search for: overkamp

...ble). Is there something in the apps I've missed that allows this already ? - In voicemailmain2 there is no option in the menu that allows creating your own messages (in fact, option 3 is defunct). Is this in the coming, or am I missing more stuff ? Thanks! Met vriendelijke groet, Florian Overkamp ObSimRef BV (http://www.obsimref.com/)

...es a long time and the successrate is about 25% :-) My feeling is this should be better if we choose to slow it down more. But who can tell me what the best modem settings would be to try ? My HAYES dialect is rather old :-)) Any experiences or hints are appreciated. -- Best regards, Florian Overkamp

Hi, I came back from a quiet weekend today and found my E400P box to have gone astray. Asterisk is loaded from inittab, and started crashing and reloading a couple of thousands of times, each time notifying my monitoring service :-P I remember there would be issues on old cvs stuff since the crash at digium so I made a clean checkout just now. Here is what happens when I load manually:

toy buy my first cisco 186 but when i read this page http://www.djernes.org/~shawn/ata186.htm i cant find in my dev page some parameters just like " UseSIP " what i need to do to show this parameters Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030626/42e66005/attachment.htm

Hi what can be wrong with * that console does not show any stderr text printed from agi script? I am starting with asterisk -vvvvvvvvvvvvvvvvvvvvvvvvrc VERBOSE command does show text on console but printing of STDERR does not I tried it from Perl and from Java and in both cases almost the same result, except in Java more things do not work. In Java for, for example, SAY DIGITS 123 78# would

For anyone that cares... I am porting James Golovich's Manager.pm over to PHP. I plan on also doing some documentation which will cover both the Perl and PHP APIs, which will be almost identical (at least, to whatever extent is practical). Will let y'all know when I have some usable code to show you. -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content &

Good Day list I am looking to buy a few Wooksung Video phones to try with my asterisk box.... http://www.wooksung.com/eng/html/pro/pro_001.html has anyone had any experience using these with asterisk? Thanks Ron

Go to www.google.com type in your search query add this to the end of your search query: site:lists.digium.com e.g. http://www.google.com.au/search?hl=en&ie=utf-8&oe=utf-8&q=Australia+site:lists.digium.com The mailing list used to be on www.marko.net, I'm not sure if the whole archive was moved across, you might want to search with site:www.marko.net OR site:lists.digium.com

Hello, I have configured my TDM01B Card (1 FXO Port ) as follows (below) but it will not pick up an incoming call. Any suggestions/tools to see what the problem is? I have looked at zttool where this line changes but I don't understand what it means (The last digit changed from 0 to 1) Total/Conf/Act: 4/ 1/ 1 /etc/zaptel.conf fxsks=4 loadzone=nl defaultzone=nl

Hi, Does anyone know of a Print-to-Fax client that works with asterisk & spandsp? Astfax is a partial solution but that only lets us email the fax in, we'ld like to set it up so the user can hit the print button and send the fax (even if all it does is email - transparently to the user - the fax to astfax). -------------- next part -------------- An HTML attachment was scrubbed...

...9;s Topics: > > 1. Re: TDM01B card configuration (Dave Cotton) > 2. Re: [SPAM:***** SpamScore] RE: > [Asterisk-Users] Call Transfer > using SIP clients (Frank Schoep) > 3. RE: presence and IM again, want to develop a > working"hack" > (Florian Overkamp) > 4. calling shell scripts from within * (Terry > Wade) > 5. Re: Sometimes yes - sometimes no (dialplan) > (Ronald_Wiplinger) > 6. Dialogic D/300 E1 (Fredrik Lith?n) > 7. Transfer and CDR's (Sebastian Zaprzalski) > 8. RE: Provider Survey (Mohamed Farid) >...

Hi all, DIAX 0.9.4 is available for download from the same place: http://www.laser.com/dante or http://www.geocities.com/tdanro The new DLL contain the latest updates made by Steve in the iaxclient library. What's new in 0.9.4: - IAX2 support (new DLL); - selectable DSP: Echo cancellation, AGC, Denoise; - plaintext and md5 authentication supported; - the phonebook is now in a separate

Did somebody already configured a Digium card on the network of Versatel in Belgium or the netherlands, and would like to share his configuration. (zaptel.conf / zapata.conf) We have HDLC errors (timings i presume)

I have a SwissVoice IP10S but can not seem to get it to have dialtone or dial on *. Calls to or from 3001 don't work. Any ideas are appreciated. Robert mgcp.conf is: [general] port = 2427 bindaddr = 192.168.0.110 [ip10] host = 192.168.0.5 context = from-sip line => aaln/1 The portion of extensions.conf is: exten => 3001,1,Dial(MGCP/aaln1,20) exten => 3001,103,Hangup

Is it possible to rewrite caller id's? I would like to have sip phones appear by their local cid (like Henk <208>) but when they call out using the PRI I would like their full DID (MSN) to appear (like 0031201234567) I could ofcourse set callerid to the main phonenumber but surely there must be a better solution? Thanks!! Remco

Hello all. Since half december we are trying to implement * as our primary PBX. We had a test machine running with 2 sip phones and a single ISDN card. Since this was working ok, we installed * on our main Debian server. Some specs: P4 2.8 Ghz HyperThreading 512 MB RAM Debian Sid updated every week. Linux 2.6.9 vanilla kernel (not the debian package) 1 Intel Pro 100 nic for internal network 2

Dear all, I'd like to know if there is a way for multiple asterisk servers to share a common SIP and/or IAX registry. The setup I imagine would be something like : - several asterisk servers called sip1.isp.com, sip2.isp.com, ... - a DNS alias sip.isp.com pointing to all the addresses (thus providing a round robin resolution on each server) - each SIP client would register with sip.isp.com

I'm using DUNDi. My lookup returns 'IAX2' for the tech, and 'dundi:q9sgTFkVMBFdmp0IDX1bYQ@xxx.187.142.204/3254101' for the destination. How do I dial this? I've tried dialling it with: "Dial" "IAX2/dundi:q9sgTFkVMBFdmp0IDX1bYQ@xxx.187.142.204/3254101" passed from my AGI script, but the other endpoint (xxx.187.142.204) is returning: May 11 09:23:41

Hello All: Has anyone configured a meetme conference to use video? I have successfully used video phones to talk through *, but I cannot seem to get video when those phones dial into a meetme conference. Is there something else that I need to be doing other than set the "v" flag on my extension for the meetme app? Thanks, Tim

.... (Anton Tinchev) > 8. Wildcard X100P usable in Germany? (Roger Schreiter) > 9. RE: Need Europian vendor for Digium hardware. (Low, Adam) > 10. RE: Asterisk Indications (Philipp von Klitzing) > 11. He really doesn't care (Bill Michaelson) > 12. RE: GSM modems (Florian Overkamp) > 13. Re: GSM modems (Max Tulyev) > 14. Re: Has Nufone gone belly-up (Vic Cross) > > --__--__-- > > Message: 1 > From: "James H. Thompson" <jht@lj.net> > To: <asterisk-users@lists.digium.com> > Subject: Re: [Asterisk-Users] Incoming SIP matching...