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...silence instead of no packets. Is this possible? 2) use better timestamps in streamed packets, ie increase timestamps even after a period of silence, and not only for each sent packet. Not sure if that makes the phone happy, though... Any chance to do one of those? Thanks in advance, Siggi

...ruct sockaddr_in bindaddr' defined but not used make: *** [ast_h323.o] Error 1 This is both with openh323-1.12.0 and their current CVS. (using current CVS snapshot of asterisk, too) Is that driver not supposed to work with current OpenH323?? Anything I'm doing wrong? Thanks in advance, Siggi

...m, and Commedian Mail seems to support multilingual prompts fine, it's just that it needs the prompts to be recorded by somebody, preferrably by a sexy female voice, similar to the English prompts. So: Has anybody translated the prompts to German, maybe even recorded them? Thanks in advance, Siggi

...g an * based VM solution here, as people particularly disliked Unity. It's still in it's early stages, but there will be the option of navigation through voicemail via the Cisco 79[64]0 XML display. So myadvice (you already guessed) is: drop Unity. Asterisk can do that better ;-) Cheers, Siggi

This Polycom phone seems to be one of the best on the market for sound quality and features. I have seen on the list that some people have gotten the IP 600 to work with Asterisk. Does anyone have the details of how to get this working i.e. XML phone config files, and any thing else I might need to know. Thank You, Chad Cowan -------------- next part -------------- An HTML attachment

---------- Forwarded message ---------- Date: Wed, 11 Jun 2003 01:10:16 +0200 (CEST) From: Siggi Langauf <langausd@fachschaft.informatik.uni-stuttgart.de> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] chan_h323 + openh323 CVS = no go? On Tue, 10 Jun 2003, Jeremy McNamara wrote: > If you would have followed the build instructions laid out by the Open > H.323 fo...

...result on August 8 at 10:06 from 0490233081 was:- > > File callerid.c, Line 278 (callerid_feed): Got this:- 080810060490233081 > > OK, Now what do I do? Well, I'd say: just strip the date and time off!. You made that call on 2003-08-08, 10:06h local time, didn't you? Cheers, Siggi

Hi Siggi/Jan, >If so, there's still a load version conflict (although I've >never seen a >7960 or 7940 care about the version communicated through SCCP): > >On the phone, press "Settings", then 4 for load information. >watch out for the "App-Load-ID". On my 79...

Hi Siggi, > > 7960 and then "Call Ended" on the Display (curious about that !!!). > > That seems to be normal for the 7920. I've sniffed the registration > procedure with Cisco's newest 3.3(3) CallManager (+patches), and it's > doing the same thing. Maybe that's...

I'm trying to set-up Asterisk server and I would like to buy 2 SIP phones. Has anybody tried Cisco 7960G? Or 7940? What audio compressions can I use with this phone and Asterisk? Reason why I'm asking is because Cisco supports G.711 and G.729a audio compression (probobaly some tohers but they are not listed on data sheet) and on Asterisk features i found that it supports G.729 but need

Hi, I wrote a little patch for the stable version (also works for the svn version) to add a new configuration parameter called "client_maxtime". With this you can set a maximum connection time limit for a connected client so that you can disallow continuous listening. When the listening time exceeds the client connection will automatically be dropped. By default this feature is

Hi, Sorry, I accidentally rejected this message instead of approving it. Geoff. ---------- Forwarded message ---------- Date: Fri, 11 Mar 2005 17:33:06 +0100 From: Siegfried Wagner <esiggi@gmail.com> Reply-To: siegfried@esiggi.net To: icecast-dev@xiph.org Subject: Re: [Icecast-dev] patch for icecast-2.2.0 to add client maxtime Hi, I did your recommended changes on my patch. But with the "signed/unsigned" mismatch: That's also wrong with other variables like source_...

<DIV><FONT face=Arial size=2>hello All,</FONT></DIV> <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV> <DIV><FONT face=Arial size=2>our Asterisk pbx is sitting behind a normal analog hardware pbx, we have to dial 9</FONT></DIV> <DIV><FONT face=Arial size=2>to take an outside call through the hardware pbx, our

On Mon, 17 Jan 2005 10:25:32 +0100, Siegfried Wagner <siggi@radiofreaks.org> wrote: > Hi, > > I wrote a little patch for the stable version (also works for the svn > version) to add a new configuration parameter called "client_maxtime". > With this you can set a maximum connection time limit for a connected > client so that y...

Hi, Is it possible to use 2 B channels simultaneously with either I4L or CAPI drivers? We use AVM A1 (Fritz) PCMCIA with I4L driver and AVM B1 PCMCIA with CAPI driver. Thanks, Michael.

...-- Playing 'vm-changeto' again, it only plays a "tick" and hangs till I press the next button. It doesn't matter how the call comes in (I've tried OH323, IAX and IAX2). Any idea what causes this? Am I the only one who gets this behaviour? Any fix? Thanks in advance, Siggi

Hi All, I've been developing all sorts of applications for use on our 79xx handsets but am having great difficulty with formatting, I just can't seem to be able to produce a line feed between lines on the stuff actually displayed on the phone. Has anyone else has experience or success with this ? Cheers, Adam ********* DISCLAIMER ********* This message and any attachment are

Can anyone point me to some online documentation showing how to reset a CP-7960 to factory default settings. I have some that are configured for Callmanager and I want to get them back to generic default config. Any info is appreciated. Thanks Cory Andrews

Does asterisk work with CCM as gateway ? When I trying call asterisk,I totally can't hear any sound. When call ohphone - works good. 10.0.1.219 is CCM, 10.0.1.207 asterisk. Trace messages here : -------------------- == New H.323 Connection created. -- Received SETUP message... == Setting up Call -- Calling party name: [5001,] -- Calling party number: [5001] -- Called party

I have three cisco 7910 phones connected to * through skinny protocol. When one of the phones is called, and the phone is ringing, you can hear what's going on in the room even though the caller hasn't answered. It's crazy and very hard to ignore when someone is calling :) God forbid you should cough while the phone is ringing. C.