- Joined
- Feb 26, 2008
- Messages
- 2
- Reaction score
- 0
Hi,
I've looked through all the help in this forum for my problem but nothing I have read has fixed it.
I can receive calls from PSTN through Linksys SPA-3102 to VOIP.
I can not make calls from VOIP through the Linksys to PSTN.
The error in the log is:
16:43:20.229 Call::Terminate [CM503008]: Call(51): Call is terminated
16:43:20.198 Call::RouteFailed [CM503014]: Call(51): Attempt to reach [sip:[email protected]] failed. Reason: Server Failure
16:43:20.198 CallLeg:
nFailure [CM503003]: Call(51): Call to sip:[email protected]:5060 has failed; Cause: 503 Service Unavailable; from IP:192.168.0.20:5060
16:43:20.151 MediaServerReporting::SetRemoteParty [MS210006] C:51.2:Offer provided. Connection(by pass mode): 192.168.0.25:60000(60001)
16:43:20.120 CallCtrl:
nSelectRouteReq [CM503004]: Call(51): Calling: PSTNline:10001@[Dev:sip:[email protected]:5060]
16:43:20.104 CallCtrl:
nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]]
16:43:20.104 MediaServerReporting::SetRemoteParty [MS210000] C:51.1:Offer received. RTP connection: 192.168.0.25:60000(60001)
16:43:20.104 CallLeg::setRemoteSdp Remote SDP is set for legC:51.1
16:43:20.104 Extension:
rintEndpointInfo [CM505001]: Ext.104: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [CMI CM5K] Transport: [sip:192.168.0.10:5060]
16:43:20.089 CallCtrl:
nIncomingCall [CM503001]: Call(51): Incoming call from Ext.104 to [sip:[email protected]]
16:43:20.089 CallLeg:
nNewCall [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25:5060;rport=5060;branch=z9hG4bK1cb44c31b714c801a1ee360a58de5509
Max-Forwards: 70
Contact: [sip:[email protected]:5060]
To: [sip:[email protected]]
From: "104"[sip:[email protected]];tag=490057ec
Call-ID: [email protected]
CSeq: 802 INVITE
Allow: INVITE, CANCEL, ACK, BYE, NOTIFY, REFER, OPTIONS, INFO, MESSAGE
Proxy-Authorization: Digest username="104",realm="3CXPhoneSystem",nonce="12848517799:d6612554064966ab07a8ecf11769c3b0",response="d9bd5ab09ef21ec8c04bd3a7821bf901",uri="sip:[email protected]",algorithm=MD5
User-Agent: CMI CM5K
Content-Length: 0
You'll see I'm dialing 8105, and the intended number I want to call is 105 - so it taking off the first digit as I wanted. 105 is a weird number to dial but trust me that number, if dialed direct on the wall socket goes through to the number I want it to, so when I dial 8105 on a VOIP phone and it sends across 105 that is accurate.
I either get a 404 error, or a 503 error. On the Linksys the PSTN and Line 1 ports are not the same port, and the I have updated the the port in 3cx in the line gateway setup.
I tried to telnet to the Linksys on port 5060 and 5061 and it doesn't respond, yet it seems to through 3cx - don't know if that is important or not.
I've tried everything - can anyone help? Not sure if the error lies in 3cx or Linksys, although I presume linksys.
Thanks in advance.
Pani
I've looked through all the help in this forum for my problem but nothing I have read has fixed it.
I can receive calls from PSTN through Linksys SPA-3102 to VOIP.
I can not make calls from VOIP through the Linksys to PSTN.
The error in the log is:
16:43:20.229 Call::Terminate [CM503008]: Call(51): Call is terminated
16:43:20.198 Call::RouteFailed [CM503014]: Call(51): Attempt to reach [sip:[email protected]] failed. Reason: Server Failure
16:43:20.198 CallLeg:
16:43:20.151 MediaServerReporting::SetRemoteParty [MS210006] C:51.2:Offer provided. Connection(by pass mode): 192.168.0.25:60000(60001)
16:43:20.120 CallCtrl:
16:43:20.104 CallCtrl:
16:43:20.104 MediaServerReporting::SetRemoteParty [MS210000] C:51.1:Offer received. RTP connection: 192.168.0.25:60000(60001)
16:43:20.104 CallLeg::setRemoteSdp Remote SDP is set for legC:51.1
16:43:20.104 Extension:
16:43:20.089 CallCtrl:
16:43:20.089 CallLeg:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25:5060;rport=5060;branch=z9hG4bK1cb44c31b714c801a1ee360a58de5509
Max-Forwards: 70
Contact: [sip:[email protected]:5060]
To: [sip:[email protected]]
From: "104"[sip:[email protected]];tag=490057ec
Call-ID: [email protected]
CSeq: 802 INVITE
Allow: INVITE, CANCEL, ACK, BYE, NOTIFY, REFER, OPTIONS, INFO, MESSAGE
Proxy-Authorization: Digest username="104",realm="3CXPhoneSystem",nonce="12848517799:d6612554064966ab07a8ecf11769c3b0",response="d9bd5ab09ef21ec8c04bd3a7821bf901",uri="sip:[email protected]",algorithm=MD5
User-Agent: CMI CM5K
Content-Length: 0
You'll see I'm dialing 8105, and the intended number I want to call is 105 - so it taking off the first digit as I wanted. 105 is a weird number to dial but trust me that number, if dialed direct on the wall socket goes through to the number I want it to, so when I dial 8105 on a VOIP phone and it sends across 105 that is accurate.
I either get a 404 error, or a 503 error. On the Linksys the PSTN and Line 1 ports are not the same port, and the I have updated the the port in 3cx in the line gateway setup.
I tried to telnet to the Linksys on port 5060 and 5061 and it doesn't respond, yet it seems to through 3cx - don't know if that is important or not.
I've tried everything - can anyone help? Not sure if the error lies in 3cx or Linksys, although I presume linksys.
Thanks in advance.
Pani