search for: asteri

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...

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...

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.

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...

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

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 ----...

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.

..."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...

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 -...

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...

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...

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/...

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...

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

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...

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

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...

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...

...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

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...