Hello Everybody,
I'm having a problem with a 3cx v 7.x (latest version) installation, on Windows Server 2003 R2, installed with 3cx cassini server, no AntiVirus, no Firewall, all LAN traffic, there are 2 network cards...
Voip Equipment:
2 Grandtream HT502 (each has 2 fxs ports... line/port 1 uses 5060 and line/port 2 uses 5062)
1 Grandstream GXW4104 (with 2 pstn lines hooked up to ports 1 and 3),
>> SIP destination port: ch1-4:5060;
>> SIP listen port: ch1-4:5060++; (the ++ increments the port number by +2 for each line...)
The incoming calls work great (forwards to the Digital Recep), but I´m having trouble with the outgoing calls:
When the extension A (using the first Grandstream FXS HT502) makes a call, the 3cx outbound rule recognizes it's extension number and routes the call via the FXO Line 1 (by adding the prefix 991)
The problem occurs when another user on the second HT502 dials, and the second OUtboud rule routes it to the other line (by appending the 993) .. the call order/command goes through to the GXW4104, it dials the PSTN number correctly, it rings on the destination phone, but when they answer there is no voice and the person who dialed the calls continues to hear ringing sound as if the call kept ringing on the destination phone....
I set the log to verbose, it didn´t work, so I manually reset the services in the services MMC, and I think it's logging correctly (see below, correct me if Im wrong and my logs aren't detailed yet..)
( I had to manually terminated/hangup becuase if not it would "seem" to keep on ringing even though someone answered..)
19:19:59.234 [CM503008]: Call(3): Call is terminated
19:19:45.609 [CM505002]: Gateway:[HYRCO4104A] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXW4104 (HW 1.1, Ch:9) 1.0.1.25] Transport: [sip:192.168.6.12:5060]
19:19:45.609 [CM503002]: Call(3): Alerting sip:[email protected]:5062;user=phone
19:19:43.562 [CM503004]: Call(3): Route 2: PSTNline:9943444804@(Ln.10000@HYRCO4104A)@[Dev:sip:[email protected]:5060, Dev:sip:[email protected]:5064;user=phone, Dev:sip:[email protected]:5066;user=phone]
19:19:43.562 [CM503024]: Call(3): Calling PSTNline:9933444804@(Ln.10001@HYRCO4104A)@[Dev:sip:[email protected]:5062;user=phone]
19:19:43.562 [MS210006] C:3.2:Offer provided. Connection(by pass mode): 192.168.6.50:5540(5541)
19:19:43.562 [CM503004]: Call(3): Route 1: PSTNline:9933444804@(Ln.10001@HYRCO4104A)@[Dev:sip:[email protected]:5062;user=phone, Dev:sip:[email protected]:5060, Dev:sip:[email protected]:5064;user=phone, Dev:sip:[email protected]:5066;user=phone]
19:19:43.546 [MS210000] C:3.1:Offer received. RTP connection: 192.168.6.50:5540(5541)
19:19:43.546 [CM503010]: Making route(s) to "03444804"<sip:[email protected]>
19:19:43.546 Remote SDP is set for legC:3.1
19:19:43.546 [CM505001]: Ext.2400: Device info: Device Identified: [Man: Counterpath;Mod: X-Lite;Rev: General] Capabilities:[reinvite, no-replaces, unable-no-sdp, recvonly] UserAgent: [X-Lite release 1103d stamp 53117] Transport: [sip:192.168.6.12:5060]
19:19:43.546 [CM503001]: Call(3): Incoming call from Ext.2400 to "03444804"<sip:[email protected]>
19:19:43.546 [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.6.50:42820;branch=z9hG4bK-d8754z-ea372660eb08f933-1---d8754z-;rport=42820
Max-Forwards: 70
Contact: <sip:[email protected]:42820>
To: "03444804"<sip:[email protected]>
From: "test ana"<sip:[email protected]>;tag=6c3e6e0f
Call-ID: ZjcyZDQ2N2QxYmM2MzZjYjcwZjM0NTJiZTVlZWJiYmQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Proxy-Authorization: Digest username="2400",realm="3CXPhoneSystem",nonce="414d535c009a849f19:1eddd9995bedf3faa5783ae96172a638",uri="sip:[email protected]",response="8d0941fdf472c33134dd8f45f888499d",algorithm=MD5
User-Agent: X-Lite release 1103d stamp 53117
Content-Length: 0
Anybody have any ideas ??
thanks
I'm having a problem with a 3cx v 7.x (latest version) installation, on Windows Server 2003 R2, installed with 3cx cassini server, no AntiVirus, no Firewall, all LAN traffic, there are 2 network cards...
