hi there, im having with making calls from an outside extension with the softphone.
i have the 3cx pbx v9 and a grandstream GXW4104 firmware version
i properly configured the 3cx tunnel, opening the needed ports;
for sip 5060
for STUN 3478
for 3cx tunnel 5090
for rtp ports 9000 - 9049
but im not sure to wich device map to... pbx? or gateway?
actually i have every port maped to the pbx.
whell everything looks fine, even the softphone connects to the tunnel and displays on hook, if a dial an external number thru the softphone via 3cx tunnel, happens that it starts dialing after 1 second the call will be droped.
but it work, because the other line that i called, starts ringing. but after the softphone drops the call, it just ring 2 or 3 times more.
this is what it says the 3cx log:
23:05:49.765 [MS105000] C:26.2: No RTP packets were received:remoteAddr=189.231.12.246:9008,extAddr=0.0.0.0:0,localAddr=192.168.1.253:7120
23:05:48.609 [CM503008]: Call(26): Call is terminated
23:05:48.609 [CM303004]: Call(26).Leg(2): SIP ping has failed. Reason: 481 Call/Transaction Does Not Exist; from IP:189.231.12.246:5060; Terminating the call.
23:05:47.437 [CM503007]: Call(26): Device joined: sip:[email protected]:5060;transport=tcp
23:05:47.421 [CM503007]: Call(26): Device joined: sip:[email protected]:58784;rinstance=642591a8c4941d9e
23:05:45.921 [CM505002]: Gateway:[FXO Gateway 1] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXW4104 (HW 2.0, Ch:8) 1.3.4.10] PBX contact: [sip:[email protected]:5060;transport=TCP]
23:05:45.921 [CM503002]: Call(26): Alerting sip:[email protected]:5060;transport=tcp
23:05:38.968 [CM503025]: Call(26): Calling PSTNline:9932666805@(Ln.10000@FXO Gateway 1)@[Dev:sip:[email protected]:5060;transport=tcp]
23:05:38.921 [CM503004]: Call(26): Route 1: PSTNline:9932666805@(Ln.10000@FXO Gateway 1)@[Dev:sip:[email protected]:5060;transport=tcp,Dev:sip:[email protected]:5062;transport=tcp,Dev:sip:[email protected]:5064;transport=tcp]
23:05:38.906 [CM503010]: Making route(s) to <sip:[email protected]:5060>
23:05:38.906 [CM505001]: Ext.501: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone 5.0.14900.0] PBX contact: [sip:[email protected]:5060]
23:05:38.906 [CM503001]: Call(26): Incoming call from Ext.501 to <sip:[email protected]:5060>
so i disabled the STUN server.
so that improve the comunication, but once the call is established, after 30 secs, the call is droped.
this is what it says the 3cx log:
23:41:05.203 Currently active calls [none]
23:40:59.500 [CM503008]: Call(10): Call is terminated
23:40:59.500 [CM303005]: Call(10).Leg(1): SIP ping has timed out. Terminating the call.
23:40:33.203 Currently active calls - 1: [10]
23:40:31.437 Session 256 of leg C:10.1 is confirmed
23:40:31.015 [CM503007]: Call(10): Device joined: sip:[email protected]:5060;transport=udp
23:40:31.015 [CM503007]: Call(10): Device joined: sip:[email protected]:58784;rinstance=642591a8c4941d9e
23:40:31.015 [MS210003] C:10.1:Answer provided. Connection(transcoding mode[unsecure]):127.0.0.1:7042(7043)
23:40:31.015 [MS210001] C:10.2:Answer received. RTP connection[unsecure]: 192.168.1.10:9000(9001)
23:40:31.015 Remote SDP is set for legC:10.2
23:40:31.000 [CM505002]: Gateway:[FXO Gateway 1] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXW4104 (HW 2.0, Ch:8) 1.3.4.10] PBX contact: [sip:[email protected]:5060]
23:40:31.000 [CM503002]: Call(10): Alerting sip:[email protected]:5060;transport=udp
23:40:24.437 [CM503025]: Call(10): Calling PSTNline:9932666805@(Ln.10000@FXO Gateway 1)@[Dev:sip:[email protected]:5060;transport=udp]
23:40:24.437 [MS210002] C:10.2:Offer provided. Connection(transcoding mode): 192.168.1.253:7044(7045)
23:40:24.390 [CM503004]: Call(10): Route 1: PSTNline:9932666805@(Ln.10000@FXO Gateway 1)@[Dev:sip:[email protected]:5060;transport=udp,Dev:sip:[email protected]:5062;transport=udp,Dev:sip:[email protected]:5064;transport=udp]
