similar to: how to detect redirect fax call

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

Hello everybody, Which one is a better choice 1. Gateway device with FXO <-> SIP ( example Addpac http://www.addpac.com/addpac_eng2/addpac_product_view_detail.php?class_id=19&item_id=59 ) 2. Digium (Wildcard TDM400P) 3. Sangoma (A200 Analog FXO/FXS) All i need is to put asterisk in place with 4-8 incomming lines (ordinary POTS ). With IVR, Voice mail and International Call via SIP.

Why is what he is doing different than having the fax machine on a Sipura ATA? Just because both those ports are on the pci card that doesn't make them not Voice in between....if I'm wrong....eh...oh well.... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F Sent: Wednesday, September 28, 2005 9:45

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

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,MeetMe,9876 When I go onto x-lite and type 9876 it gives me

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. Everyone, has been that problem? I change asterisk version 1.4.21 to

Hi, in FreeBSD kernel is options that make router stealth, packet passed throw router have no change in TTL value. Is a posibility to make similar thing in Linux, I think that in traceroute isn''t this hop, no asteris, no router, no change in TTL, nothing ...? PSIkappa psi@atlantis.sk

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 09:51:30] WARNING[737] chan_zap.c: PRI Error on

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

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

Hi, I just got a Polycom 330 and, of course, I don't have the firmware and sip.cfg files to provision it. Where can I get those? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070605/a14615c8/attachment.htm

1. Search the archives 2. Search again 3. Now search the internet 4. The fact is that faxing over VoIP without T.38 doesnt really work. Because it works 60% of the time it doesn't mean that it works, until it works 99% of the time. Since Asterisk does NOT support T.38 it doens't support faxing over VoIP other solutions are available, search the wiki. On 9/26/05, Mark Armstrong

I have a text file that I have imported into R. It contains 3 columns and 316940 rows. The first column is vegetation plot ID, the second species names and the third is a cover value (numeric). I imported using the read.table function. My problem is this. I need to reformat the information as a matrix, with the first column becoming the row labels and the second the column labels and the cover

Hello R Users, I am new to R. I have two data sets i) TOMS aerosol optical depth(AOD) and ii) TOMS ozone(O3). > > AOD data is on 1x1 grid and O3 data is on 5x5 grid. > > First I want to read AOD and O3 as it is and then I want to regrid AOD on > 5x5 grid as O3. > > Reading is first problem. > > FIRST PROBLEM READING AOD: > > AOD data is in following format: >

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. tutorials are suggestion manager.conf, BUT I DONT HAVE that file, in fact /etc/asterisk is

Hello i would like to receive incoming faxes thru' asterisk as tiff files thru' the rxfax application. I setup extensions 101 like this exten=> 101,1,rxfax(/tmp/fax.tif) then from CLI i run: dial 101 and rxfax send me his "scream" about the fax ^^ instead when i send a real fax from a faxmachine to that extension my 101+rxfax is executed but it just "does

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

I'm using Slackware 11. I unistalled the package that provides libtiff 3.8..... and installed the most current 3.7.... for lib tiff. I downloaded asterisk 1.4 beta3 and the 1.4 beta2 addons and untared them. created a simlink: ln -s asterisk-1.4.0-beta3 asterisk I've compiled spandsp from as follows cd /usr/src wget

Hi, I noticed a strange behaviour: Faxing using spandsp (TxFax) from my asterisk box to my old, common fax machine at home works fine. Faxing from the same box to my office pc-fax (Hylafax) also worke fine. Receveiving faxes on my asterisk box using spandsp (RxFax) also works fine. It is a PSTN number connected to the digium card of that asterisk box. Then I faxed from my asterisk box (TxFax)

Hey All, I am attempting to get the RXFax app working and having a hell of a time of it. I am hoping that some of you fine folks can help me out. I have installed Asterisk v1.2.14, SpanDSP v0.0.2pre26 and app_rxfax. All compiled and installed fine. When I attempt to call the extension I have created for receiving fax's then I get the following error once just as the rxfax