similar to: res_perl voor asterisk 1.2.4

Having trouble running make on res_perl: [root@charlie res_perl]# make perl -MExtUtils::Embed -e xsinit gcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/ -I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk \"-DASTVARLIBDIR=\"/var/lib/asterisk\"

Dear List after a lot googling and watching source example of FastAGI i cant find a simple way to convert a very simple perl AGI script... perhaps im not a developer.. Why i have need to use FastAGI?...Very load CPU usage on my box... with only 100 calls.. So i have two way res_perl or FastAGI on some other box.. I cant test res_perl becasue when i try to compile it i have this error:

Dear all, now we have the same problem of res_perl compilation with asterisk 1.4. It is the same problem that was present when asterisk was upgraded to version 1.2. I hope Anthony Minessale will be able to solve that problem as he did on that case. But if any of you know a hack to this problem please let us know. Here is the same compile problem again: gcc -Wall

Latest version of res_perl is up also. http://www.bkw.org/~brian/res_perl.tar.gz Brian Asterlink.com

On a new * asterisk install onto new install Gentoo 2003.4 upon startup of asterisk: WARNING[16384]: loader.c:248 ast_load_resource: /usr/lib/asterisk/modules/res_perl.so: undefined symbol: PL_thr_key WARNING[16384]: loader.c:429 load_modules: Loading module res_perl.so failed! perl -v = v5.8.5 built for i386-linux-thread-multi <= I installed ithread support in perl Have not been able to

Hi guys, as I wrote in a previous thread I was experiencing dropped audio (apparently randomly) and SIP + IAX peers getting REACHABLE / UNREACHABLE without reason, servers were in the same LAN. Investingating deeply in the problem I also noticed that 'show channels' command on the CLI, sometimes were returning strange results, for example it wasn0t showing some channels I was sure

Hello, To connect to our oracle database from an asterisk application we use res_perl. Sometimes one of our asterisk server will 'freeze' and work anymore. I have to kill the job safe_asterisk and start it again, so that the application asterisk works again. If I look in the log files it look like that asterisk will 'hang or freeze', if two callers calls exactly at the

Hi all, I'm having a problem with some Asterisk servers interconnected with each other using IAX (I also tried with SIP without solving the problem) Sometimes, with apparently no reason, some peers become UNREACHABLE (I have qualify=yes in iax.conf) and REACHABLE again as soon as another qualify test is made. Our users are also complaining about audio loss during their calls, apparently

Hello all, I've been invited to do a presentation on Asterisk for the Ohio Linuxfest in Columbus this weekend (http://www.ohiolinux.org). Rough estimates are that nearly 500 people will be attending. I've been working on an outline for a couple of weeks and I would like to have some peer review of the information presented. I am going to have to cut down the content to make it fit in

> > Now I have two phones connect to my hardware PBX,and want to Make two calls from within Asterisk and switch them together. I now have the two numbers and the other parameter should how to set. for example: the value of data, type and format ,I set the type "Local" and type AST_FORMAT_SLINEAR but I don't know it is write. and the data is don't know how to set. struct

Hi all, I'm doing some coding, i'll be thankfull if anybody can help me here. does anybody knows if ast_request_and_dial() returns differents "reasons" when dialing to SIP and ZAP devices? For example, if the phone of the callee is BUSY, i think ast_request_and_dial() should return the same reason (int) no matter if the phone is a SIP device or ZAP device but apparently is not

ClueCon is coming in 2 weeks so we urge everyone who plans on attending to register today so we get a proper headcount! ClueCon was put together by Asterlink, the same team of people who helped shape Asterisk into what it is today by writing features, fixing bugs, offering IRC support and assisting with the management of the development effort. We have produced several real-world solutions based

Time to reboot and re-start Asterisk, well, hrrm, monthly, news. It's been a hectic fall with a lot to do, both before and after Astricon. At this time, we're preparing for two Astricon shows in 2005. And no, we haven't made a decision on where to run the European Astricon, not yet. I am preparing to travel to the USA again this coming week. Today, I'm spending my time finding

Hi, I'm using the Read command the read a DTMF tone. In this read command I play a voice-file. But now when I press one off they keys of my telephone the voice-file will stop playing a the program go the next priority. Is it possible to play the voice-file until the right DTMF tone is pressed? (say for instance the Zero). Kind regards Arjan Kroon Mobillion B.V.

Can anyone give an example of the difference between the following: canreinvite=no canreinvite=yes canreinvite=update Here is the problem: I have an 800 number sent to me via SIP from a national carrier. Asterisk gets the number and rings my desk phone. Asterisk has 2 NICs, one with public IP and private IP. My phone is on private IP, the inbound call is on public. My phone rings and I answer

Has anybody used the dialplan, "Agents without agent channel" found at <http://www.voip-info.org/tiki-index.php?page=Agents+without+agent+channel#comments> Did it work for you? Did it need much customization. I already note that it requires a res_perl routine (or the removal of the call to same. Anyway, if anybody has used it, I'd like to ask you a few questions. THanks,

The agenda for Astricon Europe in Madrid June 15-17 is now coming together. It will be an international conference, with speakers from both USA and Europe. Last year, we had over 25 nationalities participating in the first Astricon - the Astricon where Mark released Asterisk 1.0 on the conference floor, during his keynote! Many active members of the Asterisk community talks at the conference, one

Hi all, I have a problem with Asterisk dropping audio. While in call, audio gets dropped for a while (from 5 to 60 secs, and obviously users often hangup, this means that I'm not sure the audio is always coming back after 60 secs), in the meantime the call remains up and no SIP signalation is generated. It happens randomly so it's very difficult to debug. I cannot see common

While setting my first couple IP phones, I set their voicemail buttons to an extension that runs VoicemailMain. exten => 8500,1,Wait(1) ; voicemail exten => 8500,2,VoicemailMain ; exten => 8500,3,Hangup ; I would like to be able to pass the mailbox number allowing each phone to go in directly but I'd rather tno have

On Thu, Oct 29, 2020 at 04:13:45PM +0100, Natxo Asenjo wrote: > hi, > using the cockpit web ui and with these instructions: > > https://libvirt.org/dbus.html#usage > > we allow successfully that a group of users can access the console of the > system vms in different kvm hosts. > > Oddly enough, in the same cockpit web interface I can use a terminal, and > if I run