23:40:24.375 [CM503010]: Making route(s) to <sip:[email protected]:5060>
23:40:24.375 [MS210000] C:10.1:Offer received. RTP connection: 127.0.0.1:10326(10327)
23:40:24.375 Remote SDP is set for legC:10.1
23:40:24.375 [CM505001]: Ext.501: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone 5.0.14900.0] PBX contact: [sip:[email protected]:5060]
23:40:24.375 [CM503001]: Call(10): Incoming call from Ext.501 to <sip:[email protected]:5060>
23:40:24.375 [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-d8754z-67231c481f2fd211-2---d8754z-;rport=5080;received=192.168.1.253
Via: SIP/2.0/UDP 189.231.82.8:59172;branch=z9hG4bK-d8754z-tunneltid-1---d8754z-;rport
Via: SIP/2.0/UDP 127.0.0.1:58784;branch=z9hG4bK-d8754z-67231c481f2fd211-1---d8754z-;rport=58784
Max-Forwards: 68
Record-Route: <sip:[email protected]:5080;user=proxy;uri="189.231.82.8:59172">
Contact: <sip:[email protected]:58784;rinstance=642591a8c4941d9e>
To: <sip:[email protected]:5060>
From: "501"<sip:[email protected]:5060>;tag=e1000304
Call-ID: MWE0MmU3OTgzOThkYTM2Y2Q1YTBiYjE4NmQ5YzJmOWY.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Proxy-Authorization: Digest username="501",realm="3CXPhoneSystem",nonce="414d535c043aefc746:e522463a092ef0cafef5820b095ded33",uri="sip:[email protected]:5060",response="382fb0e708083c15ca9469063cea4456",algorithm=MD5
Supported: replaces
User-Agent: 3CXPhone 5.0.14900.0
Content-Length: 0
what brings my attention is where says "SIP ping has failed" on both logs.
i dont know what im doing wrong.
hope anyone could help me, on the mean time i will try to do more research
regards!
i have the 3cx pbx v9 and a grandstream GXW4104 firmware version
i properly configured the 3cx tunnel, opening the needed ports;
for sip 5060
for STUN 3478
for 3cx tunnel 5090
for rtp ports 9000 - 9049
but im not sure to wich device map to... pbx? or gateway?
actually i have every port maped to the pbx.
whell everything looks fine, even the softphone connects to the tunnel and displays on hook, if a dial an external number thru the softphone via 3cx tunnel, happens that it starts dialing after 1 second the call will be droped.
but it work, because the other line that i called, starts ringing. but after the softphone drops the call, it just ring 2 or 3 times more.
this is what it says the 3cx log:
23:05:49.765 [MS105000] C:26.2: No RTP packets were received:remoteAddr=189.231.12.246:9008,extAddr=0.0.0.0:0,localAddr=192.168.1.253:7120
23:05:48.609 [CM503008]: Call(26): Call is terminated
23:05:48.609 [CM303004]: Call(26).Leg(2): SIP ping has failed. Reason: 481 Call/Transaction Does Not Exist; from IP:189.231.12.246:5060; Terminating the call.
23:05:47.437 [CM503007]: Call(26): Device joined: sip:[email protected]:5060;transport=tcp
23:05:47.421 [CM503007]: Call(26): Device joined: sip:[email protected]:58784;rinstance=642591a8c4941d9e
23:05:45.921 [CM505002]: Gateway:[FXO Gateway 1] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXW4104 (HW 2.0, Ch:8) 1.3.4.10] PBX contact: [sip:[email protected]:5060;transport=TCP]
23:05:45.921 [CM503002]: Call(26): Alerting sip:[email protected]:5060;transport=tcp
23:05:38.968 [CM503025]: Call(26): Calling PSTNline:9932666805@(Ln.10000@FXO Gateway 1)@[Dev:sip:[email protected]:5060;transport=tcp]
23:05:38.921 [CM503004]: Call(26): Route 1: PSTNline:9932666805@(Ln.10000@FXO Gateway 1)@[Dev:sip:[email protected]:5060;transport=tcp,Dev:sip:[email protected]:5062;transport=tcp,Dev:sip:[email protected]:5064;transport=tcp]
23:05:38.906 [CM503010]: Making route(s) to <sip:[email protected]:5060>
23:05:38.906 [CM505001]: Ext.501: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone 5.0.14900.0] PBX contact: [sip:[email protected]:5060]
23:05:38.906 [CM503001]: Call(26): Incoming call from Ext.501 to <sip:[email protected]:5060>
so i disabled the STUN server.
so that improve the comunication, but once the call is established, after 30 secs, the call is droped.
this is what it says the 3cx log:
23:41:05.203 Currently active calls [none]
23:40:59.500 [CM503008]: Call(10): Call is terminated
23:40:59.500 [CM303005]: Call(10).Leg(1): SIP ping has timed out. Terminating the call.
23:40:33.203 Currently active calls - 1: [10]
23:40:31.437 Session 256 of leg C:10.1 is confirmed
23:40:31.015 [CM503007]: Call(10): Device joined: sip:[email protected]:5060;transport=udp
23:40:31.015 [CM503007]: Call(10): Device joined: sip:[email protected]:58784;rinstance=642591a8c4941d9e
23:40:31.015 [MS210003] C:10.1:Answer provided. Connection(transcoding mode[unsecure]):127.0.0.1:7042(7043)
23:40:31.015 [MS210001] C:10.2:Answer received. RTP connection[unsecure]: 192.168.1.10:9000(9001)
23:40:31.015 Remote SDP is set for legC:10.2
23:40:31.000 [CM505002]: Gateway:[FXO Gateway 1] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXW4104 (HW 2.0, Ch:8) 1.3.4.10] PBX contact: [sip:[email protected]:5060]
23:40:31.000 [CM503002]: Call(10): Alerting sip:[email protected]:5060;transport=udp
23:40:24.437 [CM503025]: Call(10): Calling PSTNline:9932666805@(Ln.10000@FXO Gateway 1)@[Dev:sip:[email protected]:5060;transport=udp]
23:40:24.437 [MS210002] C:10.2:Offer provided. Connection(transcoding mode): 192.168.1.253:7044(7045)
23:40:24.390 [CM503004]: Call(10): Route 1: PSTNline:9932666805@(Ln.10000@FXO Gateway 1)@[Dev:sip:[email protected]:5060;transport=udp,Dev:sip:[email protected]:5062;transport=udp,Dev:sip:[email protected]:5064;transport=udp]
23:40:24.375 [CM503010]: Making route(s) to <sip:[email protected]:5060>
23:40:24.375 [MS210000] C:10.1:Offer received. RTP connection: 127.0.0.1:10326(10327)
23:40:24.375 Remote SDP is set for legC:10.1
23:40:24.375 [CM505001]: Ext.501: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone 5.0.14900.0] PBX contact: [sip:[email protected]:5060]
23:40:24.375 [CM503001]: Call(10): Incoming call from Ext.501 to <sip:[email protected]:5060>
23:40:24.375 [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-d8754z-67231c481f2fd211-2---d8754z-;rport=5080;received=192.168.1.253
Via: SIP/2.0/UDP 189.231.82.8:59172;branch=z9hG4bK-d8754z-tunneltid-1---d8754z-;rport
Via: SIP/2.0/UDP 127.0.0.1:58784;branch=z9hG4bK-d8754z-67231c481f2fd211-1---d8754z-;rport=58784
Max-Forwards: 68
Record-Route: <sip:[email protected]:5080;user=proxy;uri="189.231.82.8:59172">
Contact: <sip:[email protected]:58784;rinstance=642591a8c4941d9e>
To: <sip:[email protected]:5060>
From: "501"<sip:[email protected]:5060>;tag=e1000304
Call-ID: MWE0MmU3OTgzOThkYTM2Y2Q1YTBiYjE4NmQ5YzJmOWY.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Proxy-Authorization: Digest username="501",realm="3CXPhoneSystem",nonce="414d535c043aefc746:e522463a092ef0cafef5820b095ded33",uri="sip:[email protected]:5060",response="382fb0e708083c15ca9469063cea4456",algorithm=MD5
Supported: replaces
User-Agent: 3CXPhone 5.0.14900.0
Content-Length: 0
what brings my attention is where says "SIP ping has failed" on both logs.
i dont know what im doing wrong.
hope anyone could help me, on the mean time i will try to do more research
regards!
