search for: playsip

...should contain just a single Speex frame. - Some existing clients (such as "linphone") will need to be modified to use the string "speex" rather than "speex-<version-number>" in their SDP "a=rtpmap:" line. Also, FYI, my command-line SIP client "playSIP" <http://www.live.com/playSIP/> can also record - into a file - the Speex audio from a SIP call. (Unfortunately, this output file will be raw Speex data, rather than an 'ogg' format file, so "speexdec" currently can't decode it.) Ross. --- >8 ----...

HI I want to write my own SIP client (compatible with ASterisk) is there any good API available for this purpose . any help in this regard would be very helpful 4 me Obee _________________________________________________________________ Tired of spam? Get advanced junk mail protection with MSN 8. http://join.msn.com/?page=features/junkmail

Hi folks, I'm wondering if it is currently possible to configure Asterisk to forward the conversations from 2 channels into a streaming daemon, such as Icecast, so that other people connected to the Internet can listen. The concept is similar to a radio talk-show. The show host would connect to Asterisk via an X100P or through VOIP. His or her voice would then be sent to the streaming

...to > use the string "speex" rather than "speex-<version-number>" in their SDP > "a=rtpmap:" line. The latest version of Linphone had been made up-to-date with the RTP spec, so this is no longer a problem. > Also, FYI, my command-line SIP client "playSIP" > <http://www.live.com/playSIP/> can also record - into a file - the Speex > audio from a SIP call. (Unfortunately, this output file will be raw Speex > data, rather than an 'ogg' format file, so "speexdec" currently can't > decode it.) I see a po...

Hello All, Is it possible to monitor and record a SIP to SIP call? If so, how? I gathered from some previous posts this would not be possible. -- Thanks, Tim

...he sampling frequency is also the RTP timestamp frequency, so if Asterisk thinks that the stream is 8 kHz, but the Speex client thinks that it's 16 kHz, then each side will probably misinterpret the RTP timestamps sent by the other side. > > Also, FYI, my command-line SIP client "playSIP" > > <http://www.live.com/playSIP/> can also record - into a file - the Speex > > audio from a SIP call. (Unfortunately, this output file will be raw Speex > > data, rather than an 'ogg' format file, so "speexdec" currently can't > > decode...