similar to: time includes

Hi all. I've been trying to build some new systems, and no matter what I do, if I load the zaptel and tor2 drivers, the system panics within an hour, even with no traffic. These systems are using dual Athlon MP 2800 chips with one, two, or three T400P boards and 2 GB of system memory. I'm currently using Fedora Core 1, but I also went back to our old reliable Red Hat 7.3 and the systems

Does * support ISDN Primary Rate Interface (PRI) - 2B Transfer Capability for T-1/PRI? In other words the ability to take a call and join it to another call and then drop off letting the CO-switch take over. -Kevin Kevin Fjelsted, President AltiCom CTI, Inc. Track Me Down! One number Access, Press 11# during the voice mail message greeting to have me F-O-U-N-D! Phone: 612.259.0722 Fax:

Hi. I've spent the past few hours trying to get SSL working right in Dovecot 1.0.5 and now I must turn to you for help. I purchased an SSL certificate from Go Daddy. I pointed ssl_cert_file to the .crt file and ssl_key_file to the .key file, but the client (Mail.app) complains: Mail was unable to verify the identity of this server, which has a certificate issued to

Hi. I'm using /var/spool/asterisk/outgoing files to place automatic calls, but I'm having trouble setting the Caller ID for the second half of the call. In other words, when we call the first number, we want the Caller ID set to our number, but then when we connect them to the second number, we want _their_ number to be the Caller ID. I've tried the following (and various

I've been testing both T400P and TE405P boards and I'm running into some kind of hard limit on the number of simultaneous calls. This is on x86 with 2 Athlon MP 2800+ CPUs running Fedora Core 1. Everything is fine up to 190 channels, but the 191st call fails every time with errors like: Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on Zap/201-1 Dec 14 15:44:00 WARNING[1215]: Failed

Semi-offtopic, Anyone know of voip providers who can provide tollfree number service? E.g. route 800-xxx numbers to our * ? Even better if they are familiar with * or can speak IAX ... -Dan

We upgraded to the latest version of asterisk (because we needed some newer features), only to find all our PIN applications accepting any number the caller makes up! I traced this to the System application completely changing the way it deals with success or failure of the program it calls. Previously, if the PIN was completely bogus, we exited with -1, which caused asterisk to jump to priority

I tried to test Asterisk 1.4 on FC6 x86_64. I have it working on FC5 x86_64 very good, but since FC keeps updating, I tried to follow newer kernel versions. I can't pass the zaptel compilation. Everything is OK, but when I finished, and tried to load it, allways got module not found when I run modprobe zaptel, and modprobe ztdummy. I already tried to modify is with the sed 1 option but

Hi, I''ve been looking for a light weight alternative to rails for a few small projects, and just came across merb, which looks perfect. The only issue is that merb seems to be tied to mongrel, and I have to deploy to our internal infrastructure which uses FastCGI. How difficult would it be for me to modify merb to support a fcgi interface (actually a rack interface - rack is

Did someone monitor the PRI's or trunks some way? I tried with MRTG and Andrea Fino module but it never worked for me. Any other experience? I want to track the use of my PRI's and trunks using graphical as MRTG does each 5 minute, day, week & Year. But the option of the 5 Minutes I don't think is usefull, We need something more realtime. Thanks, Carlos Alperin

Eli Gottlieb <eligottlieb at gmail.com> writes: > I compiled and installed it to the prefix /usr, but that's not the > issue. Once I actually compile and install LLVM with CMake by hand, I > get the share/llvm/cmake stuff installed correctly (can those files be > included in "normal" builds, or will LLVM switch to CMake as its > primary build system?). Now

Zitat von Israel Gottlieb <isrlgb at gmail.com>: > If you the c option in the dial command it will send answered > else where sip message to the phone and most ip phones understand that > The cell will always display a missed call? I'm very sorry, but I can't understand what you mean... Could you explain, maybe with an example? Thanks Luca Bertoncello (lucabert at

I've asked this question in the IRC Channel, and have had no happiness yet :-( I have incoming lines hooked to asterisk using X100P's. Unfortunately, when we cal forward our lines using the phone company, the line still rings about a half of a time. This is enough to get * to start 'simple switch' and after my 2 second wait, answer the line. Unfortunately, * doesn't see the

I have a directory dumps on my laptop containing several dumps of various levels. local-0-2007-03-03.gz local-4-2007-02-12.gz local-4-2007-02-19.gz local-4-2007-02-26.gz local-4-2007-03-05.gz local-5-2007-03-04.gz local-5-2007-03-06.gz local-5-2007-03-07.gz local-5-2007-03-08.gz Naturally the level-0 is the largest and rarely changes. On the target (access.cims.nyu.edu) I have a directory

Hi guys i upgraded to asterisk 16.10 and in most recordings you here only leg A in the recording sometimes you might hear a word of leg B Did any body hit this problem? Thanks, israel -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200512/90ee8dc2/attachment.html>

I am thinking of getting an asterisk user group together for either SE Michigan or just Metro-Detroit. How much interest in asterisk in Michigan is there on this list? I am already on the board of glimasoutheast, with is a group for technology professionals. (very broad range) It is a spin-off from Automation Alley, which is SE Michigan's version of Silicone Valley. -- Steven

After I actually get everything compiling, install the library, and load it from my Java program, I get the following: > Exception in thread "main" java.lang.UnsatisfiedLinkError: > /usr/lib/libjllvm.so: /usr/lib/libjllvm.so: undefined symbol: > _ZTVN10__cxxabiv120__si_class_type_infoE If I have to guess, this means that the CMake stuff given is linking to the C++ libraries

Yes, I can call the phones, they ring, etc, and call call out, just no outbound audio. Is their any difference in the inbound & outbound audio streams in Asterisk that could cause it, e.g., different ports, protocols, connection/discovery methods, etc? Thanks, Ross ---------- Original Message ---------------------------------- From: "Carlos Alperin"

Another request make me test my t1 card, which has no quality problems, but all that I get is: [root@SIP2-MI zaptel-1.2.1]# ./zttest Opened pseudo zap interface, measuring accuracy... 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%

I finished to try to install Asterisk Now 1.4.0 on an AMD 3800 dual processor machine. The install lookups on the search for the Sata drive, since however it loads the sata_sil driver it doesn't work. Did someone knows what version of Linux is using on Asterisk Now? Thanks, Carlos Alperin -------------- next part -------------- An HTML attachment was scrubbed... URL: