similar to: outgoing call restriction

Hi all I have the raw out put of a video camera in following format: mjpegb, yuvj422p, 1920x1080, 24.00 fps(r). I cannot directly convert this raw footage to Theora due to the YUV4MPEG2 only restriction of the encoder_example. Isn't it better to support other raw formats in encoder_example, such as what I have, mjpegb, yuvj422p? I presume even if I convert the mjpegb, yuvj422p to

I'm currently implementing A rtp header info and using speex for audio. can someone tell me what the rtp payload type value is for speex. i've checked the submitted speex draft and it says its beyond its scope and i can't find anything in RFC3550 that might suggest what it is. any help would be appreciated thank you, Greg --------------------------------- Boardwalk for $500?

Hi List; If I set nat=yes, then asterisk will send the packets to the public IP address or to the private IP address (which will be for the endpoint that is behind the nating)? And by setting the nat=yes, then what exactly will be ignored at asterisk side when reading the registeration messages from the endpoint? Any help. Regards Bilal

Hi: Please every that work with A102d say how about is it?Is it really difficult to install card for me new in asterisk? Best regards. --------------------------------- Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. -------------- next part -------------- An HTML attachment was scrubbed... URL:

hmm...forgive my ignorance here. icould have explained it wrong. the rtp header has the pt (payload) field as a 7 bit value. i was under the impression speex had a particular value i should set it to. is this so? if no what value should i assign it, whether by convention or otherwise? Note that i'm implementing a simple rtp header and combining it with the speex payload i'm not using

Hi List; I need to configure a softphone to be client and use it with Asterisk, which is the recommended one? Is it iax2? Regards Bilal ____________________________________________________________________________________ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.

plz....read out the attachment....my prob is attached... --------------------------------- Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it''s updated for today''s economy) at Yahoo! Games. _______________________________________________ LARTC mailing list LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/cgi-bin/mailman/listinfo/lartc

Hi list; ASTCC supports IVR or there is a separate module for IVR? Can someone advise me a link to start download and ready about ASTCC to do the configuration? Regards, --------- ITS IP Telephony and Contact Center Engineer Bilal Ghayad Mobile: 00865 9849460 ____________________________________________________________________________________ Boardwalk for $500? In 2007? Ha! Play

Dear all I want to configure 2 port E1 card on my asterisk so which version is best 1.2.x or 1.4.x can anyone suggest me which one is best right now for asterisk and anyone have configuration file to configure E1 card and zaptel.conf so i can configure it --------------------------------- Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's

--- Brian Willoughby <brianw@sounds.wa.com> wrote: > are there any car stereo flac players? > > My CD transports are all broken or stolen, and I tend to ride with my > PowerBook attached digitally to my car stereo anyway. Seems like if > I buy anything to replace what I had, it should be a hard drive > player that supports flac, maybe with a removable drive. I

I am practically a newbie in all of this but i have to set up an online radio station for a client in two weeks. I am running on a linux based web hosting on a virtual server. i have read through the icecast documentation but do not know how to execute the unix commands and other stuffs like building and compiling those extra packages. I could really use some help. Also, I cant access the irc

I am practically a newbie in all of this but i have to set up an online radio station for a client in two weeks. I am running on a linux based web hosting on a virtual server. i have read through the icecast documentation but do not know how to execute the unix commands and other stuffs like building and compiling those extra packages. I could really use some help. Also, I cant access the irc

Hi! I'm using Slackware 11 and Wine 0.9.40. The problem is, Wine is not detecting my network settings, though it(Internet) works fine in my Slack. i can access the entire network (Company's), see the sambha shares in my slack, but when i try to run "ipconfig" or "ping", i got an error saying it was trying to user ICMP and this would not work unless used as root. Even

Maybe this has already been stated or is a feature request or I could be out of my mind. Is there anyway to access a remote mysql database from within let''s say...a puppet function? BC ____________________________________________________________________________________ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it''s updated for today''s

Hi, I'm a novice trying out an experiment with Asterisk and was unsure of the hardware needs for it. I'm wondering if its possible to receive a call from an external number (PSTN) say A. Then make a call to another external number (PSTN) say B - and then bridge the two calls so that A is talking to B? What hardware will I need to be able to do this. Secondly, if I had x number of

Harry, I wrote a little command line utility for this in C# - the lib it uses is a little overkill for the task but it does work. As Josh noted there is no compression gain, so I wouldn't bother unless you have FLAC's encoded with older versions. I also think Josh said there's a .BAT file out there that does the same thing. Here's the file:

Dear all i have one confusion about how to dial outgoing call through asterisk like when i press 0 i got dial ton of exchange for outgoing call my setup is [sip_phone]-----[*]----[mediant2k]-----[Avaya_PBX]------e1-----[Exchange_PSTN] now i want to setup whn i press 0 in my sip phone i got dialton of PSTN so i can call outside people is there any special configuration to give

I need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal ---------------- No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone,

Dear all I have beginer in Voip and i have configured Asterisk server with 100 IP SIP phone ( SNOM ) everything is fine but problem is how to transfer call from one user to other means i call to some one and then someone want to transfer call to another person how it is possible i have also try with feartue.conf but it is now working i have also read document on voip-info website

Hi In our office environment few servers mostly database servers and yesterday it happened for one application server(first time) the partion is getting "read only". I was checking the archives, found may be similar kind of issues in the 2007-July archives. But how it has been solved if someone describes me that will be really helpful. In our case, just at the problem started found