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Addition to previus post Here is my sip.conf [skype] type=friend username=skype secret=xxxx context=from-skype host=dynamic canreinvite=yes callerid="Incoming Skype" port=5070 insecure=very disallow = all allow = alaw

Hello I have a question: Does Uplink use silence suppresion ? and if so can i disable it ? I have big problem with chopping sound (there is info in other topic) and im trying to find reason and solution... on http://www.voip-info.org/wiki-Asterisk+config+sip.conf a have found infomation: "Asterisk uses the incoming RTP Stream as a timing source for sending its outgoing Stream. If the incoming stream is interrupted due to silence suppression then musiconhold will be choppy. So in conclusion, you cannot use silence suppression. Make sure ALL SIP phones have disabled silence suppression. There is a solution for the silence suppression problem, see bug 5374 for details." PS. I don't have problem with MOH (Music on Hold), my problems exist in normal coversation. If you have any ideas for this problem PLEASE help! Thx in advice

Hello again and again... If you have problem with freezing Uplink (even with ver 1.21) and you have processor with HyperThreading or Dual Core then this feature could be a problem. After switching off HyperThreading (I didn't test it with Dual Core well) problem is solved. You can turn off HyperThreading (and probably Dual Core) in your BIOS. I have read on this forum that Ben is working on Uplink ver 1.3 with will solve this issue.

Hello again... I have a problem with sound from Uplink to other party. Situation look like this: 1. Skype caller --> Skype with Uplink --> Asterisk --> IP Phone When I call to Skype user i have a problem with chopping sound but only in one direction. Skype user hear clearly but IP Phone user hear chopping sound. 2. So I have made a test with connection like this: SIP Soft Phone ---> Asterisk --> IP Phone and there is no problem... So i think something is not right with voice coming from Uplink. Got any idea what could be wrong? Greetz

Hello all Uplink Guys! I have some sugestions about improving Uplink 1. Possiblity to choose between codec (ulaw,alaw, gsm...) VERY IMPORTANT 2. Possiblity to manipulate jitter buffer size VERY IMPORTANT 3. Silence suppresion VERY NICE 4. Sending DTMF codes (Skype --> Uplink --> Asterisk) VERY NICE 5. Changing UserAgent name to some thing else .... COSMETIC 6. Any other option that will allow user to manipulate SIP setting will be very nice... Thanks in advice Maciej J. Hajduk