search for: svtinc

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2004 Jan 21
11
Digium X100P for $43
Digium X100P / new cards are is available on ebay for $43. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3073050567&category=3309 <http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3073050567&category=3309 > Hope this helps to who want to play with X100P! Are these being sold by Digium ? I don't know ?? - SamW -------------- next part -------------- An HTML
2004 Jan 23
2
Latest cvs * compile error anyone?
I downloaded asterisk and was trying to compile fresh, It end up in error, Any help appreciated. cvs checkout asterisk cd asterisk make clean make END UP with following error, (Previously I was able to compile without any errors. After a make clean stopped compiling.) gcc -shared -Xlinker -x -o chan_iax2.so chan_iax2.o iax2-parser.o -lmysqlclient -lz /usr/bin/ld: cannot find -lmysqlclient
2004 Jan 05
2
Codec Negotiation Does not seem to work as expected ?? Help Please !!
Hello, I have been trying to get my coders to work without a conversion. I have read all the available asterisk documentation and support groups without any luck. Here is my issue. (Please feel free to ask questions if you do not understand what I am talking about.) I am using Cisco ATA-186 set to g729 codec. (But it will switch to g711 if sip-server request g711) I have 2 SIP-services to
2004 Jan 11
0
NuFone Network H323 configuration?
I am using Nu Fone Network's h323 drivers. I can place H323 calls using following in extensions.conf file, exten => _1732.,1,Dial(H323/${EXTEN}@192.168.1.2) If I need to use h323.conf to do the same I cannot configure h323 to do the same. I get everyone is busy message and I do not see IP packets being generated by * trying to communicate to 192.168.1.2. Can someone point out what I
2004 Jan 12
2
'*' call conference?
I read the feature list of asterisk and I cannot see if it is possible to conference a call between extensions. Is it a supported feature of asterisk or is it done in the UA (ATA186 in my case) Here is what I try to do. phone-a -dial-> phone-b tap the cradle (flash on phone-a) phone-a -dial-> phone-c tap the cradle (flash on phone-a) Now I like all 3 phones in a conference call.
2004 Jan 05
0
Codec Negotiation Does not seem to work as e xpected ?? Help Please !!
Steve, My Problem is not a problem, with the codec negotiation between end points. But when asterisk does it with canreinvite=no, * do not do it right. I replied with a lengthy discussion about my findings here, This behavior can be reproduced. But '*' do not seem to do the negotiation correctly. http://lists.digium.com/pipermail/asterisk-users/2004-January/032197.html