similar to: German voicemail prompts, anybody?

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

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

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

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 All, Hope all is going well. Was just wondering if anyone has implemented a grep style output page of hits using Ferret as the index/query engine? Any thoughts about how best to implement it? The previous thread discussess highlighting - would that be the best approach to follow or is there a better way? Cheers, Marcus -- Posted via http://www.ruby-forum.com/.

On Fri, 8 Aug 2003, Dave Cotton wrote: > The x100p does get the CID in France. It is now a question of how to break it down. > > I changed callerid.c line 278 to :- > > ast_log(LOG_NOTICE, "Got this:- %s\n", cid->rawdata); > > and the result on August 8 at 10:06 from 0490233081 was:- > > File callerid.c, Line 278 (callerid_feed): Got this:-

<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

I have had two users on two separate systems indicate that they could "not hear a new message" When I investigate I find that the user has marked a message for deletion. The .WAV .wav and gsm files are gone but the .txt file remains thus giving asterisk and the user the impression that a new message exists when it does not. Has anyone else encountered this issue? Is there a fix?

There isn't much documentation on adsi, but I called NETXUSA (the vendor of my 480e) and they helped me along. My experience: 1. I really had no experience with ADSI so I had (probably still have) some misconceptions on how the configuration is loaded onto the phone. 2. I set the following in my /etc/asterisk/asterisk.adsi (most of this is the stock asterisk.adsi script): ;

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, 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

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I'm have a bit of a problem with temporary greetings. I'm using 1.6.0.1 with IMAP storage. If I go into comedian mail and record a temporary greeting, I get a Message [0] on my IMAP server. In addition, I get /var/spool/asterisk/voicemail/default/134/temp.[gsm|wav|WAV] files. I have imapgreetings=no in the voicemail configuration file, as

---------- 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

Hi, I''m currently trying to set up a solution involving multiple servers using the same index over nfs. The problem is that from what I have seen, ferret doesn''t support multiple processes writing to the same index. Using a DRb service is not an option since this would create a single point of failure. I tried using Ferret::Store::FSDirectory to create a write lock on the the

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... >

Folks, A few people have made me aware of some omissions in my files (not my fault, they weren't in the Script from the Wiki) which I shall be tackling this weekend. Whilst I'm making the files are there any other files you want? IVR's etc. If so make sure I have a script sent by email. -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ

i was looking but i dont find how do this: configure the password for the extensions read the messages and some other things related with this can some bady help me with some material or a explicit example. thanks in advance Rodney Acosta Coya.