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...ore flexible. You may not care about cheap and flexible, and if not, maybe it's not what you want. I've not tested products like CallWeaver or others. People claim some of these are more reliable, but Asterisk seems more popular. Norman Franke Answering Service for Directors, Inc. www.myasd.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080319/95288680/attachment.htm

I've been evaluating Asterisk for a while, and things seem to be going very well. The issue of redundancy and automatic fail-over is now on my mind. I searched the archives and googled for solutions, but didn't really come up with much. We'll be using queues (modified), which precludes some of the standard redundancy solutions, since the queue needs to know all the agents

I'm trying to understand something that just doesn't seem to compute. How can companies like Cisco justify selling their hard phones for as much as they do? I know there is a matter of recouping R&D costs but when you look at the iPhone with all its amazing features for less than $500.00 it just doesn't make sense. Am I the only one that thinks this? Roy Anciso Director of

What does it take to get ztdummy to work correctly? I have a new laptop HP HDX9200. I am running asterisk 1.4.19 and zaptel 1.4.9.2 Zaptel compiles fine. asterisk compiles fine. ztdummy loads asterisk runs. Problem is playback() does not work. So then I stop zaptel, asterisk runs and playback() now works. However, meetme()'s dont work. I need ztdummy I'm pretty sure for that. I am

...ry a more automated provisioning method on it. I know that getting the polycom's to auto provision wasn't very straight forward. I do provision some the linksys PAP2Ts via HTTP and that works quite well, so I suspect the SPA's to be relatively similar. Norman Franke ASD, Inc. www.myasd.com On Sep 24, 2007, at 7:06 AM, asterisk-users-request at lists.digium.com wrote: > Linksys are great phones. I like them but there only problem is > limited line > appearances. I prefer Aastra over them because Aastra has more lines > appearances. They both are good. If you are...

Hi * users, I would like to extract the information present in the SIP REFER message that comes to asterisk. Would SIP_HEADER() allow me to do that ? I have used SIP_HEADER() for extracting the to and from SIP headers previously. Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998

...applications (using Apache Tapestry) to slice and dice our data for reporting (mostly graphically) that we use a lot. Since it's all quite specific to how we work and our custom solutions, it wouldn't help anyone, I'm sure. Norman Franke Answering Service for Directors, Inc. www.myasd.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080306/c0e53e04/attachment.htm

...mf=yes" in sip.conf, as well. I added "dtmfmode=inband" to the SIP peer for the Cisco. Nothing. Tones are generated by phones connected to our PBX that we don't have a problem with otherwise or even from a cell phone. Norman Franke Answering Service for Directors, Inc. www.myasd.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080319/09e9e181/attachment.htm

...ealing with the high-performance timer. I enabled: HPET Timer Support Enhanced Real Time Clock Support HPET - High Precision Event Timer HPET Control RTC IRQ Allow mmap of HPET I'm not sure if you can eliminate some of those, but this works for me and is stable. Norman Franke ASD, Inc. www.myasd.com On Mar 30, 2008, at 1:00 PM, asterisk-users-request at lists.digium.com wrote: > This is a reasonably common problem. ztdummy uses the Linux kernel > Real > Time Clock (RTC) and something is wrong with it. The solution is to > recompile your kernel, you should search the m...

...t issue than your own. I've tried 2.6.8, 2.6.18-5, 2.6.19, 2.6.21.3 and perhaps more. These are the only ones I recall. I tuned in late and didn't see they wanted Xen support, but I figure others may find it helpful via google. Norman Franke Answering Service for Directors, Inc. www.myasd.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080331/02110b02/attachment.htm

...lient, everything works fine (instead something via the Cisco.) If I set "canreinvite=no" for the Cisco everything works. It seems like the "g" option should disable canreinvite for that call, so why the difference? Norman Franke Answering Service for Directors, Inc. www.myasd.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080502/e9f5b59e/attachment.htm

...ype fxs-loop-start ds0-group 9 timeslots 10 type fxs-loop-start ds0-group 10 timeslots 11 type fxs-loop-start ds0-group 11 timeslots 12 type fxs-loop-start The cisco then sends calls to Asterisk, and that part works great from a PRI. Norman Franke Answering Service for Directors, Inc. www.myasd.com

...Try just a simple application that writes the input to a file, e.g. /tmp/output and see if that works. Maybe the asterisk process can't open your parallel port? I'm using a USB-based device for digital IO and that works great. Norman Franke Answering Service for Directors, Inc. www.myasd.com

Is anyone aware of a service where we can lookup phone numbers to determine a name and/or name + address available in bulk? We want to look up every number called to our call center, so it will be tens of thousands per day. Services that charge 3 to 5 cents per lookup will get way too expensive very quickly. Thus, I'm looking for a service that can either license a database or

Hello All, We have been experiencing some ongoing reliability problems with Asterisk for quite some time, and I am trying to find out if anyone else has experienced the same problems. We are running asterisk 1.4.17~dfsg-2+b1 on Debian Lenny, with a Digium PRI card, and have approximately 120 sip peers, mostly Snom 360s, with a few Grandstream GXP2000 and a handful of Handytone 486 units. The

Hello all, i read a few articles online about the possibility to setup a "buzzer" door system to PBX using asterisk! currently my setup contains asterisk of course, and a sipura 3102.. what do i need to get such a feature done?! or should i ask if its possible?! _________________________________________________________________ Connect to the next generation of MSN Messenger?

Hello all, I've been lobbying for some time at the #asterisk IRC channel. Until now, I still can't find a solution to my one way audio problem. I rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS (channel 1). My SIP extension phone located inside the LAN is a SNOM 300 IP phone. This one way audio

I've had it with Dell server garbage. They seem to change RAID controllers as much as I change socks, and then the controllers don't work with Linux, unless you load a new driver. They sell servers with a PCI-e slot in them, but then you get it and find out the RAID controller is using the PCI-e slot! Their sales folks are dumber than rocks, and they change them more often than I

Hey, all. We're rolling out VoIP, and I'm wondering about PoE recommendations, as we're going to have to replace our current network equipment. My first inclination would be to just plunk down the cash and do a Cisco system, but I'm relatively certain that would get shot down by finance. Any recommendations for a couple-hundred-port solution with VLANs, PoE, and QoS? Don't

How does one get asterisk to timeout realtime request via res_odbc to unixODBC? I've set timeouts as appropriate for freetds (which unixODBC is using.) However, it doesn't seem to work. It takes over 3 minutes to timeout a connection and queries never seem to timeout, so a channel waiting on a query never terminates. I did notice that res_odbc.c never sets a timeout on the query