search for: maron

...gateway for that purpose. Does anyone have an idea of which product from digium would be suitable as a hardware timer for our setup. We need to be able to have conferences with up to 20-30 people, and listen in conferences with 2 or 3 speakers that support up to 200 listeners. Best regards, Maron Kristofersson

If you wan't to create a ringtone with makering.pl for firmware 1.0.50, be sure to create it as ring.bin and then rename it to ring1.bin / ring2.bin or ring3.bin. This seems to be the only change between the format from 1.0.4.68. Regards, Maron

...but this has not been recommended since Debian 4. Alternatively, are there any unofficial repositories with packaged "blktap2"? Will "blktap2" return to Debian at some point, hence waiting would be the best solution? Or does anyone have any other solutions? TIA Jonathan Maron Text Control GmbH Firmensitz: Admiralstr. 54, D-28215 Bremen Registergericht: Amtsgericht Bremen HRB 25927 Geschäftsführer: Heiner Suhr _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

...n cfg.txt format but really vague. Has anybody successfully done this without using GAPS, or has anybody got a correctly formatted cfg.txt file that works (from GAPS). I would be happy to write a script or a java program that creates such a file, but I need the format to do that. Regards, Maron Kristofersson

Hello, I want to know if there is any way to avoid source() stopping when there is an error. Here is the content of my Main.R script: source("~/R/source/Constructor1.R") # Object1 should be constructed ifelse(exists("Object1"), # It's an S4 object print("Object1 exists"), # I can't avoid using 'validity'

Hi all, I'm running Asterisk since two days, and it's really one of the phatest software available on the net!!! Respect!!! I have connected Asterisk as a call manager for a cisco gatekeeper. Everything works fine internal, but if I want to ring to a PSTN over another call manager, which is connected over ISDN, I get the following output. Has anyone experience in this or can help me?

...office. I'm also wondering if there is any isdn based solution since there is a possibillity of another staff member going abroad, and then I would like to have 2 numbers, 1 for each user. Is the phone on the user end software-based, ip-telephone or an analog telephone? Best regards, Maron Krist?fersson Reykjav?k Iceland

...y grateful to get some comments, and even if somebody else has a similar setup with Asterisk and Cisco CME working together, or Global company with VOIP (we have 15 offices in 11 countries quite evenly distributed around the globe.) Hoping for a good, fair and interesting discussion. Regards, Maron

I have an analog Fax machine which I wish to connect to the network and the Asterisk server. It will connect through a GS Handytone 286 converter and then into the LAN. Is there any information out there on what I need to write in *sip.conf* and/or *extensions.conf* to make sure the fax works as a fax? Channel 8 on my T1 is a reserved, dedicated line for the fax number. Do I need to

...f the hotels). I would like to ask the list if you can suggest a good "Reception" IP phone of decent hardware quality, and if you know if the build quality of the Snom 360 is better than the Snom 200 as I'm very happy with the software and usability of the Snom's. Best regards, Maron Kristofersson

About every hour I see the yellow alarms on all or a number of channels of my PRI which is connected to the dutch telephony network, Asterisk keeps on working fine.... Here's an example where channel 1-24 went into alarm: WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event): Detected alarm on channel 1: Yellow Alarm WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event):

I'm planning on setting up some remote agents and before doing so, I did some simple PING tests to measure latency. The average latency I got was 250ms. Does anyone have experience in terms of quality of calls when there is such high latency? Can anyone comment? Thanks, Waldo

Hi everyone I'm interested in using the Telappliant/voip-talk offering as an alternative to my DDI analog problem. (see [Asterisk-Users] Multiple DDI & Hunting on Analog Lines (UK) for details) Does anyone on the list have any recent comments on reliability etc? I would really appricated some positive and negative comments. Cheers Matt

Like many others on this list, I had been experiencing periodic lockups with my Grandstream products (Handytone 286 ATA & BudgeTone 101). The lockups consisted of seemingly dead devices, no dialtone or response, until I power cycled via software or hardware. The workaround had been to reboot the device every 30 minutes with a cron job. I contacted Grandstream and although they didn't

Hi All, I decided to have a go at installing Asterisk on FC2 which now runs on Kernel 2.6.. Unfortunately I didn't get very far.. When trying to build zaptel it required me to link /usr/scr/linux-2.6 to the default source dir which is /usr/src/linux-2.6.5-1.358.. I guess thats still the RH infulence.. :) After than I tried again but the page rolls with errors and finally ends with..

Are they any issues still with hyperthreading processors, I've read and been told by a few people to make sure its disabled in bios if I want to use * on a hyperthreading machine. Kind Regards, Chris Bond

Hi, I need a Diameter protocol stack (DDA/Ro) for Asterisk. I came across Traffix Systems Diameter stack. according to the site they got Asterisk ready Diameter stack and also Diameter Gateway that can interface to Asterisk, has anyone tried those out ? thanks Maria