similar to: nat=yes

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.

Hi List; I need to use an prepaid billing system with Asterisk, and I do not know which one is more stable and integrated with Asterisk? A2Billing or AstBill or ASTCC? Also, from where I can download it and ready about its configuration? Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 009659849460

hi I have some questions about re-encoding existing FLAC files to FLAC 1.2.0.: - can older 1.1.x FLAC files be re-encoded to FLAC 1.2.0 by using the FLAC 1.2.0 encoder? - can FLAC files encoded with the FLAC Flake SVN encoder (or any other 'unofficial' FLAC encoder) be re-encoded by using the FLAC 1.2.0 encoder? thx in advance! -------------- next part -------------- An HTML attachment

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,

Hello, I'm working with our SIP provider to nail down some call quality issues we're having, and they've asked me to provide SIP debug log files from our asterisk server. Is there a way to make asterisk 1.4 output only SIP debugging to a specific log file? Or it is best just to use tcpdump? Thank you! -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY

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: 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:

hi i was wondering if it's possible to tell me what the theoretical best compression ratio the flac encoder can do, because i was wondering what the future of flac will bring us. So this question is probably best addressed to Josh: can we suspect much better compression ratio in the future or is the encoder already at his *almost* peak compression level. thx in advance!

Hi, I have a legitimate request for information about how I can contribute to the development of FLAC. I am not a developer, but would like to make a financial contribution. FLAC has immensely enhanced my enjoyment of music. Please let me know if this is an inappropriate forum for this request. Please respond. Thank you. Joe Palermo -------------- next part -------------- An HTML attachment

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 remember hearing about Linux players many years ago ... anybody know whether I could

hi i have some small flac files and i want to concatenate to 1 big flac file, but without using ogg flac. Is this possible and how do i do this? i also was wondering if there exists a way to let the flac encoder remove audio frames of silence in the beginning and end of a wav input file when encoding? thx

hi does the flac encoder officially supports wav64? i have a big collection of wav64 files (26 TB for each file and bigger) but didn't had the courage to encode them with flac. Cpu power is no problem (i have intel V8) but i just want to know if flac supports wav64 or not as input

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

Dear Sirs, we ... 1) buy minutes from other providers 2) sell minutes to out clients some calls terminate to our equipment, others - to h323 proxies. we want calls to be routed according to costs (a route is chosen from many by lowest cost). at the end of it, we'd like to bill our clients and see how much have we earned (money we receive from client on one side, money we pay to proxies on

Hi List; How can one Asterisk Box A to send a SIP call for another Asterisk Box B, and that call to be authorized based on the username and password, and not on the IP (as the IP address of the source is not known because it keep changing)? I think the trick in the Dial command, how to write it properly in a way that other Asterisk Box can recognize the sip username and password which are existed

Dear List; I am trying to create a link between Asterisk and My softswitch, the link to be SIP Trunk. I did the below configuration and I do not know if any one can help me and advise me to have better configuration to be sure that link is fine. But I do not know how to determine the SIP username to be sent for my softswitch as sometimes the softswitch need to check it. Also, does asterisk

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

Hello..I am not very proficient in Linux kernel stuffs although I know my basics. I have a question and all suggestions/solutions will be highly appreciated... I got to know that RHEL 5 has inbuilt Xen Support...So I tried to install a RHEL 5.0 Server on my P4 machine. On top of it I tried to install Xen specific RPMs and some other RPMs needed by Xen. The additinal RPMs added after base

Hi List; I noticed that if I disabled secret in the context by placing ( ; ) before it, then at the asterisk the log will be: -- Registered SIP 'bilal_sip' at 0.0.0.0 port 5060 expired The IP address of the endpoint was not captured!!! Why? If secret enabled, then some endpoints can not register (maybe due to compatibility in reading the negotiation packets), so what is the solution?