tobiastromm
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- Joined
- Jun 26, 2010
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Hi.
I am using a GrandStream GXW4104 to dial out over PSTN line.
My GXW4104 is over a Cliente side of a PPTP Site-to-Site VPN between two houses.
Everything is working fine, but a little thing is boring me.
My network is like that:
3CX Server is also PPTP Server with Local Network (10.0.0.1/24) and PPTP Bridge (10.1.1.1/24)
PPTP Client (10.1.1.2/24 - 192.168.2.1/24)
Networks 10.0.0.0/24 and 192.168.2.0/24 can ping each side.
The small problem is that to GXW4104 can dial out I need to register it with SIP Server 10.1.1.1 instead of the correct Server IP 10.0.0.1 (under Accounts -> General Settings).
I have two Grandstream 486 in PPPT Clieent side connected to 3CX Server and working OK with Sip Server 10.0.0.1.
Here is the error when set the Sip Server to 10.0.0.1:
22:19:24.924 [CM503020]: Normal call termination. Reason: No answer
22:19:24.924 [CM503016]: Call(63): Attempt to reach <sip:[email protected]> failed. Reason: No Answer
22:19:24.924 [CM503003]: Call(63): Call to sip:[email protected]:5060 has failed; Cause: 408 Request Timeout; internal
22:18:53.533 Currently active calls - 1: [63]
22:18:52.815 [CM503025]: Call(63): Calling PSTNline:10314@(Ln.10001@EVN - 34445652)@[Dev:sip:[email protected]:5060;transport=udp]
22:18:52.815 [MS210002] C:63.2:Offer provided. Connection(transcoding mode): 10.1.1.1:7296(7297)
22:18:52.768 [CM503004]: Call(63): Route 1: PSTNline:10314@(Ln.10001@EVN - 34445652)@[Dev:sip:[email protected]:5060;transport=udp]
22:18:52.768 [CM503010]: Making route(s) to <sip:[email protected]>
Here is the call ok when set the Sip Server to 10.1.1.1:
22:21:30.223 [CM503008]: Call(64): Call is terminated
22:21:16.332 Session 14450 of leg C:64.1 is confirmed
22:21:16.301 [CM503007]: Call(64): Device joined: sip:[email protected]:5060;transport=udp
22:21:16.301 [CM503007]: Call(64): Device joined: sip:[email protected]:20404
22:21:16.301 [MS210003] C:64.1:Answer provided. Connection(transcoding mode[unsecure]):10.0.0.1:7298(7299)
22:21:16.301 [MS210001] C:64.2:Answer received. RTP connection[unsecure]: 192.168.2.2:5004(5005)
22:21:16.301 Remote SDP is set for legC:64.2
22:21:15.457 [CM505002]: Gateway:[EVN - 34445652] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXW4104 (HW 2.0, Ch:8) 1.4.1.6] PBX contact: [sip:[email protected]:5060]
22:21:15.457 [CM503002]: Call(64): Alerting sip:[email protected]:5060;transport=udp
22:21:12.941 [CM503025]: Call(64): Calling PSTNline:10314@(Ln.10001@EVN - 34445652)@[Dev:sip:[email protected]:5060;transport=udp]
22:21:12.941 [MS210002] C:64.2:Offer provided. Connection(transcoding mode): 10.1.1.1:7300(7301)
22:21:12.894 [CM503004]: Call(64): Route 1: PSTNline:10314@(Ln.10001@EVN - 34445652)@[Dev:sip:[email protected]:5060;transport=udp]
22:21:12.894 [CM503010]: Making route(s) to <sip:[email protected]>
Can someone help?
Thanks.
I am using a GrandStream GXW4104 to dial out over PSTN line.
My GXW4104 is over a Cliente side of a PPTP Site-to-Site VPN between two houses.
Everything is working fine, but a little thing is boring me.
My network is like that:
3CX Server is also PPTP Server with Local Network (10.0.0.1/24) and PPTP Bridge (10.1.1.1/24)
PPTP Client (10.1.1.2/24 - 192.168.2.1/24)
Networks 10.0.0.0/24 and 192.168.2.0/24 can ping each side.
The small problem is that to GXW4104 can dial out I need to register it with SIP Server 10.1.1.1 instead of the correct Server IP 10.0.0.1 (under Accounts -> General Settings).
I have two Grandstream 486 in PPPT Clieent side connected to 3CX Server and working OK with Sip Server 10.0.0.1.
Here is the error when set the Sip Server to 10.0.0.1:
22:19:24.924 [CM503020]: Normal call termination. Reason: No answer
22:19:24.924 [CM503016]: Call(63): Attempt to reach <sip:[email protected]> failed. Reason: No Answer
22:19:24.924 [CM503003]: Call(63): Call to sip:[email protected]:5060 has failed; Cause: 408 Request Timeout; internal
22:18:53.533 Currently active calls - 1: [63]
22:18:52.815 [CM503025]: Call(63): Calling PSTNline:10314@(Ln.10001@EVN - 34445652)@[Dev:sip:[email protected]:5060;transport=udp]
22:18:52.815 [MS210002] C:63.2:Offer provided. Connection(transcoding mode): 10.1.1.1:7296(7297)
22:18:52.768 [CM503004]: Call(63): Route 1: PSTNline:10314@(Ln.10001@EVN - 34445652)@[Dev:sip:[email protected]:5060;transport=udp]
22:18:52.768 [CM503010]: Making route(s) to <sip:[email protected]>
Here is the call ok when set the Sip Server to 10.1.1.1:
22:21:30.223 [CM503008]: Call(64): Call is terminated
22:21:16.332 Session 14450 of leg C:64.1 is confirmed
22:21:16.301 [CM503007]: Call(64): Device joined: sip:[email protected]:5060;transport=udp
22:21:16.301 [CM503007]: Call(64): Device joined: sip:[email protected]:20404
22:21:16.301 [MS210003] C:64.1:Answer provided. Connection(transcoding mode[unsecure]):10.0.0.1:7298(7299)
22:21:16.301 [MS210001] C:64.2:Answer received. RTP connection[unsecure]: 192.168.2.2:5004(5005)
22:21:16.301 Remote SDP is set for legC:64.2
22:21:15.457 [CM505002]: Gateway:[EVN - 34445652] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXW4104 (HW 2.0, Ch:8) 1.4.1.6] PBX contact: [sip:[email protected]:5060]
22:21:15.457 [CM503002]: Call(64): Alerting sip:[email protected]:5060;transport=udp
22:21:12.941 [CM503025]: Call(64): Calling PSTNline:10314@(Ln.10001@EVN - 34445652)@[Dev:sip:[email protected]:5060;transport=udp]
22:21:12.941 [MS210002] C:64.2:Offer provided. Connection(transcoding mode): 10.1.1.1:7300(7301)
22:21:12.894 [CM503004]: Call(64): Route 1: PSTNline:10314@(Ln.10001@EVN - 34445652)@[Dev:sip:[email protected]:5060;transport=udp]
22:21:12.894 [CM503010]: Making route(s) to <sip:[email protected]>
Can someone help?
Thanks.