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- Feb 1, 2010
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Hi ! This is my first install and first problems
I'm trying to set up a sip trunk with a provider called COMS.com. I know it's not supported by 3cx but still believe I should be able to make it work
I have two problems:
1. when dial my VoIP number from my mobile - the call comes in, I pick it up - silence. The funny thing - my mobile doesn't realise that the call was answered - I still get ringing tone
here is the log
18:33:56.462 Looking for inbound target: called=02071111243; caller=02073883050
18:33:56.460 [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected]:5060;rinstance=8e2abad9f018aae8 SIP/2.0
Via: SIP/2.0/UDP 85.90.225.100;branch=z9hG4bK0f33.141c031d845501c4aa9ec4b0d1f139af.0
Via: SIP/2.0/UDP 85.90.225.100:5061;branch=z9hG4bKfc50344d489c73f717ef78cff502e2ad;rport=5061
Max-Forwards: 16
Record-Route: <sip:85.90.225.100;ftag=375981e35ecb7d3759075654fc276836o;lr>
Contact: "Anonymous"<sip:85.90.225.100:5061>
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=375981e35ecb7d3759075654fc276836o
Call-ID: [email protected]
CSeq: 200 INVITE
Expires: 300
User-Agent: Sippy
Content-Length: 0
cisco-GUID: 570762408-2141142455-618591599-2729389085
h323-conf-id: 570762408-2141142455-618591599-2729389085
Portabilling-notify: aor=5101402835
second problem - when i dial out, the call ends instantly
18:24:15.810 [MS105000] C:3.1: No RTP packets were received:remoteAddr=85.90.225.100:42104,extAddr=195.137.29.160:32350,localAddr=195.137.29.160:32350
18:24:08.902 [CM503008]: Call(5): Call is terminated
18:24:08.849 [CM503020]: Normal call termination. Reason: Server Failure
18:24:08.849 [CM503016]: Call(5): Attempt to reach <sip:[email protected]> failed. Reason: Server Failure
18:24:08.848 [CM503003]: Call(5): Call to sip:[email protected]:5060 has failed; Cause: 500 Server internal error (TM); from IP:85.90.225.100:5060
18:24:08.712 [CM503025]: Call(5): Calling VoIPline:02071149834@(Ln.10000@COMS)@[Dev:sip:[email protected]:5060]
18:24:08.708 [MS210002] C:5.2:Offer provided. Connection(transcoding mode): 195.17.129.160:32362(32363)
18:24:08.502 [CM503004]: Call(5): Route 1: VoIPline:02071149834@(Ln.10000@COMS)@[Dev:sip:[email protected]:5060]
18:24:08.500 [CM503010]: Making route(s) to <sip:[email protected]>
18:24:08.500 [MS210000] C:5.1:Offer received. RTP connection: 192.168.5.14:16400(16401)
18:24:08.498 Remote SDP is set for legC:5.1
18:24:08.497 [CM505001]: Ext.100: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:[email protected]:5060]
18:24:08.496 [CM503001]: Call(5): Incoming call from Ext.100 to <sip:[email protected]>
18:24:08.493 [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.5.14:5060;branch=z9hG4bK-eeaff3d0
Max-Forwards: 70
Contact: "100"<sip:[email protected]:5060>
To: <sip:[email protected]>
From: "100"<sip:[email protected]>;tag=d5f65b6b4dcd37bfo0
Call-ID: [email protected]
CSeq: 102 INVITE
Expires: 240
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Proxy-Authorization: Digest username="100",realm="3CXPhoneSystem",nonce="414d535c0177ad4812:81a656e27479e2aa96723b687e271f08",uri="sip:[email protected]",algorithm=MD5,response="befcaf9231ab0b3531b089ab35e9762b"
User-Agent: Linksys/SPA921-4.1.15
Content-Length: 0
am I missing something obvious?
I'm trying to set up a sip trunk with a provider called COMS.com. I know it's not supported by 3cx but still believe I should be able to make it work
I have two problems:
1. when dial my VoIP number from my mobile - the call comes in, I pick it up - silence. The funny thing - my mobile doesn't realise that the call was answered - I still get ringing tone
here is the log
18:33:56.462 Looking for inbound target: called=02071111243; caller=02073883050
18:33:56.460 [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected]:5060;rinstance=8e2abad9f018aae8 SIP/2.0
Via: SIP/2.0/UDP 85.90.225.100;branch=z9hG4bK0f33.141c031d845501c4aa9ec4b0d1f139af.0
Via: SIP/2.0/UDP 85.90.225.100:5061;branch=z9hG4bKfc50344d489c73f717ef78cff502e2ad;rport=5061
Max-Forwards: 16
Record-Route: <sip:85.90.225.100;ftag=375981e35ecb7d3759075654fc276836o;lr>
Contact: "Anonymous"<sip:85.90.225.100:5061>
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=375981e35ecb7d3759075654fc276836o
Call-ID: [email protected]
CSeq: 200 INVITE
Expires: 300
User-Agent: Sippy
Content-Length: 0
cisco-GUID: 570762408-2141142455-618591599-2729389085
h323-conf-id: 570762408-2141142455-618591599-2729389085
Portabilling-notify: aor=5101402835
second problem - when i dial out, the call ends instantly
18:24:15.810 [MS105000] C:3.1: No RTP packets were received:remoteAddr=85.90.225.100:42104,extAddr=195.137.29.160:32350,localAddr=195.137.29.160:32350
18:24:08.902 [CM503008]: Call(5): Call is terminated
18:24:08.849 [CM503020]: Normal call termination. Reason: Server Failure
18:24:08.849 [CM503016]: Call(5): Attempt to reach <sip:[email protected]> failed. Reason: Server Failure
18:24:08.848 [CM503003]: Call(5): Call to sip:[email protected]:5060 has failed; Cause: 500 Server internal error (TM); from IP:85.90.225.100:5060
18:24:08.712 [CM503025]: Call(5): Calling VoIPline:02071149834@(Ln.10000@COMS)@[Dev:sip:[email protected]:5060]
18:24:08.708 [MS210002] C:5.2:Offer provided. Connection(transcoding mode): 195.17.129.160:32362(32363)
18:24:08.502 [CM503004]: Call(5): Route 1: VoIPline:02071149834@(Ln.10000@COMS)@[Dev:sip:[email protected]:5060]
18:24:08.500 [CM503010]: Making route(s) to <sip:[email protected]>
18:24:08.500 [MS210000] C:5.1:Offer received. RTP connection: 192.168.5.14:16400(16401)
18:24:08.498 Remote SDP is set for legC:5.1
18:24:08.497 [CM505001]: Ext.100: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:[email protected]:5060]
18:24:08.496 [CM503001]: Call(5): Incoming call from Ext.100 to <sip:[email protected]>
18:24:08.493 [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.5.14:5060;branch=z9hG4bK-eeaff3d0
Max-Forwards: 70
Contact: "100"<sip:[email protected]:5060>
To: <sip:[email protected]>
From: "100"<sip:[email protected]>;tag=d5f65b6b4dcd37bfo0
Call-ID: [email protected]
CSeq: 102 INVITE
Expires: 240
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Proxy-Authorization: Digest username="100",realm="3CXPhoneSystem",nonce="414d535c0177ad4812:81a656e27479e2aa96723b687e271f08",uri="sip:[email protected]",algorithm=MD5,response="befcaf9231ab0b3531b089ab35e9762b"
User-Agent: Linksys/SPA921-4.1.15
Content-Length: 0
am I missing something obvious?