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2003 Jun 10
0
chan_h323 + openh323 CVS = no go? (fwd)
---------- 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 you wouldn't...
2003 Jul 24
1
Cisco's CallManager and * (was: Cisco 7960g) (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
2003 Nov 10
3
Asterisk and Polycom Soundpoint IP600
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
2003 Jul 23
5
Asterisk as a stand alone voice mail server
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
2003 Jul 30
4
SCO/Linux concerns
...biz> > Date: Wed, 30 Jul 2003 17:25:23 -0300 > Subject: [Asterisk-Users] %unsuscribe > Reply-To: asterisk-users@lists.digium.com > > > %unsuscribe > > > > --__--__-- > > Message: 13 > Date: Wed, 30 Jul 2003 23:31:58 +0200 (CEST) > From: Siggi Langauf <langausd@swt.uni-stuttgart.de> > To: Asterisk user list <asterisk-users@lists.digium.com> > Subject: Re: [Asterisk-Users] SetCIDName > Reply-To: asterisk-users@lists.digium.com > > On Wed, 30 Jul 2003, Jeremy McNamara wrote: > > > Because H.323 doesn't have a specific ...
2003 Apr 26
2
German voicemail prompts, anybody?
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
2003 Jul 26
0
app_voicemail2 became a bit silent, lately...
Hi, after cvs upgrading my * installation yesterday, the prompts in both VoiceMail2 and VoiceMailMain2 have become silent. All I get is the initial "blip" followed by the Voice taking breath and being cut off before she has a chance to say "Comedian Mail". All other prompts (ie the Playback application) seem to work fine. I can still login to VoiceMailMain2, however, each
2003 Aug 09
1
Does Wildcard x100p support Caller ID outside the US? (fwd)
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:-
2003 Aug 09
2
Asterisk as a stand alone voice mail server (fwd)
On Fri, 8 Aug 2003, Maik Schmitt wrote: [...] > I just tried to use it with our 7960 (sip-version). > > I've set the services_url in SIPDefault.cnf to > "http://xxx.xxx.xxx.xxx/xmlservices/vm/index.php&amp;user=1234&amp;pin=1234" > > It didn't work with ?user=...&pin=.... cause the phone then tried to > get
2003 Jun 11
2
filling suppressed silence with chan_oh323
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
2003 Jun 10
4
chan_h323 + openh323 CVS = no go?
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