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...2196 3e83 8ce6 3ed8 13ca 2722 1388 0028 4893 8062 b576 0000 7800 6cee c076 19c0 227f a415 aebc d7e5 eab3 e9cf 8ba9 fbb7 c039 So, each UDP package with 20 bytes speex-data, we send: 20 bytes speex 12 bytes ogg headers (and others?) 28 bytes UDP/IP headers (2 IP numerbers, 2 portnumbers, checksum, etc, etc) and, if it goes over the phone, each package has a few ppp headers. <p>Am I overlooking something, or does this fixed frame size mean that the "8kbps" in reality means at least 24kbps? Or is there something ohphone/gnomemeeting/etc can/should d...

Joost Witteveen (joost@iliana.nl) wrote: > > So, each UDP package with 20 bytes speex-data, we send: > > 20 bytes speex > 12 bytes ogg headers (and others?) > 28 bytes UDP/IP headers (2 IP numerbers, 2 portnumbers, checksum, etc, etc) > > and, if it goes over the phone, each package has a few ppp headers. > > Am I overlooking something, or does this fixed frame size mean that > the "8kbps" in reality means at least 24kbps? > > Or is there something ohphone/gno...

Hi all, I'm quite new on this mailing list, and I discover the asterisk world. I m experimenting a PBX with SIP phones, grandstream budgetone (not expensive for tests) All the features I need work just not one : the supervised call transfers. I know there are a lot of posts about that, but none gave me the correct answer (unless I missed it). Here is my config : 2 sip phones BT102 with

Hi, I'm a student and my thesis work consist in testing Asterisk with Swyx(SwyxWare). My approach is to declare asterisk as h323 gateway for the Swyxserver using oh323 Plugin. Is there any possibility to connect Asterisk with Swyx? how? the outgoing call must pass from Swyxit->to Swyxserver-> Asterisk->to PTSN Thanks