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Below is a snipped debug on our PRI. We are getting number only for the CallerID but the telco says they are sending us Name and Number. We are getting the Name in a second frame but Asterisk is not passing it to the device it rings. The message below says "Presenation allowed of network provided number" which leads me to believe Asterisk thinks it should not be displaying it. Can anyone

I'm getting these messages every 7-10 seconds. -- Registered SIP '532' at x.x.x.x port 52956 expires 60 -- Registered SIP '532' at x.x.x.x port 56988 expires 60 -- Registered SIP '529' at x.x.x.x port 51444 expires 60 -- Registered SIP '529' at x.x.x.x port 64044 expires 60 -- Registered SIP '532' at x.x.x.x port 52956 expires 60 -- Registered SIP

No I got confused....yes they are FXO modules with POTS lines coming from bell attached. The only thing I can think of is that since the card supports 3.3v or 5v PCI slots that maybe on a 3.3v slot it requires the other connection all the time because it really does need the 5v and is just not picky about where it comes from. -Jonathan -----Original Message----- From:

How does one go about connecting Asterisk to a Meridian PBX to handle voicemail? I have a customer who is out of capacity on their voicemail system (which connects to their meridian via several FXS cards) and I would like to see if I could use Asterisk to handle their voicemail. -Jonathan

The only phones I have much experience with are the sipura spa-841's, the netweb 301/302 phones (which I really don't like) and the polycom 300/301's. It applies to the sipuras and the polycom's for sure. I can't remember about the netweb, we quit using them sometime last year. -Jonathan -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Here is my file to parse and load extensions. No wise cracks about my code.... DB.php is the Pear DB module. (pear.php.net) <?php include('DB.php'); $db_host = ''; $db_name = ''; $db_login = ''; $db_pass = ''; $db_table = 'extensions_table'; define(DBINFO,"mysql://$db_login:$db_pass@$db_host/$db_name"); $db =

You are right. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tarpo, Louie Sent: Thursday, August 11, 2005 5:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Newbie Question: Building anAsterisk systemtoreplace an old PBX but using existing phone You write out a

My polycom phone is now hanging at Running "sip.ld". I modified it's config via the web interface to register with my asterisk box. I have tried to restore the default settings wth 468* and it doesn't seem to work. Any ideas? -jonathan

Just a poll because I am curious, what kind of call volume do you see? Calls/Day -Jonathan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050909/9b8adf09/attachment.htm

I am getting the following message when trying to lookup up a number via Dundi: Feb 1 13:39:24 NOTICE[20146]: pbx_dundi.c:1309 update_key: No such key 'office.pbx.bluegrass.net.pub' for creating RSA encrypted shared key for '00:a0:c9:55:91:89'! I have created keys on each box with "astgenkey -n office.pbx.bluegrass.net" using the host name for each box of course. I

I'm trying to create an extension that will connect caller to asterisk sound card. I've followed the example at http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+console. With no luck. What I get is: Jul 13 09:56:45 VERBOSE[1315]: -- Executing Dial("SIP/300-3bd6", "CONSOLE/dsp") in new stack Jul 13 09:56:45 WARNING[1315]: No channel type registered for

The best place for Astri Con 2006 would definatly be Omaha, Nebraska! ;) very central ...ah one could hope. __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

Just Google the archive on 'IRQ issues'. You can pretty much bet that 6 TDM cards on 6 PCI slots would suck hugely. Unless echo is your goal, you are not going to be pleased. If you have to use 24 existing POTS lines, look into a channel bank and interface it to a T1 card. If you are planning new, just get a PRI T1 and be done with it. Cheers, W -----Original Message----- From:

Anyone have anything on this? (I'm sure someone will complain about me bringing it up again, chill out...) -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jonathan k. Creasy Sent: Friday, October 14, 2005 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] No Audio

Situation: 8 POTS lines, 3 companies, 1 system. Channel banking the POTS lines onto a T1 thru an ADIT 600. The only way our carrier will provide DNIS is thru Analog DID #'s. Anyone know of a piece of hardware that can add DNIS digits to a particular line? -Darren

https://bugzilla.mindrot.org/show_bug.cgi?id=3477 Bug ID: 3477 Summary: Support environment variable or %u token for User Product: Portable OpenSSH Version: v9.0p1 Hardware: All OS: All Status: NEW Severity: enhancement Priority: P5 Component: ssh Assignee: unassigned-bugs at

No argument here..... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Esben Stien Sent: Thursday, August 11, 2005 8:11 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Firewall will definately increasejittersinyourvoice conversation "Jonathan k. Creasy"

My Polycom IP301 hangs on "Processing Cfg..." Here is the boot log: 0930155446|so |4|00|---------- Initial log entry ---------- 0930155446|so |4|00|+++ Note that bootrom log times are in GMT +++ 0930155446|wdog |4|00|Initial log entry 0930155446|cfg |4|00|Initial log entry 0930155446|copy |4|00|Initial log entry 0930155446|cdp |4|00|Initial log entry 0930155446|cdp

-----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Sunday, October 16, 2005 2:59 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] No Audio from Console but mpg123 from shellworksfine. >One possibility is that the volume is set to 0. aumix can be handy here. Does

Thanks. I was only loading OSS. I installed the alsa development libraries and then loaded alsa instead of oss and everything is working now. Thanks! -Jonathan -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of brett@websmyths.com Sent: Sunday, October 16, 2005 9:00 PM To: asterisk-users@lists.digium.com