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After two full days of fiddling with the settings I figured it was time to post my problem here... hopefully someone can shed some light on this.
The situation is as follows: we've always had 3CX version 8 running happily at the office, but for several reasons we decided to move it out of the office and install version 9 on a remote server in a data center. So far, so good... but the two Aastra 6731i devices we have are now "external" extensions because of this change, which - as I found out the hard way - makes life a whole lot more difficult in VoIP-land.
I found out that registering wasn't working as expected (that, and phones weren't ringing for some reason) so enabled STUN along with Rport. The STUN server is stun.3cx.com:3478, for the record. Registering and ringing in both ways is working just fine, but we're still having one-way audio. The calling party hears us, but we do not hear the calling party... so the upload RTP stream isn't getting to the phone. Thinking it was a firewall issue I took one of the phones to my home, plugged it in and voila, it registered just fine with our old, version 8 install, and it had two-way audio out of the box. Perfectly, but very unexpected. Switch the server details to our new version 9 install... one-way audio. [ edit: this no longer applies ]
The remote server is on a static IP (obviously), and routers at both the office and home location are completely locked-down devices... quite common here in the Netherlands, sadly enough. It wouldn't mind giving up if I could say it was just another unsolvable firewall issue, but it strikes me that audio worked both ways when going remotely to the version 8 install. There seems to be a difference between the two installs, somehow.
I've taken a few random entries from the log (called from Aastra to mobile device, 0-ed in the logs); please let me know if you need specific entries and I'd be happy to share them.
The situation is as follows: we've always had 3CX version 8 running happily at the office, but for several reasons we decided to move it out of the office and install version 9 on a remote server in a data center. So far, so good... but the two Aastra 6731i devices we have are now "external" extensions because of this change, which - as I found out the hard way - makes life a whole lot more difficult in VoIP-land.
I found out that registering wasn't working as expected (that, and phones weren't ringing for some reason) so enabled STUN along with Rport. The STUN server is stun.3cx.com:3478, for the record. Registering and ringing in both ways is working just fine, but we're still having one-way audio. The calling party hears us, but we do not hear the calling party... so the upload RTP stream isn't getting to the phone. Thinking it was a firewall issue I took one of the phones to my home, plugged it in and voila, it registered just fine with our old, version 8 install, and it had two-way audio out of the box. Perfectly, but very unexpected. Switch the server details to our new version 9 install... one-way audio. [ edit: this no longer applies ]
The remote server is on a static IP (obviously), and routers at both the office and home location are completely locked-down devices... quite common here in the Netherlands, sadly enough. It wouldn't mind giving up if I could say it was just another unsolvable firewall issue, but it strikes me that audio worked both ways when going remotely to the version 8 install. There seems to be a difference between the two installs, somehow.
I've taken a few random entries from the log (called from Aastra to mobile device, 0-ed in the logs); please let me know if you need specific entries and I'd be happy to share them.
Code:
INVITE sip:0000000000@[server-public]:5060 SIP/2.0
Via: SIP/2.0/UDP [home-public]:49514;branch=z9hG4bK69c814492f5e655d1;rport=49514
Max-Forwards: 70
Contact: <sip:101@[home-public]:49514;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D1364DE>";isfocus
To: "0000000000"<sip:0000000000@[server-public]:5060>
From: <sip:101@[server-public]:5060>;tag=27cae1a3a3
Call-ID: dc55d861953fbf7f
CSeq: 25754 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Proxy-Authorization: Digest username="101",realm="3CXPhoneSystem",nonce="414d535c02e23e9734:f3b70a2ffcc566d1df4da854afe58958",uri="sip:0000000000@[server-public]:5060",response="551a731d7a44ffa7983dad38086195a8",algorithm=MD5
Supported: gruu, path, timer, 100rel, replaces
User-Agent: Aastra 6731i/2.6.0.1008
Allow-Events: talk, hold, conference, LocalModeStatus
Content-Length: 0
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP [home-public]:49514;branch=z9hG4bK69c814492f5e655d1;rport=49514
Contact: <sip:0000000000@[server-public]:5060>
To: "0000000000"<sip:0000000000@[server-public]:5060>;tag=f90a3a2d
From: <sip:101@[server-public]:5060>;tag=27cae1a3a3
Call-ID: dc55d861953fbf7f
CSeq: 25754 INVITE
Content-Type: application/sdp
User-Agent: 3CXPhoneSystem 9.0.14474.0
Content-Length: 272
SIP/2.0 200 OK
Via: SIP/2.0/UDP [home-public]:49514;branch=z9hG4bK69c814492f5e655d1;rport=49514
Contact: <sip:[email protected]:5060>
To: "0000000000"<sip:0000000000@[server-public]:5060>;tag=f90a3a2d
From: <sip:101@[server-public]:5060>;tag=27cae1a3a3
Call-ID: dc55d861953fbf7f
CSeq: 25754 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhoneSystem 9.0.14474.0
Content-Length: 272