Hi Guys, Bit of an odd one here. We've been using UDP as the Transport Protocol for a Good while with a number of different SIP Phones but main Yealink SIP-T22P's. Due to some recently changes around the network needing to make things a little more robust we've had to look at shifting registration to TCP (and eventually to TLS but I have this issue to master first). We've had our SIPPhones working on multiple sites using UDP on a setup similar to this: [sipPhone]----------[SonicwallTMG]--------------(internet)---------------[SonicwallTMG] --------[Freeswitch] When changing to TCP phones can register fine and call to the outside world via freeswitch and our gateway providers however when attempting to make a call from the freeswitch box to the sip phone the call times out with NORMAL_TEMPORARY_FAILURE. This would suggest that the problem is related to NATing but we have exactly the same NAT rules for UDP as TCP in both firewalls and calls from the Freeswitch box with UDP can reach the sipPhone fine. Also doing a local test with TCP/TLS in our office works fine with STUN disabled and no other firewalls / routers in the way. I've tried a combination of various phones and configurations all with STUN enabled but as yet to no avail. I'm going to dig deeper on this tomorrow however I thought I'd send something out here to see if there is something simple that I've missed. The problem stinks of an issue with basic configuration of either Freeswitch or the firewalls we have in place but I can't see anything immediately that could be causing an issue as fundamental as this. If Anyone has any suggestions or has experienced this before I'd appreciate any help or direction on offer! Thanks in Advance Rob.
Rob this is related to FS-3877 (Reuse TCP connection for both directions) freeswitch will attempt to establish a TCP connection back to the sip phone when sending it that incoming call and that is not working causing the failure would be the best guess. Ensuring the proper contact port may allow it to reuse this connection and not try to create this reverse connect, but otherwise ensuring the FS server can properly connect back to the client will probably solve it. Crank sofia loglevel should show the exact issue. ~Mitch
Awesome thanks Mitch, I'll give that a blast. Thanks for the advice, I'll reply with how I get on in a bit -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Mitch Capper Sent: 29 March 2012 03:03 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Registration VIA TCP Rob this is related to FS-3877 (Reuse TCP connection for both directions) freeswitch will attempt to establish a TCP connection back to the sip phone when sending it that incoming call and that is not working causing the failure would be the best guess. Ensuring the proper contact port may allow it to reuse this connection and not try to create this reverse connect, but otherwise ensuring the FS server can properly connect back to the client will probably solve it. Crank sofia loglevel should show the exact issue. ~Mitch _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (2)
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Mitch Capper
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Rob Moore