similar to: Does Wildcard x100p support Caller ID outside the US? (fwd)

On Wed, 23 Jul 2003, Yifang Dai wrote: > I wish! My company just spend a lot $$ on the shinny CCM phone system, so I > don't think I can change that easily... But if I can get asterisk to > talk to CCM via h323, and prove it's usefulness, I might have a chance > to use * in the branches... Well, good luck, then! > By the way, do you know if we can get *'s VM to

After some more analysis of my "dropped fragment" problem, things look like this: Cisco 7940 phone -- RTP --> chan_oh323 --> Asterisk (running, eg., VoiceMailMain) That RTP connection was negotiated via H.323 on a third machine running Cisco CallManager 3.2, but this part should not be relevant. Connections work fine, with one

Hi, trying to build the h323 channel driver that comes with asterisk works fine, but only as long as I use openh323-1.11.7. Unfortunately, that setup seems to have a bug which misguides one of the audio streams. (So while * can "hear" me, the phone remains silent.) I suppose that bug is fixed at least in openh323 CVS. At least, I got things mostly working using the external

Hi all, I'm trying to build a little voicemail server based on asterisk here, using Asterisk's "Commedian Mail" application. Unfortunately, I'd expect some people to have trouble using the English prompts that come with asterisk. However, I can't imagine I'm the first person who has this problem, and Commedian Mail seems to support multilingual prompts fine, it's

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 folks

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

<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

Hello Since I am getting a bit concerned about the SCO vs IBM issue, I was wondering if can I can setup Asterisk on FreeBSD is it supported ? Are drivers for Digium cards available on FreeBSD ? Thanks Ajit ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 3:05 PM Subject: Asterisk-Users

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 7940, this is >"P00305000300". Yours >is most likely a smaller number... >

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 some odd way of testing if the > CallManager ("CCM") really works... >

I see from the archives that Siggi Langauf was wanting to do exactly what I want to do back in November 2003. Here is what he asked: I would like to do a pilot with some legacy gear, however. Accordingly, I'd like to be able to have * dial 1000X where X is the box that has a new voicemail message and 1001X when the user of mb X deletes the new message(s). The dialing should occur

I'm just curious if anyone has the X100P & Callerid receiving working outside US. Replies are appreciated. Also if it's not working for you in a certain coutry you can respond too. regards Martin

I'm sure asterisk would make a great stand alone voice mail server. Basically I want to get rid of our voice mail system and replace it with *, but the problem is we use a cisco cluster with skinny clients. So I was thinking the way to contact a * server, would be through our 3640. But so far any attempt has failed. I am wondering if anyone has done something similar. Just want to verify the

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

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

[root@www root]# cat /proc/zaptel/1 Span 1: WCFXO/0 "Wildcard X101P Board 1" 1 WCFXO/0/0 FXSKS (In use) Asterisk CVS-HEAD-02/13/05-00:32:03, Copyright (C) 1999 - 2005 Digium. Feb 15 22:33:48 NOTICE[3002]: callerid.c:307 callerid_feed: Caller*ID failed checksum Feb 15 22:33:51 ERROR[3002]: callerid.c:261 callerid_feed: fsk_serie made mylen < 0 (-6) Feb 15 22:33:51

Hi, I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm having problems with Caller ID. I have run clidtest, and it seems happy enough, returning:- server clidtest # ./clidtest /dev/zap/1 Number: 0412222222, Name: MOBILE (that number's fake.) However, I'm not getting the caller ID passed through with *. Sometimes I get a "failed checksum" like

Hello, I recently purchased a Wildcard X100P FXO Card primarily as a learning tool and to provide myself with voicemail and a few other perks that are better than an answering machine. I followed the setup in the FAQ, and it works, just not well. The sound is very choppy and staticy whenever a call is placed. Sometimes it clears up, and sometimes it degrades very badly. I also recently

Hi all, The Wildcard X100P card or the Asterisk Developer's Kit (LITE) are available somewhere in Europe or must be ordered in US? Thanks, Dan