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...teway? This seems a bit harsh, Have i got everything mixed up? I'm not a H.323 expoert so I don't know if someone is breaking protocol in the chain Cisco GW -> GnuGK -> Asterisk. Anyone with Ideas? Anyone using Asterisk with H323 support in a GW mode? Regards, Niclas Gustafsson CodeSense AB

Hello! I've got a question regarding the Q.931 Setup-field Calling Party Number. It contains five things: Type of number, Number Plan, Presentation and Screening indicators and the actual number. Our provider uses some of those to decide if the numer should be presented or not to the outside world. I've done a crude hack in our GnuGK to always change those so that our numbers are

Hi, Is there any (easy) way to get Asterisk to include CIC-information in the SIP INVITE? CIC: http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vvfax_c/callc_c/sip_c/sipc1_c/chapter3.htm#1314580 I need my SIP INVITE to look something like: INVITE sip:5550001;cic=+16789@172.18.202.60:5060 SIP/2.0 I'v tried a couple of different things but can't find anything

Hello, I just can't seem to get the MusicOnHold function to work out ok. I' managed to get the MP3Player app to work out fine, but when I run the MusicOnHold all i get is siliece. I can see that Asterisk executes mpg123 properly (I think) #ps axuww|grep mp gk 4383 0.0 0.4 3736 552 pts/4 S 15:06 0:00 /usr/bin/mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 sample-hold.mp3

Hello! Anyone with some insight into the oh323 channel driver please shed some light on the code block below from wrapendpoint.cxx. When enabling trace on the channel driver i get this, for me, strange debug info: WrapH323EndPoint::OpenAudioChannel: Direction => PLAYER, Buffer => 320 WrapH323EndPoint::OpenAudioChannel: FrameSize 8, FrameTime 8, TimeUnits 8