similar to: Level 3 SIP <--> asterisk

Hi all, I have to transfer thousands of files across my network. We are looking at using rsync via a squid proxy to help improve the network tuning. What I am witnessing now is that with each successive file, the transfer speed resets to 0 and ramps up to 2.5MB/s. Is there anyway that I can group these transfers together to take better advantage of our bandwidth? I should be able to peak at

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 all, I am looking for examples of the extensions.conf that puts all incoming calls into a context where extensions can be dials, and all phones in a context where extensions and outside calls can be dialed. i.e. I have seen: [incoming] include => sip-extensions [sip-extensions] include => longdistance [longdistance] .... Doesn't this allow any internal callers to make external

Hi all, I have a working (it was a pain) set of Cisco 7960 phones. In order to dial I have to either pick up the handset or select the line and then dial the extension or outside line. How do I configure the dialplan so I can: - Start dialing via the keypad and have the phone automatically go to speaker on the first line? - Give the user dialtone after they dial '9'? A while ago I

Hi all, So I have been reading through the docs available online and the different threads on this list, but I cannot seem to get this phone to work. I have configured the OS79XX.TXT and SIP/SEP*.cnf files (see attached), when I configure the phone to point to my tftp server and reboot it I get this message: Connection received from 10.6.0.224 on port 50608 [09/02 12:16:11.750] Read request

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

Hi All. I have notice a problem with FM feature (screen macros) on Asterisk CVS version. When call goes via IAX and calling part "accept the call" on Dial command with option M, in macros context it's setting MACRO_RESULT=CONTINUE, but anyway it hangups both channels. If anyone faced with such problem please let me know. I need to know whether it's bug or just configuration

Hi all, I am working on building a new VoIP PBX. Looking at the current market for phones it seems my best "enterprise" options are the Cisco and Polycom phones. I have some experiance with the Cisco 7940G, but the process of flashing the phone with the SIP firmware left a bad taste in my mouth (not to mention the added expense for the phone). What is the general consensis about

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

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

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

Is it possible to rewrite caller id's? I would like to have sip phones appear by their local cid (like Henk <208>) but when they call out using the PRI I would like their full DID (MSN) to appear (like 0031201234567) I could ofcourse set callerid to the main phonenumber but surely there must be a better solution? Thanks!! Remco

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>