Displaying 6 results from an estimated 6 matches for "svtinc".
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svbin
2004 Jan 21
11
Digium X100P for $43
Digium X100P / new cards are is available on ebay for $43.
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3073050567&category=3309
<http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3073050567&category=3309
>
Hope this helps to who want to play with X100P! Are these being sold by
Digium ? I don't know ??
- SamW
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An HTML
2004 Jan 23
2
Latest cvs * compile error anyone?
I downloaded asterisk and was trying to compile fresh, It end up in
error, Any help appreciated.
cvs checkout asterisk
cd asterisk
make clean
make
END UP with following error, (Previously I was able to compile without
any errors. After a make clean stopped compiling.)
gcc -shared -Xlinker -x -o chan_iax2.so chan_iax2.o iax2-parser.o
-lmysqlclient -lz
/usr/bin/ld: cannot find -lmysqlclient
2004 Jan 05
2
Codec Negotiation Does not seem to work as expected ?? Help Please !!
Hello,
I have been trying to get my coders to work without a conversion. I have
read all the available asterisk documentation and support groups without
any luck. Here is my issue. (Please feel free to ask questions if you do
not understand what I am talking about.)
I am using Cisco ATA-186 set to g729 codec. (But it will switch to g711 if
sip-server request g711)
I have 2 SIP-services to
2004 Jan 11
0
NuFone Network H323 configuration?
I am using Nu Fone Network's h323 drivers.
I can place H323 calls using following in extensions.conf file,
exten => _1732.,1,Dial(H323/${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