search for: zagler

...unmap(0xbf334000, 4096) = 0 exit_group(1) = ? -------------------------------------------------------- System description: Fedora Core 1 Kernel 2.4.22 Sudo 1.6.7p5 Apache httpd 2.0.50 Asterisk 1.0-RC2 Can anyone please help? Thank you in advance! Roland Zagler mailto:r.zagler@fog.at @fog smart partners

Hi, I am having trouble configuring CAPI so that call transfers work. I make a SIP call to asterisk which goes out on ISDn via CAPI. Then I try to do a transfer from the SIP phone which doesn't work and results in the call being disconnected. The error message given by asterisk is that it chan_capi can't find an entry for the outgoing msn for the transfer however the outgoing msn is the

Hello! Is there a way to use AVM Fritz!PCI as a ZAP interface and have it configured for ZAP channels? Thanx in advance! Roland Zagler mailto:r.zagler@fog.at @fog smart partners

Thanks for the suggestion, C F, but the problem is there is a rather big database application behind with many users, so a static configuration is not suitable for my needs. i am working mostly with realtime and agi. regards, roland -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F Sent: Sunday, July 03,

Hello! Is it possible to run Asterisk as a SMS Service Center and does it work over a directly connected ISDN (CAPI) interface card? Did anyone already use Asterisk for that? Roland Zagler mailto:r.zagler@fog.at @fog smart partners

Hello everyone, I want to build an Asterisk Box where i need 8 FXS interfaces to connect 8 phones to. The problem is, that there is only one PCI slot available. What i have is 4 USBs 2.0 interfaces free (if this helps). So here's my question: how am i going to do this? i tried to find any PCI cards supporting 8 FXS interfaces, but without success. does anyone know such hardware? Thanks in

Hello list, does anyone know how to change the "interdigit timeout" when using Cisco IP Phone 7940/7960 with SIP-Firmware and Asterisk? it's default value is 15 sec. but i have nothing found to set this in tftp-config file etc. Thanks in advance, Roland

Please can someone send me the .tar.gz file of spandsp, the site is offline and i didn't find it anywhere! Thanxxxx! Roland Zagler mailto:r.zagler@fog.at @fog smart partners

...Digium E1/T1 cards like TE410P. does anyone know about any manufactorer of those cards or someone who is able to develop/build such cards? Specifications: PCI or MiniPCI up to 120 concurrent transcodings Codecs: G.729/G.729A or G.723.1 or GSM or combinations of them Thank you in advance, Roland Zagler

...and waits for a user input, but the caller should be connected to SIP Phone 100 after it has answered and the announcement has been played. Before connecting to SIP Phone 100 the caller should hear a soundfile... wiki says nothing about an Dial-option to play a soundfile to the caller ;-( Roland Zagler -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Robert Goodyear Sent: Saturday, July 02, 2005 8:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] play message to ca...

.../21/2005). just for your info, Peter, my dmesg command shows: "TE410P version c01a009b" Hope this helps you, folks! Best regards, Roland -----Original Message----- Sent: Wednesday, July 20, 2005 10:26 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: Roland Zagler Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server What is your dmesg output when you fire up the card. There were some problems with TE410P and the intel chipset used in the DL380 G4's. You need firmware at least 'TE410P version c01a010b' Contact Digium...

...Or if you are feeling more adventurous you could load the Manager Bridge patch that I posted to the bugtracker two months ago. It allows bridging of any two existing channels together through a manager action: http://bugs.digium.com/view.php?id=4297 MATT--- -----Original Message----- From: Roland Zagler [mailto:r.zagler@fog.at] Sent: Saturday, July 02, 2005 4:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] play message to callee before connect toincomingcall sorry for the misunderstanding, robert! the point is: during the caller is listening to th...

...999,2,playback(~.mp3) exten => 999,3,dial (sip/100) exten => 999,4,playbackground(~.mp3) exten => 999,h,Hangup() not sure abt playbackground should be before the dial command or after ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Roland Zagler Sent: Sat 7/2/2005 8:23 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] play message to callee before connect to incomingcall Hello, i try to do the following: 1) call comes in on extension 999 2) caller should hear music (NOT MoH!!!) 3) a call should be initiated to SIP Phone...

...om [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Robert Goodyear Sent: Saturday, July 02, 2005 9:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] play message to callee before connect toincomingcall On Jul 2, 2005, at 12:19 PM, Roland Zagler wrote: > Thank you, Robert! > > The announcementfile plays well, but at Dial-option "m" i have to > specify a MoH class, > that is something i cannot use (as i wrote in my post). Noted, which is why I offered option two. > > Background command waits for a user inpu...

Hello, I use Asterisk 1.0.2 on a RedHat Enterprise Server 3.0 (Kernel 2.4.21) and i experienced that the memory consumption of the asterisk-process started by the init.d-script raises continously. Now, after 3 hours of operation (on our testing-system we have 30 concurrent connections to another asterisk box using IAX2 and GSM codec) there is already 66MB allocated. I think this could be ok, but

Scenario: Both boxes are behind firewall, port udp 4569 is open. If I don't want the username and password in dialing string do I have to use register statement in IAX.CONF. Can anybody post some working samples; I have a hard time making it to work with the samples posted on wiki. -- #Joseph

Hello list, is there anyone out there that could grab the new SIP firmware 7.5 for the 7940/7960 from Cisco's Site and mail it to me (r.zagler@fog.at)? i already ordered a support contract but did not get my access data yet! Thanks, Roland

...um.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Robert Goodyear Sent: Sunday, July 03, 2005 9:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] play message to callee before connecttoincomingcall On Jul 2, 2005, at 1:00 PM, Roland Zagler wrote: > sorry for the misunderstanding, robert! > > the point is: during the caller is listening to the soundfile played to > him > the dialplan should continue to dial the sip phone 100 and after sip > phone > 100 is answered and the announcement file is played the caller sh...

Hi, after you have done "make", "make install" and maybe "make samples" in asterisk source-dir just do a "make config" and all will be done for you. to check if it worked, simply issue "chkconfig --list asterisk" to see the runlevels asterisk is started or not. to start zaptel drivers do the same after "make install" ("make

...D: 6dccb2ab1469a1c87fff22d2076fd449@[myIP]..Contact: user@62.99.1 90.238..Content-type: application/sdp..Max-Forwards: 70..Content-Length: 133....v=0..o=none 0 0 IN IP4 [peerIP]..s=-..c=IN IP4 198.31.231.1 7..t=0 0..m=audio 18691 RTP/AVP 8..a=rtpmap:8 PCMA/8000..a=ptime:30.. Thanxxxx Roland Zagler mailto:laureen@laureen.at @fog smart partners