similar to: AGI problem (crash) in RH9

Hi Every time I hangup on my AGI script Asterisk crashes if it is not running in console mode. (happens when using python and perl AGI scripts) I'm desparatly trying to get my employer to let me use Asterisk. So I must get this to work. I've posted about this before, I'm sorry, but I'm desperate. I'm running RedHat 9.0 (kernel 2.4.20-8 everything else updated) I'm

What types of ISDN BRI cards work well in Europe (Guadeloupe, Martinique and France) ? For example: AVM C2 or AVM C4 or eicon Diva server 4 BRI? Any others? Which driver is appropriate? Ray Burkholder ray@oneunified.net http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -------------- next part

We needed the client browser to be open all the time for dynamic data to load without the page refreshing. After looking at all of our options we decided on programming it ourselves using flash rather than java. We have a flash frontend thats tied to our backend mysql DB. We use it for loading web site traffic data, email opens, click-throughs, bouncebacks, stats, etc. It could also be used with

I have a few 7960 Skinny phones. I've edited the skinny.conf file, but I'm a little unsure as to how get the phone to figure out which ip address it should register with when it boots. How do I do that? I already have a tftp server for my SIP based phones. Do I need a tftp server for skinny configs at all? And if so, can it be the same tftp server as the SIP ones use (I'm not sure

I have several Cisco SIP gateways sending calls to Asterisk. Because the gateways don't have user-agents, they don't authenticate with Asterisk. And because they don't authenticate, they use the default context in the sip.conf file. Is there a way to either: A) identify the inbound gateway with a variable, in channel info, or the manager interface? If there was a ${SIPDOMAIN} for

> The documentation mentions that the Digium channels can be split into some > voice channels and the remainder of the channels used for routing IP > traffic. > > Does any one have this in use in conjunction with Asterisk? Does it work > well? Would you recommend it for a production server? > > Obviously, if this works, this makes for a cost effective platform where

Jeremy, In some posting in the mailing lists, you mentioned that docs for h323 had been submitted but hadn't made it into distribution. Could you post those docs in your download directory? I'm trying to understand the nuances of your driver, gnugk, and extensions. Ray Burkholder ray@oneunified.net http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content

Ok, I've managed to get inbound and outbound calling to work with chan_h323 and gnugk. A few questions: 1) if I do a reload in *, chan_h323 loses its registration with gnugk, and will no longer pass calls to it. A second reload will crash *. Is this supposed to be? 2) For a configuration in h323.conf like: [office] type=h323 prefix=9 context=outbound I get a message saying:

I'm trying to come up with an elegant solution to handle roaming users in a branch office scenario. I have a number of possible scenarios, none of which seem to completely solve the problem. Perhaps someone with a better feel of the interactions can help me out. Is the 'switch' statement useful in some way? What are the ins and outs of the 'switch' statement? Come to think

Hi I work at a small company that has some IVR solutions that use Dialogic hardware for everything. Everything is written in C++ using MS VC++ using the Dialogic API and runs only on Windows. Being the rebel that I am, I would like free myself from Dialogic. To do this without porting all our existing code to run on Linux I was thinking of controlling the Asterisk from a Windows machine running

I am currently using a copy of Asterisk checked out as the code of 10 days ago from Asterisk and the: sip show inuse reports that I have 3 incoming connections to one of the Grandstream phones, even though that isn't the case. I believe I have tracked the problem down to the following error message, which also (conveniently) showed up 3 times: -- Got SIP response 481 ""

Hi Just installed Asterisk on a Debian Woody/testing. I want to create a AGI script that is run after an outbound call is answered. I did this a while back (many versions ago). The problem is Asterisk does not seem to know the AGI application. I create a file test.call and place it in the outbound spool directory: the test.call file looks like this: #Simple test call script. #call my

>I spoke with someone today who is interested in an IP Centrex solution that >starts with about 3500 extensions in a multi-tenant application. And >growing from there. > >I'm wondering about scalability of Asterisk. I'm trying to put my head >around how to put the whole thing together, if it can be put together. > >The nice thing about it is that if I can show

Hi I've just started to play around with AGI scripts and have run into problems. When I run Asterisk in console mode everything works just fine. If I run Asterisk in 'regular' mode (not console) it crashes if I hang up on the script. I have used Python scripts to test this and also the "agi-test.agi" script. (the Asterisk code was compiled from CVS code just a few days

I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/07/04-11:28:50\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

Do the Polycom IP phones have some programmability so you can do some programmable phone buttons like you can on the Cisco phones? If there is programmability, such as for soft-keys and the like, how would you rate Polycom's vs Cisco's capabilities? And where can one find the programming documentation? Thanx. Ray. -- Scanned for viruses and dangerous content at

Anyone know a sip-client that will work on a Handheld PC running WINCE for HPC. I can find some for PocketPC, but the wont work on my HPC ?? /HHA _________________________________________________________________ Scope out the new MSN Plus Internet Software — optimizes dial-up to the max! http://join.msn.com/?pgmarket=en-us&page=byoa/plus&ST=1

Hi *ers, I recently got an Email from Redhat about the dropping of support for Redhat 9 on the 30 of April and that Fedora Project is the recommended future, otherwise, RedHat enterprise ($$$). Considering this, I would like some feed back on the Fedora Project from users who may be using it, and how its going with Asterisk? Are there any problems? Is the Asterisk development team got Fedora

I read about this intercom stuff on page 62 & 63 of the book "Developing Cisco IP Phone Services" isbn 1-58705-060-9. Primary calls take place on streaming channel 0. When streaming channel 0 is not in use, streaming channel 1 can be used for asynchronously streaming (in and out) stuff like voicemail, email, and, yep the one we want, intercom. Page 87-88 of the book talks about

Hello, I've been playing around with * for quite a while now, and have run into a problem that I just cannot seem to figure out. When using * and any IAX client (I have tested with GnoPhone and both clients from iaxclient.sourceforge.net) I have incredibly bad jitter on the connection. What I'm running is a P3-1Ghz machine with 512mb ram for a server. The other end has been