Voip Equipment:
2 Grandtream HT502 (each has 2 fxs ports... line/port 1 uses 5060 and line/port 2 uses 5062)
1 Grandstream GXW4104 (with 2 pstn lines hooked up to ports 1 and 3),
>> SIP destination port: ch1-4:5060;
>> SIP listen port: ch1-4:5060++; (the ++ increments the port number by +2 for each line...)
The incoming calls work great (forwards to the Digital Recep), but I´m having trouble with the outgoing calls:
When the extension A (using the first Grandstream FXS HT502) makes a call, the 3cx outbound rule recognizes it's extension number and routes the call via the FXO Line 1 (by adding the prefix 991)
The problem occurs when another user on the second HT502 dials, and the second OUtboud rule routes it to the other line (by appending the 993) .. the call order/command goes through to the GXW4104, it dials the PSTN number correctly, it rings on the destination phone, but when they answer there is no voice and the person who dialed the calls continues to hear ringing sound as if the call kept ringing on the destination phone....
I set the log to verbose, it didn´t work, so I manually reset the services in the services MMC, and I think it's logging correctly (see below, correct me if Im wrong and my logs aren't detailed yet..)
( I had to manually terminated/hangup becuase if not it would "seem" to keep on ringing even though someone answered..)
19:19:59.234 [CM503008]: Call(3): Call is terminated
19:19:45.609 [CM505002]: Gateway:[HYRCO4104A] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXW4104 (HW 1.1, Ch:9) 1.0.1.25] Transport: [sip:192.168.6.12:5060]
19:19:45.609 [CM503002]: Call(3): Alerting sip:[email protected]:5062;user=phone
19:19:43.562 [CM503004]: Call(3): Route 2: PSTNline:9943444804@(Ln.10000@HYRCO4104A)@[Dev:sip:[email protected]:5060, Dev:sip:[email protected]:5064;user=phone, Dev:sip:[email protected]:5066;user=phone]
19:19:43.562 [CM503024]: Call(3): Calling PSTNline:9933444804@(Ln.10001@HYRCO4104A)@[Dev:sip:[email protected]:5062;user=phone]
19:19:43.562 [MS210006] C:3.2:Offer provided. Connection(by pass mode): 192.168.6.50:5540(5541)
19:19:43.562 [CM503004]: Call(3): Route 1: PSTNline:9933444804@(Ln.10001@HYRCO4104A)@[Dev:sip:[email protected]:5062;user=phone, Dev:sip:[email protected]:5060, Dev:sip:[email protected]:5064;user=phone, Dev:sip:[email protected]:5066;user=phone]
19:19:43.546 [MS210000] C:3.1:Offer received. RTP connection: 192.168.6.50:5540(5541)
19:19:43.546 [CM503010]: Making route(s) to "03444804"<sip:[email protected]>
19:19:43.546 Remote SDP is set for legC:3.1
19:19:43.546 [CM505001]: Ext.2400: Device info: Device Identified: [Man: Counterpath;Mod: X-Lite;Rev: General] Capabilities:[reinvite, no-replaces, unable-no-sdp, recvonly] UserAgent: [X-Lite release 1103d stamp 53117] Transport: [sip:192.168.6.12:5060]
19:19:43.546 [CM503001]: Call(3): Incoming call from Ext.2400 to "03444804"<sip:[email protected]>
19:19:43.546 [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.6.50:42820;branch=z9hG4bK-d8754z-ea372660eb08f933-1---d8754z-;rport=42820
Max-Forwards: 70
Contact: <sip:[email protected]:42820>
To: "03444804"<sip:[email protected]>
From: "test ana"<sip:[email protected]>;tag=6c3e6e0f
Call-ID: ZjcyZDQ2N2QxYmM2MzZjYjcwZjM0NTJiZTVlZWJiYmQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Proxy-Authorization: Digest username="2400",realm="3CXPhoneSystem",nonce="414d535c009a849f19:1eddd9995bedf3faa5783ae96172a638",uri="sip:[email protected]",response="8d0941fdf472c33134dd8f45f888499d",algorithm=MD5
User-Agent: X-Lite release 1103d stamp 53117
Content-Length: 0
Anybody have any ideas ??
thanks
