.168.1.100:2228(2229)
09:28:18.751 CallLeg::setRemoteSdp Remote SDP is set for legC:5.1
09:28:18.751 Extension:
rintEndpointInfo [CM505001]: Ext.100: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084] Transport: [sip:192.168.1.104:5060] 09:28:18.721 CallCtrl:
nIncomingCall [CM503001]: Call(5): Incoming call from Ext.100 to [sip:[email protected];user=phone] 09:28:18.460 CallLeg:
nNewCall [CM500002]: Info on incoming INVITE:INVITE sip:[email protected]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK8341724F92B8B3
Max-Forwards: 70
Contact: [sip:[email protected]:5060]
To: [sip:[email protected];user=phone]
From: "100"[sip:[email protected]];tag=58CD8CF5-9B9DE92C
Call-ID: [email protected]
CSeq: 2 INVITE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Proxy-Authorization: Digest username="100",realm="3CXPhoneSystem",nonce="12852307697:87795fff08742192d1adc42ea51ea06a",uri="sip:[email protected]:5060;user=phone",response="053c26b1b50f9d76b41095100d22cce8",algorithm=MD5
Supported: 100rel, replaces
User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084
Allow-Events: talk, hold, conference
Content-Length: 0
Thanks!
Phone System 