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...ed. Pretty basic RH6.2 with qt-2.2.3, KDE2.1beta2 and Helixgnome, 128 MB RAM. I notice there is itcl-8.x.rpm available for Mandrake, but all the RH itcls I have found have been in the 3.x series. Is this the prob? What to do? PS: This message may have been posted twice; if so, my apologies (cwnet.com's news transport failed during the attempt to send the first, so I resent. <blush>)

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...an extension. I've tried many solutions, played around with all dtmf options...but no luck... Note that after the first digit (which get 'lost'), others get processed successfully. Cheers, Simone. -- ___________________________________________ Simone Ricci E-mail: simone.ricci@cwnet.it CWNET s.r.l. http://www.cheapnet.it ___________________________________________

Anyone knows if there's a way to select the bitrate of those codecs supporting multiple bitrates (eg. g.726)? I've tried searching and googling a lot, but without useful results... Cheers, Simone.

Hi, I've a problem with a cisco 827-4v and asterisk (1.0.9) acting as sip-to-pstn GW. The issue is that when a call comes in from the pstn, asterisk correctly contacts the router, which in turns send a "183 Session progress". Obviously, asterisk thinks that the telephone is not ringing (because it expects a "180 Ringing") and we have no ringback on the pstn side. Putting a

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I've got an old HFC-S card to play with, and I would like to use it in NT mode. I've a problem only: wiring. I can't fully understand the instructions I was able to find online. Someone can point me to a site which explains the whole procedure clearly (like with some schematics, even in ASCII)? TIA, Simone.

Hi, I recently purchased a brand new 7905g with his SIP firmware (licensed). Now, I want to play a little with chan-sccp, but I'm unable to find the appropriate firmware for my phone. I know that I must get it directly from cisco, but before purchasing it will be very good for me to try a bit; does someone can provide me (or even tell me where can I find) the SCCP image (version doesn't

I've got a problem with DTMF, again. My asterisk box is connected with the outside world (PSTN) via a sip proxy. The problem is that for some reason, I need to use rfc2833 for signaling digits to the gateway and inband to accept digits from outside (eg. when someone dials one of our DIDs). It's possible to do this? I've ever tried splitting 'peer' and 'user' part in

Anyone knows where can I find spandsp? Official site seems permanently down... TIA, Simone.