search for: rkle

...as of documentation, this sometimes emulates flash => not in my setup * Tried the above from snom sip phone, sip ata with analogue phone and flash-key, mobile phone called in via another zap channel => no difference between the incomings Has somebody any hints for me? Stefan -- Stefan M?rkle Netpioneer GmbH Leitender Systemarchitekt Beiertheimer Allee 18 <stefan.maerkle@netpioneer.de> 76137 Karlsruhe *** Besuchen Sie uns vom 09.03.- 15.03.2006 auf der Cebit 2006 in Hannover. Sie finden uns in Halle 3 auf Stand D31 als Mitaussteller der Imper...

...r vendors as Digium) - Risk management. Asterisk becomes SPOF for whole telephony system (at least for inbound/outbound calls) In our company we will probably setup your scenario 2 for integration of some voip-Users and voicemail functionality. Just my 2 cents, Stefan -- Stefan M?rkle Netpioneer GmbH Leiter Knowledge Center Beiertheimer Allee 18 <stefan.maerkle@netpioneer.de> 76137 Karlsruhe

...to only fight on the 'bleeding edge', especially when not being paid for their work. So this is something only Mark or Digium could achieve or a person/team that has full support by them. Just my 2 Euro-Cents, and sorry for the long text, hope somebody made it to the end ... ;-) Stefan M?rkle -- Stefan M?rkle Netpioneer GmbH Leitender Systemarchitekt Beiertheimer Allee 18 <stefan.maerkle@netpioneer.de> 76137 Karlsruhe

...nd receive audio with sco, impementing the other end of the bluetooth headset profile. http://www.dcs.gla.ac.uk/~jp/snd-bt-sco/ But codec and signalling looks quite symetric on a first glance, so the actual handling of the channel's audio data should be similar to the above project. Stefan M?rkle

I've got spanDSP and Asterisk Stable using an X100p running on Fedora Core 3 but there is a curious phenomenon when I receive faxes. If I start Asterisk from the command line (usually, asterisk -c or asterisk -vvvvvvvvvvvvvvvcp ) I can receive faxes and mailtofax sends them to me OK. If I start the asterisk service (service asterisk start) that uses safe_asterisk, faxes appear to be

using "synchronized" subsecond timestamps imho only makes sense with synchronized time (ntp) through that ssh tunnel, too. and with this assumption a "full VPN ssh usage" instead of "only filesystem timestamps" [maybe trying with target systems without subsecond timestamps?] seems impractical to me. or at least "... [sry, didnt have internet to send, incomplete

For all the peoples that wanted to test my windows IAX2 phone, I've put it up on a server where it can be downloaded. For the ones that wanted the DLL, it's available on the same page. For the DLL, I will post a list of the functions in it and the parameters it expect as soon as I have some free time. All comments (good or bad) are welcome The phone can be used mostly with the keyboard

Hi all, I have just been reading an article on the asterisk-doc site about ISDN X-Over cables. The article mentioned the converting of an NT1 to make this possible, has anybody got the information required to modify a BT NT1? Or any information on the BT NT1. Thanks in advance. Regards Dave

Hi, I'm trying to start with Asterisk, but I could not put 2 softphones to talk. The asterisk server rejects the connections always when I dial. May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from 192.168.0.106 What is necessary to put it to work? There is no need to configure external lines. extensions.conf [internal1] exten => 311000,1,Dial(SIP/teste1) [internal2] exten

I've read quite a bit in the older mailing list posts and the wiki but I'm missing some simple point. 1) What is required to send an SMS to a mobile outside the office given: Channel: ZAP/1 send it to $SMS_RECIPIENT (which includes the final "extra" digit) via $SMS_CENTER=the national message center server for sending messages $MESSAGE= the message text How is the .call file

hi list. after my unsuccessfull experiences with mISDN i tried again to implement a zaphfc based solution. problem is: sound on calls via capi is stuttering/broken and therefore unuseable. my conf: - cel 1300, 256mb ram - avm b1 via capi connected to my outgoing ISDN - acer surf pci via zaphafc and crosslink cable with termination as internal bus - siemens gigaset 3035 as internal phone -