similar to: Changing Transfer key

Can someone tell me where I can get just app.c from. Mine somehow got corrupted, and no updates or anything else will fix it. I just need the one file from the latest cvs. 8-1-04. Please help

Is features.conf included in the cvs as of 8-1-04? I have updated, but am not seeing it?

Is anyone else having problems with Sipuras not being able to re-register to Asterisk after applying the cvs update last night? Just curious if I need to roll back or take all of my Sipuras out back and beat them.

What is the best free stun server out there? The one that I have looked at from vovida requires two NICs. Is this neccessary?

Greetings, I am trying to implement distinctive ringing on a Cisco 7960. I have tried setting alert_info to chirp1 or chirp2 before dialing the phone, but it has no affect. If you have successfully implemented distinctive ringing on a 7960, I would really appreciate seeing the snipit of code that works. Thanks in advance Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775)

Is any one else having problems with Voicepulse today? Suddenly, I can't register and calls to my Voicepulse numbers get a fast busy. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

Any help with this will be greatly appreciated. When re-compiling * to include voicemail access from a MySQL database, I recieve the follwing error. Anybody know how I can fix this? Am I missing packages somewhere? app_voicemail.c:44:25: mysql/mysql.h: No such file or directory In file included from app_voicemail.c:247: mysql-vm-routines.h:7: parse error before '*' token

I have a problem that is almost certainly nat-related, but I can't figure out what's happening. Since moving the Sipura behind a NAT server (Linksys), I am no longer able to call between the two lines on the same Sipura. When I dial one extension from the other, it rings, but immediately after I pick up the ringing phone, the call is uncerimoniously dumped. I can tell the call

Hi all, I bought a couple phones for really cheap just for a simple solution. I'm trying to get a few 7910 to work with *. I'm just not sure how to get them to work. The 7910 just sits there "configuring IP" Here is a copy of my skinny.conf. the extensions.conf is default. I just want to bring the system up in default before a start making changes. Do I need to make

If you have a PRI terminated in a Cisco router talking SIP to * and would be willing to share your Cisco config, please respond. Also, I would be interested in knowing what version of IOS you are using. We are using an NM-HDV in a 3640. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

Asterisk apparently has five echo cancellation algorithms: STEVE, STEVE2, MARK, MARK2 and MARK3. The current default appears to be MARK2. My question is, has anyone had any experience with any of the others (other than MARK2), and is there some conventional wisdom as to when to use one over another? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

I am routing outgoing calls through a sip gateway. The calls go through no problem, however the ringing in the callers ear begins as soon as the last digit is dialed. This has two nasty side effects. First, the caller hears 1-2 more rings than the callee. Second, and more importantly, if the callee's line is busy, the caller continues to get hear ringing, even though the gateway has

Since connecting a PRI to a Digium T100P, I have been seeing the following messages in syslog every few minutes: Jun 4 06:51:54 pbx asterisk[13435]: WARNING[1209214400]: chan_zap.c:6176 in zt_pri_error: PRI: Read on 56 failed: Unknown error 500 Jun 4 06:51:54 pbx asterisk[13435]: NOTICE[1209214400]: chan_zap.c:6913 in pri_dchannel: PRI got event: 8 on span 1 Sometimes, these messages come out

Greetings, I tried to build astcc, but the Makefile is looking for Asterisk/AGI.pm. Anyone have any idea where this file is supposed to be and where it comes from? I've dragged in everything I can think of from cvs, and * is otherwise running fine. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

I've been using realtime to store my voicemail configuration in a mysql table for several months now, and have had no problems...until today. A few weeks ago, I upgraded to the latest CVS and today I noticed voicemail is not updating the password when the user changes it through option 0. I'm not sure when this started happening, but I assume it was sometime after I upgraded. Has anyone

I am now at a total loss. Using Sipura spa-2000s connected to *, I get DTMF working just fine for internal extensions, voicemail, etc. If making an outgoing call like this spa --> * --> Cisco AS5350 --> PSTN, I get no dial tone. I am working unsuccessfully with Cisco right now on this, but they cant find anything wrong. I have tried all suggestions I can find from the list and elsewhere.

I have been searching for the last two days and I cannot seem to set Asterisk to work from a database, can someone please tell me what I am doing wrong here? Here are my files [app_dbodbc.so] => (Database access functions for Asterisk extension logic) == Parsing '/etc/asterisk/odbc.conf': Found > app_dbodbc: dsn is MySQL-asterisk > app_dbodbc: username is

Anybody out there use Ser along with *? Any advantages disadvantages? Is this even a good idea?

Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3234 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090226/a46e68fa/attachment.bin

Hello, I was running a very simple test setup with * HEAD 7/15/2004 on Fedora Core 2 and things were working fine. Today I upgraded to RC1 and my asterisk service will no longer start. I downloaded the tarball, extracted, ran 'make', ran 'service asterisk stop', ran 'make install', removed all files in /etc/asterisk, ran 'make samples' and then ran 'service