search for: klitz

Hi ho, is there anyone out here that is making use of the regcontext and regexten settings in sip.conf? I've tried this on two Asterisk boxes (1.2.10 and 1.2.12.1) and in both cases I don't see the Noop priority 1 being created upon SIP client registration, "show dialplan xxx" reveals no change. And yes, I have also read and checked bug 7144; if I go down that route and no

Hi All, Is there a way to change the mappings of disconnect reasons to certain SIP messages? E.G. I need to change the mapping for SIP 402 ?Payment Required? from 16 (normal termination) like it is in 1.4.24 to 21 (call rejected) as defined in RFC 3398. For me this is a big issue because my dial plan will look for alternative termination in the event of network error (e.g. reason 3 or 21 which is

...already played with > > nat/qualify/canreinvite/directrtp/externip etc parameters. > > > > regards, > > > > Nasir Javaid > > > > > > ------------------------------ > > Message: 13 > Date: Tue, 03 Aug 2010 13:21:23 +0200 > From: Philipp von Klitzing <klitzing at pool.informatik.rwth-aachen.de> > Subject: Re: [asterisk-users] mapping of disconnect reasons > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: > <4C5817D3.976.37D9D61 at kli...

Hi Guys, i have recently connected my (working) asterisk 1.2 server, with two 1.4 asterisk servers (one using SIP the other using IAX), since then (i believe) people starts complaining about a high background noise when using the handset on Polycom phones (but when using the speaker it's fine, and i noticed that my self), my question is, can anybody tell me any step to begin diagnosing the

Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/XYZ at 192.168.0.20:5060 SIP/XYZ at 192.168.0.10:5678 i dial using following dial string Dial(SIP/XYZ at

Hi, looking for your valued input on suitable suggestions for high quality VoIP DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and looking to a new manufacturer. -- Thanks, Phil

Hi there, is gizmo the first user of the Digium Skype solution, or do they use a different approach/product - any clue? http://www.gizmo5.com/pc/opensky/ Philipp

Hi, I'm tying to delay the HungUp. I tried this way: exten => h,1,NoOp(Start) exten => h,n,Wait(5) exten => h,n,NoOp(End) exten => h,n,Hangup() but it doesn't work, Any idea? Thanks in advance.

Ok, here's the challenge: I would like to be able to find, match - and then react - upon prompts that are presented by the outbound/remote side of a call. Think mobile phone and "This user is temporarily unavailable". Collecting a limited number of known prompt snippets should not be a problem, but how would you then detect their presence in a longer recording (or live audio

Hi there, is this a problem with the Wiki software or the DB? The delay is still tolerable, but not exactly nice to work with. http://www.voip-info.org/tiki-index.php?page=Asterisk+billing Page generated in: 2.35 seconds Philipp

...e to be a possibility to configure multiple ISDN cards (e.g. AVM B1 PCI) through capi.conf. How? Or does chan_capi support only one ISDN-Card? Thanks for your reply, Stefan -------- Original Message -------- Subject: Re: [Asterisk-Users] Some questions for chan_capi (16-Okt-2003 18:18) From: klitzing@pool.informatik.rwth-aachen.de To: s.speckenheuer@posservice.de Hi! > - how to configure multiple isdn-cards in capi.conf? According to this link you can have only 1 passive card in your system: http://graphics.cs.uni-sb.de/VoIP/en/index.html Greetings, Philipp To: asterisk-us...

I didn't know if it would work or not, but I figured I'd try slow-speed half-duplex TDD over GSM & Vonage. I called a AGI script I have that speaks to TTYs, by calling from Vonage to one of my Voicepulse lines. I don't control the Vonage codec, so I have no idea what it uses, but I am using GSM for the Voicepulse line. Everything worked fine - echo canceling didn't cause any

...t; As per ATA, it is by default using rfc2833. I tried setting it up as inband > by setting Audiomode, but nothing helped. I was thinking the * is ONLY > recognizing the DTMF if there is telco board installed. Is it? > > > ----- Original Message ----- > From: "Philipp von Klitzing" <klitzing@pool.informatik.rwth-aachen.de> > To: "Jess Magnaye" <jess@arretni.com> > Sent: Tuesday, December 23, 2003 12:36 PM > Subject: Re: [Asterisk-Users] Fw: Questions and finding > > > > Hi! > > > > > 1.) First test > > &...

I just moved my system over to a new server with * 0.7.1. The old machine was using a cvs from August/Sep timeframe. On the new machine I did an make samples but then ovewrote with tar files of the production configs in the /etc/asterisk /var/spool/asterisk /var/lib/asterisk folders. Now the system seems to be working fine but only records blank audio in the voicemail files. Same thing with

Hi All, I'm tring to do some DB operations before and after a call. I see the 'g' option in dial to continue in context if the destination hangs up, but what if the originator hangs up? Basically I do a DB get/put before the call is placed. After the call is completed I want to do another get/put; however the dial application dies when the originator hangs up. Any way to get around

Hi, I seem to have a problem with chanisavail and the call limits on sip phones(incoming and outgoing) The problem seems to be that chanisavail when trying create to create channels and hanging them up afterwards screw up the current usage limit on the phones. Example with chanisavail: Phone A calls voicemail (usage now 1) Phone B tries to call Phone A and uses ChanIsAvail in the dialplan.

...f all of the current answering services actually have to click a button on a headset and on their screens? Does anyone currently use asterisk in an answering service? What kind of phone/headset/call connection system do you use? Thanks Michael Blood -----Original Message----- From: Philipp von Klitzing [mailto:klitzing@pool.informatik.rwth-aachen.de] Sent: Wednesday, June 30, 2004 4:58 PM To: Michael Blood, Matraex, Inc. Subject: Re: [Asterisk-Users] Answering Service Agent Auto Login Hi! > So... a phone with auto answer COULD work if we could find an > inexpensive enough one (less...

Funny - I downloaded the latest Asterisk CVS, and it's pretty much working. Will report when I have some more success. PaulH -----Original Message----- From: Philipp von Klitzing [mailto:klitzing@pool.informatik.rwth-aachen.de] Sent: Tuesday, 28 September 2004 9:46 PM To: Paul Hales Subject: Re: [Asterisk-Users] Leader IP10S Hi! > I have been lent a Leader IP10S phone (SIP) for evaluation. > No luck with making or receiving calls - the phone registers OK though....

Hi, For various reasons a customer of mine is moving from a SER-based to an Asterisk-based installation, mostly because of problems with SIP devices behind NAT trying to reach each other and because it's easier to do accounting when all calls go through Asterisk (canreinvite=no is the idea). The database-based SIP registration mechanism of Asterisk seems to have one shortcoming - it

...Because I really can not filter this from the wiki.<br> <br> Looking forward to your answer, thx !<br> <br> <br> Jonas.<br> <font face="Helvetica, Arial, sans-serif"><br> </font><br> On 06/15/2010 06:09 PM, Philipp von Klitzing wrote: <blockquote cite="mid:4C17C1F2.2275.146F2DA at klitzing.pool.informatik.rwth-aachen.de" type="cite"> <pre wrap="">Hi! </pre> <blockquote type="cite"> <pre wrap="">How to do this ?? To proc...