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January 3rd, 2006, 01:50 AM
| | Junior Member | | Join Date: Dec 2005
Posts: 5
| | Strange problem with SIP extension, asterisk and a vpn I have setup a test system running asterisk-1.0.8. I have setup xten-lite as an extension and I have that working ask expected when connecting from the local network. When I connect via a vpn (no firewall blocking rules through the vpn) and I call from that extension, I can't hear anything on the other end until I send sound from my end (ie tap the mic or say hello etc.). Once I say something, things seem to work normally. Any idea what I'm missing? | 
January 5th, 2006, 04:13 AM
| | Junior Member | | Join Date: Dec 2005
Posts: 5
| | RE: Strange problem with SIP extension, asterisk and a vpn I have found a little more information. While xten-lite has this problem, I am able to use kphone without the same symptoms. The only problem with kphone is very poor sound quality. Does anyone have any ideas on what I might need to change on xten-lite's setup to get it to work properly? | 
January 5th, 2006, 02:39 PM
|  | Senior Member | | Join Date: Aug 2004 Location: USA or Japan
Posts: 5,013
| | RE: Strange problem with SIP extension, asterisk and a vpn How much true bandwidth do you have through your VPN tunnel?
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January 5th, 2006, 02:44 PM
| | Junior Member | | Join Date: Dec 2005
Posts: 5
| | I don't know the specific throughput, not enough for kphone to sound good.  . It is enough for xten-lite, but that is the one that does not seem to listen unless I'm talking. I don't think that problem is related to throughput, but if you do, I can measure it and let you know. | 
January 5th, 2006, 04:16 PM
|  | Senior Member | | Join Date: Aug 2004 Location: USA or Japan
Posts: 5,013
| | I was just grasping. 'Tis a puzzler.
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January 13th, 2006, 04:02 AM
| | Member | | Join Date: Jan 2005 Location: Australia
Posts: 81
| | hakimian,
After trying to fix a problem that I thought was to do with SIP and TDM400 bridging, I now find my problem seems to be identical to yours. I'm also using xlite as a SIP phone into asterisk, out through a TDM400 into the PSTN. If I yell into the PSTN side, the connection comes to life. It seems asterisk doesnt answer the call properyl.
-- Executing Dial("SIP/532-30d8", "ZAP/35/92125532") in new stack
-- Called 35/92125532
-- Zap/35-1 answered SIP/532-30d8
The last line appears when I yell into the PSTN phone. Until I see that line I get no audio out of the PSTN. If I use IAX instead of SIP, it all works fine.
Did you find a solution to this problem ?
Thanks,
Andrew | 
January 13th, 2006, 04:45 AM
| | Member | | Join Date: Jan 2005 Location: Australia
Posts: 81
| | hakimian,
I went through a similar path as you. I downloaded another SIP softphone (SJPhone) and tried it. It works first time every time (I havent tried it too many times yet).
Must be some type of issue with xlite. Strange as I havent had problems with it before.
Andrew | 
January 13th, 2006, 12:16 PM
| | Junior Member | | Join Date: Dec 2005
Posts: 5
| | The problem does seem to be just with xlite. I have tried 3 or 4 other phones and they don't have the problem. Unfortunately, none of the other phones I have tried are as good as xlite in sound quality. |  | | Thread Tools | | | | Display Modes | Rate This Thread | Linear Mode | |
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