search for: nccom

...ow 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 "imap.nccom.com". The error was: There is no root certificate for this server. So I tried downloading Go Daddy's root certificate and pointing ssl_ca_file to that file, but that didn't help. So I tried pointing ssl_ca_file to the intermediate certificate sent to me by Go Daddy, but that brea...

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

...ers-bounces@lists.digium.com] On Behalf Of > Jim Gottlieb > Sent: Monday, September 26, 2005 10:22 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] system() app changed > drastically! How do I useit now? > > On 2005-09-26 at 18:15, Jim Gottlieb (jimmy@nccom.com) wrote: > > > But since (as far as I know, without using AEL) there is no > > conditional branching based on a variable, how am I > supposed to use this? > > OK, I forgot about GotoIf. However, the doc is wrong (or at > least incomplete), because it only mention...

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:

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

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 a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the

We have bought and installed 10 channels worth of G.729. We reset our ATA186 boxes to use this encoding, and it seemed to be working. However, we analyzed the data stream and saw that the ATAs were sending about 4 kbps, while the stream from asterisk was still 64 kbps, which says to me that asterisk is still using ?-law. I looked in the sample sip.conf file and don't see any place to set

On 2003-04-26 at 00:42, Jim Gottlieb (that's me) wrote: > [ccmenu] > exten=s,1,Ringing > exten=s,2,Wait,2 > exten=s,3,BackGround(5045) > exten=s,4,Goto,outtrunk|17005554223|1 ; if they just wait > exten=_X,1,Goto,outtrunk|17005554223|1 ; if they press 0-9 > exten=_*,1,Goto,outtrunk|17005554223|1 ; if they press * > exten=_#,1,Goto,outtrunk|17005554223|1 ; if they

I just loaded from CVS this afternoon and in the debug output I see... DEBUG[76820]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: m on Zap/16-1 DEBUG[76820]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: u on Zap/16-1 I knew about DTMF 0-9, A-D, *, and #, but I didn't know about m and u :-).

Hi. I need to send calls to different programs depending on where the call originates. For example, I need calls from San Diego (NPA 619 and 858) to to be routed differently than L.A. calls. I tried entries like: exten => 4044633/_619.,1,OurApp,sandiego-queue exten => 4044633/_858.,1,OurApp,sandiego-queue exten => 4044633/_213.,1,OurApp,losangeles-queue exten =>

I upgraded to the lastest CVS (from a version several months old) last night, and a machine set up for Feature Group B no longer waits for digits. I go off-hook, asterisk winks back, and then immediately says: Unknown extension 's' in context 'intrunk' requested I checked to make sure immediate is set to no. Full debug output is: -- Starting simple switch on

I'm trying to filter calls that don't have a proper ANI. This is what I did: ; only if they a real-looking ANI exten=_1XXXXXX1118/_.N.,1,Newt,1118-config ; Otherwise, send them to the loser partyline exten=_1XXXXXX1118,1,Goto(outtrunk,19096611234,1) This properly deals with null ANIs, but for some reason those with ten zeroes get matched by the first line. I also tried to be a bit more

Hi. I'm trying to do a pretty generic IAX2 call between two asterisk machines, but when the call arrives, I get a SEGFAULT. The receiving machine is running the latest code from the stable branch, though this also happened with a snapshot from 2004-01-30 so I don' think it's a recent problem in the code. More likely something I'm doing wrong, but I can't figure out what.

Hi. Can anyone point me to some docs detailing how to set up a connection with Level 3 Communications? A customer of ours wants us to terminate some inbound service via Level 3 to our asterisk server. I've tried all sorts of settings but nothing yet has worked. SIP debug shows a 407 Proxy Authentication Required error. I haven't been able to find anything on the web, and the techs at

If I'm doing a time include in extensions.conf, do I want 04:00-23:00 and 23:00-04:00 or 04:00-22:59 amd 23:00-03:59? I want to make sure that at no time are both or neither included. In other words, does the second time go to HH:MM:00 or HH:MM:59? Thanks.

Hi. Has anyone written anything that can take CDR output and calculate traffic intensity? We're interested in figuring out the maximum number of simultaneous calls we were handling for various phone numbers / services. Thanks...