search for: asteri

Displaying 20 results from an estimated 73 matches for "asteri".

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2003 Mar 02
1
Serious memory leak in asterisk (manager)
hi all after getting some (or - a lot) of messages from nagios, claiming asterisk to be down, I found this out... astping xxx times repeatedly, and the manager fails to start. this little script was used for testing. below, I've pasted the output from 'ps axfv' before and after the DoS, showing asterisk having allocated ~2GB RAM. roy #!/usr/bin/perl -w use st...
2005 Jan 07
2
Asterisk 1.0.2 - Unable to allocate channel structure
Hi, This morning I had some failed calls. On the console (and in the log) I saw the error "Unable to allocate channel structure". Before I restarted the process, I checked it's memory usage in ps and glanced at my free memory in top. Asterisk was using a normal ammount of memory, about 40M. I don't think this was a system limit. This was running Asterisk v1.0.2. Below is an excerpt of my messages log as well as the output of ps and top, if it helps. Has anyone seen this sort of error before? Any ideas what could be causing it...
2005 Jun 14
2
AVAYA & Asteris & H323 chanel
I'm trying to make H.323 trunk between AVAYA&Asterisk. But call from AVAYA is terminated inmediatelly when apps DIAL on Asterisk is started. Does any one use AVAYA and h.323 channel? Thanks Bob.
2004 Jan 23
3
Problem installing Asterisk with Mandrake 9.1
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 c...
2007 Jun 01
1
Asteris et winsip
Does anyone tried the Winsip sotware to test Asterisk? _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE
2007 Nov 26
0
How to manage several AMI connections to an Asteris server ?
Hi, What is today's status of Asterisk connections management ? Is Astmanproxy still recommended or shall something else be used ? Astmanproxy works for me with Asterisk 1.4 but it seems this software is not updated recently (recent patches are not merged, last modification dates from 16 months). Cheers -------------- next part ----...
2008 Dec 20
2
how to get /var/run/asteris/asterisk.ctl
Hello there everyone, Well I have set up Asteriks 6.0 and almost have Freepbx working too. However, freepbx is showing me that /var/run/asterisk/asterisk.ctl is not found. I confirmed that by going to the directory. How do I get /var/run/asterisk/asterisk.ctl put in correctly? I am using a Ubuntu 8.10 system. Thanks much.
2005 Jul 15
0
OT (kinda): Justification for adding Asteris kto the business plan
..."lock-in" model of the telecom world was nasty and monopolistic and >expensive, but by jove it just worked. >Because as you've seen from this list ANYTHING and EVERYTHING can, and >does, go wrong with this technology. Yes, true, but this is the nature of the beast. It (Asterisk) is an infinitely configurable platform that runs on anything. This essentially is it's greatest strength, and it's greatest weakness. It's easy to make a crappy Asterisk box. It's hard to make one that's five nines. Unlike a closed box solution, where you just pick up the man...
2010 Dec 01
0
<solved!> Asteris 1.8 and mISDN - 'mISDN' (cause 66 - Channel not implemented)
Hello I fixed my problem. I changed user and group in /etc/mISDN.conf: <devnode user="asterisk" group="asterisk" mode="644">mISDN</devnode> now it works again! Thanx 4 help! ;) Ingrid -- "Bonnie & Clyde der Postmaster-Szene!" approved by Postfix-God http://wetterstation-pliening.info http://dokuwiki.nausch.org -------------- next part -...
2009 Jul 08
10
q: install asterisk + asteris-gui
hi, i @asterisk - svn-ed asterisk from digium 1.6 - make install >> its running and i can access the CLI @gui then i -svned asterisk-gui from digium - installed - repointes apache /var/www/1234 >> /var/lib/asterisk/static_html >> now, i see the login box, but i dont have any credentials. tuto...
2007 Mar 07
0
gtalk2voip and Asteris
What kinds of problems were you having? I'm on 1.4.0 and chan_gtalk.so simply doesn't load. Of the 146 files in the /usr/lib/asterisk/modules/ directory, asterisk loads 144 of them, omitting only chan_gtalk.so and res_jabber.so. Connected to Asterisk 1.4.1 currently running on monkey (pid = 9371) Verbosity is at least 3 foo*CLI> module load chan_gtalk.so [Mar 7 10:23:07] WARNING[9407]: loader.c:362 load_dynamic_module: Er...
2016 Feb 10
2
Best place to issue tickets for Digium phones ?
Hello, I've recently given a try to a Digium D70 phone. At the moment, I'm configuring them though config files with a DHCP server and not using DPMA. Of course, I'm connecting them to Asteris (PJSIP stack on 13.7.0). Which is the best place to: - read about past issues - open new tickets for remaining issues. Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160210/d2876c66/...
2004 Apr 29
2
conference & sip
Good day all I've installed asterisk with sip on my LAN,no special cards,if done sip.conf and extensions.conf and all work 100,I'm using x-lite as a client. I'm trying to do conferencing.What I did was to has out the meetme.conf looks like [rooms] conf => 9876 conf => 2345,9938 and extension.conf exten => 9876,1,M...
2006 Jun 15
5
Anyone see this?
Dunno if anyone else has seen this yet: http://www.scmagazine.com/us/news/article/563800/vulnerabilities+put+asterisk+telephone+systems+risk/ -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198
2004 Jun 28
1
cannot make app_prepaid
hai there, today i tried to implement the prepaid application to my * box. I do the step that mantion in to voip-info. i copy the app_prepaid.c and Make file to my asteris/apps, then i run the make. but it show an error like : gcc -shared -Xlinker -x -o app_prepaid.so app_prepaid.o -lpq /usr/bin/ld: cannot find -lpq collect2: ld returned 1 exit status make[1]: *** [app_prepaid.so] Error 1 make[1]: Leaving directory '/usr/src/asterisk/apps' make: *** [subdir...
2004 Aug 26
4
Codec
Good day all I want to know what the best codec is to use for asteris for VOIP We have two towns connected with a 64k line that's going to do VOIP with astersik.At the moment with the default installation the quality is bad and the bandwith is high. Is this even a codec problem Pleas help ALtus
2007 Aug 24
1
TE120P digium card PRI_CPE error
Dear all I got one more error my asterisk E1 card connected with avaya E1 card [avaya]-------E1-----[asterisk] i got this 2 error what is start asteris on consol mode asterisk -vvvvc [Jul 27 09:51:29] WARNING[737] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jul 27 09:51...
2008 Mar 22
1
how to detect redirect fax call
Hi, I want to try detecting if a call is a fax from Zap/1 channel and if it is, forward it to a fax number. How to do it? I have iaxmodem and Hylax working, but it can only receive, but not redirect the fax call. Also, I have read that Asteris has a tool call rxFax. Could someone help me to understanding the difference in terms of functionality between rxfax and hyfax? Which one is better? Thanks, Pete -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/atta...
2008 Jun 19
1
Mapping multimedia keys: "pressed key not recognized"
...ress a key", but when I press the PLAY button on my keyboard, myth says "pressed key not recognized". How do I get myth to recognize the multimedia keys? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080618/b3cc266c/attachment.htm
2009 Aug 13
1
Help for Alcatel asterisk
Hello everybody I have an asterisk with an integration of alcatel pbx, by sip trunk, all calls are fine, but tha calls calls that originate from a analog line, the recipient is not listening, and that if they hear the call originates, the lines are E1 in alcatel pbx. When a asteris user call to analog line the call is ok. Every...