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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

---------- 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...

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

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:-

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 d...

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

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

<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

...ptsn (Brian West) > 8. Re: sip -> h323 -> ptsn (Patrick) > 9. X100P and incoming Context + CDR? (Darren Smith) > 10. Re: CVS Problem? (Kyle Hagan) > 11. Re: sip -> h323 -> ptsn (Eric Wieling) > 12. %unsuscribe (Carlos Crembil) > 13. Re: SetCIDName (Siggi Langauf) > 14. RE: X-Lite and Call transfer using Asterisk (Stuart Hirst) > > --__--__-- > > Message: 1 > From: "Todd Lieberman" <todd@tlsolutions.net> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] voicemail file access problems >...

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

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