search for: sterisk

Hi All, I am trying to get Asterisk up and running on my new Mandrake 9.1 install. I've installed Linux in the "standard" mandrake security mode, and "su" to do my attempts at install. I managed to obtain the source from CVS, and have been able to compile Zaptel. I then ran insmod zaptel, and also make con...

Hi. Dont know if this is an Asterisk or Dahdi or LibSS7 Error. So Im writing to Asterisk List. If somebody knows where to search (dahdi lists or libSS7 lists) will be appreciated. Im getting this error after a certain time, My config is: Hardware: 3 Digium Quad E1 TE4XXP libss7 version: SVN-branch-1.0-r286 DAHDI Version: 2.4.0 Ech...

Hi, I recently installed asterisk, i used sterisk-1.4.20.1, i i set directrtpsetup to yes, no whow would i know if the rtp/media is not passing to asterisk. any tool> or can u just sniff? regards, ron

...sts from your own SER box(es) to land in another context (that DOES have access to PSTN). You can achieve this by adding an entry to your sip.conf for your SER box with it's IP address (and context) specified. ----- Original Message ----- From: braincrew.com To: serusers@iptel.org ; asterisk-users@lists.digium.com Sent: Thursday, March 17, 2005 5:00 AM Subject: [Serusers] ser+asterisk - security Hi there, I'm using ser and =sterisk together. Asterisk for voice mail etc and ser for registration of the =sers usig database. I can restrict =orwarding calls from another s...

...;ZOMBIE> == Spawn extension (hc_fxs, 501, 1) exited non-zero on 'Zap/13-1' -- Hungup 'Zap/13-1' -- Started music on hold, class 'default', on IAX2[ec@ec]/4 == Parked IAX2[ec@ec]/4 on 501 Ouch ... error while writing audio data: : Broken pipe == Parsing '/etc/asterisk/asterisk.conf': Found sterisk CVS-01/20/04-17:15:14, Copyright (C) 1999-2001 Linux Support Services, Inc. ritten by Mark Spencer <markster@linux-support.net> ======================================================================== == Parsing '/etc/asterisk/logger.conf': Found ste...

Hi, I'm trying to add mysql CDR onto a vanilla Asterisk 1.2 install. The add-ons pack has been installed for a while, so now I'm trying to add the Mysql config. I've created a mysql database, added the grants for a user acces, and can run a mysql -u asteriskcdruser -p and can connect to the database. I've been using this as a guide: http:...

When I make a call using sip, I get the line NOTICE[327696]: File rtp.c, Line 292 (ast_rtp_read): Unknown RTP codec 19 received Repeated many times on the console ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to ;bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls allow=gsm allow=ulaw allow=alaw [iconnect] type=friend username=******** password=**** host=sipauth.deltathree.com ;host=...

Hi, I've spotted weird crash of Asterisk cvs Stable. I have defined queue in queues.conf : [prodaja] music = default announce = queue-markq strategy = ringall context = from-pstn timeout = 15 retry = 5 maxlen = 0 announce-holdtime = no announce-frequency = 30 announce-holdtime = yes monitor-format = gsm|wav|wav49 monitor-join = yes even...

, as it seems to be running Asterisk-11. =A0I&#39;ve previously installed A= sterisk-11+FreePBX in a VM, and this appears to be very similar. =A0Is ther= e any upside to using AsteriskNOW vs. Asterisk+FreePBX? Other than the obvi= ous fact that everything is nicely placed on an iso for ease of installatio= n?<br> <br>...

Asterisk Version: 1.2.9.1 Zaptel Version: 1.2.6 LibPri Version: 1.2.3 Hey List, we are running an asterisk server in connection with an octopus telephone system. I have expired some random drops of zap channels bridged to SIP Telefones ( snom 190 ). Asterisk Messages shows something like that: Jun 9 0...

I have a handful of 7912's connected to my asterisk 1.2.14 server. (6 to be exact). I get the X on the display sometimes for loosing registration. I have the config file for the 7912's SipRegInterval: 60 and asterisk is the default. ; maxexpirey=3600 ;defaultexpirey=120 I've not changed them. How can I keep these phones online and stop...

Has anybody have any experience using the KX-TGP500 Cordless DEC 6.0 "System" with asterisk ? I run a small asterisk server at home using two SPA3102s, and thinking of upgrading my cordless analog phones to something a little newer. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20...

...ot; in the message, but now i can't even get this to work ;-(. >As i said, my C knowledge is very limited, it's more like cut&past :-) > >>In my mind, the main objective is to create a special field and force >>its value in chan_capi.c and check wether it goes through asterisk or >>not... What do you think of that? >> >> >Well, i guess the easyest way would be a (new) variable inside >sterisk, ie >${UUS1}, on an incoming call it would contain the message, and on an >outgoing call it could be set to the message we would like to send. > &gt...

<snip> Replaces: pickup-9582-c0a80101-d-4 at 192.168.101.102 <snip> This Replaces header refers to RFC3891 which is not yet supported in Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA) This INVITE fails with : <snip> chan_sip.c: Trying to pick up 7792 at subs <snip> app_directed_pickup.c: No target channel found for 7792. If I'm dialing *87792 instead of using BLF, then I'm entering the dialplan part in...

...iax2.so WARNING........:Unable to bind to 0.0.0.0 port 4569: Address already in use WARNING .......chan_iax2.so: load_module failed, returning -1 Is anyone getting the same errors? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030916/0a47fdde/attachment.htm

or we need to implement a SIP server to integrate with Asterisk in order to= provide full picture of VOIP system? Thanks. > Date: Wed, 5 Sep 2007 13:30:21 +1000 > From: devraj at gmail.com > To: asterisk-users at lists.digium.com > Subject: Re: [asterisk-users] How to make call from asterisk? >=20 > Helps us help you further, what do you...

Hi folks, I am building a new box. Want it to look pretty much like an older Asterisk 1.2, Debian box that is in production. The new box will used as a test box before we implement changes to the production box. New box: ================================================ # cat /etc/issue; uname -a Debian GNU/Linux 4.0 \n \l Linux ServerName 2.6.18-6-686 #1 SMP Mon Oct 13 16:13:09...

I have a second-hand 7960 which I am attempting to upgrade to use a SIP image. The phone currently has a firmware release which doesn't seem to be listed in Cisco docs - P003AM30. On reboot, it finds the tftp server and requests the firmware image listed in OX79XX.txt correctly, displaying "Upgrading Software" on the screen. It then continues to re-request the same image from the

Has anyone written a script that can be used as a cron job or similar that will test if Asterisk is running and if not restart it?? I have just had an issue where asterisk crashed and someone was trying to call me.. it would be nice if it could have been automatically restarted.. I was thinking of a simple bash script something like running "ps -aux |grep asterisk" and then some ki...

Hi All, I am trying to compile meetme2 in my asterisk box and getting the following compilaton error. Please help me to sort it out. cc -fPIC -c -o app_dial.o app_dial.c In file included from app_dial.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:317: `PTHREAD_MUTEX_RECURSIVE' undeclared (fir...