similar to: FGB not waiting for digits

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

I am trying to make an in/out trunk group comprised of 4 DS0's using E&M Wink signalling. The first four channels of a DS1 on a T100P are being used for the group. Outbound calls work fine, but inbound calls fail. The other 20 DS0 channels are used for a PRI. Does the configuration shown below look okay? I've tried setting 'immediate => yes' without success, but it

Hello All, I'm trying to connect Asterisk to an Executone phone system with an analog DID card and I'm hoping someone can help me figure out what I'm doing wrong. The Executone DID card provides battery to the telco, when the telco wishes to dial a DID it goes off-hook, waits for a wink from the Executone and then dials the last three digits on the number with pulse (as opposed

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 can't receive/send calls with Asterisk. Could someone please give me a few pointers on my configuration? Regards, Norman Zhang ; sip.conf [general] disallow=all allow=ulaw port=5060 bindaddr=0.0.0.0 externip=x.x.x.x localnet=192.168.22.0 mask=255.255.255.0 context=inbound-sip maxexpirey=180 defaultexpirey=160 tos=reliability srvlookup=yes register =>

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

Hi! I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my dialplan has an error but my extensions doesnt have the error that show me asterisk. I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist. Any help or any cluees? Verbosity was 5 and is now 7 -- Starting simple switch on 'Zap/1-1' ==

Hello all, I've been playing around with early audio, and I'm able to get some things working We have PSTN calls coming in to asterisk in SIP from a Cisco AS5300. If I do the following: Exten => i,1,Playback(ss-noservice,noanswer) Exten => i,2,Congestion(15) Exten => i,3,Hangup() The PSTN caller does not get an answered call (doesn't get billed) but hears the ss-noservice

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

Hello all. I have installed AsteriskNow 1.7.1-64bits with freePBX. The system has 1 DAHDi card with 2 analog FXO ports (to pstn) and 1 FXS port connected to a FAX machine. I want the every call received on port FXO-2 to be redirected to the FAX machine. So, what I configured was that every call with DID 12345 (let's suppose that 12345 is the DID connected to the FXO-2)to be redirected to the

Hi, I can't get dial-out working. I'm trying to call 523936. Is there something wrong with my setup here? Could someone please give me a few pointers? Regards, Norman Zhang [fwd-out] exten => _8.,1,SetCallerID(${FWDUSERID}) exten => _8.,2,SetCIDName(${FWDUSERNAME}) exten => _8.,3,Dial(SIP/${EXTEN}@fwd,70) exten => _8.,4,Macro(fastbusy) exten => _8.,5,Hangup *CLI>

> Post your extensions.conf and what's on the CLI (asterisk -r) As requested: # cat /etc/asterisk/extensions.conf [incoming] exten => s,1,Answer() exten => s,n,NoOp(CallerID is ${CALLERID}) exten => s,n,NoOp(DID is ${DNID}) exten => s,n,Background(enter-ext-of-person) exten => 1625,1,Playback(digits/1) exten => 1625,n,Goto(digits/1) exten => i,1,NoOp(CallerID is