search for: damon

What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions?

Hello all I'm trying to get my asterisk config to listen to multiple ports. This is since some clients have port 5060 blocked by their ISP. Does anyone know how to do this in sip.conf or if it is even supported? Thanks!

Has anyone been able to get the IP501 to discover the FTP server IP address (via dhcp or dns) and download 100% of the config from a provisioning server? We are still having to touch each unit to enter the ftp server address and password, as well as set many of the options that will not take from the config file. Have a sample config file you are willing to share? What is required in

Ok, so now Users need to be associated with Organizations. I''ve created a migration and added a ''organization_id'' column to the users table. I want the default organization_id to be the first Organization. So I have :default => Organization.find(:first). But it''s complaining about not being able to find the constant ''Organization''. Any

...r and one for the participants. Participants get music until the moderator joins (to avoid wild, un-moderated tangents). Call is ended and all participants are kicked out when the moderator leaves (or the moderator can kick everyone out via phone keypad). Asking too much, or simple stuff? Damon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060511/8b17f402/attachment.htm

Recent SUP. Installed on Dell PowerEdge 4600. Getting tons of: Sep 30 12:23:16 zige /kernel: bge0: gigabit link up Sep 30 12:23:46 zige last message repeated 98 times Also getting some TCP retransmits. This causes noticable delays in pretty much everything I do now. Not a problem prior to the SUP. I was on 4.8-STABLE before. When pulling/replacing cables, it went into a mode where it would

All, It looks like rake db:schema:dump does not dump primary keys. I have some legacy table that don''t follow the convention ie pk = id and none of the primary keys for these tables are getting dumped... Seems strange as this should be readily available from the db. I am using MySQL. Is anybody else seeing that ? Incidentally how does one specify a primary key using

...: Presentation Prohibited, Failed Screen prohib : Presentation Prohibited, Network Number unavailable : Number Unavailable Have a look at this doc for more info on keypad protocol http://www.ecma-international.org/publications/files/ECMA-ST/Ecma-156.pd f Damon > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michel Koenen > Sent: Monday, August 29, 2005 1:55 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] How to use * a...

Can anyone point me to a reference or sample config for bypassing a nailed up (point to point) t1 between two PBXs with asterisk and a pair of t1 cards? Right now I have 2 Nortel norstars connected to each other via a leased line t1. I also have a solid 10mbps low latency microwave link between the 2 sites. My goal is to run an asterisk box at each end with a t1 card and Ethernet card to

Looking for someone with a successful experience similar to this; I have a need to cross connect a 3COM NBX PBX PRI interface to asterisk, but over a long distance. We do not need any IP connectivity and the solution requires G.711u audio so there is no benefit to using IP. Has anyone here successfully cross connected any PBX PRI interface expecting NI2 PRI signaling B8ZS/ESF with an

Is there a way to get * to use wav files instead of gsm files for the voicemail, agents, and queues applications? Gsm does not give all the quality we would like to have, and we use no low bit rate codecs.

...n managing the list at > asterisk-users-owner@lists.digium.com > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Asterisk-Users digest..." > > >Today's Topics: > > 1. RE: How to use multiple VOIP provider trunks (Damon Estep) > 2. RE: Asterisk on a dialup connection? (Kerry Garrison) > 3. Re: How to use multiple VOIP provider trunks (Tim Pushor) > 4. Re: Comedian Voicemail Issues (Matias G.) > 5. RE: How to use multiple VOIP provider trunks (Damon Estep) > 6. How to park/transfer a call r...

Anyone know how to override the 99 message limit in voicemail? (yeah, we have a public VM that gets that many a day). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050812/1bff12e4/attachment.htm

Just testing the list. I see my messages going through, but didn''t know if anyone else was seeing them. Still having mind boggling problems with Lost MySQL connections. Is no one else seeing this problem? Thanks, M Damon Hill Project Manager IFWORLD, Inc. www.ifworld.com <http://www.ifworld.com/> "..as we''re sung to sleep by philosophies that sing save the trees and kill the children...." -- while you were sleeping (casting crowns) _______________________________________________ Rails...

...roviders WHERE npa = (digits 2 thru 4 of dialed number) AND nxx = (digits 5 thru 7) Then take the provider alias returned and Dial(SIP/${EXTEN}@${provideralias},60). Next step would be to add a loop for multiple providers, starting with the lowest cost. Any hints or comments from the pros? TIA Damon

Hi all, I''m trying to debug some code in my paypal instant payment notification action. Why can I not access the breakpoint placed inside the action that paypal POSTs to? It just doesn''t find the server, but it works fine when placed inside other actions. I''ve appended the code to the end of this post. Thanks everyone! Tom -- def paypal_ipn notify =

Hi Everybody, Has anybody tried the above (or indeed any other 4XBRI cards) successfully with Asterisk. As far as I can see the above mentioned card is an active ISDN card but supported by it's own I4L driver. This leads to interesting questions particularly regarding echo cancellations (which usually doesn't work on the cheap passive cards with one exception as far as I can see).

...ller id name, not number. I know the number is transmitted from the Telco and received by * since the number shows on the incoming call event at the * console. We are not setting the caller id in the extensions.conf, simply passing on what * receives from the PRI (via a single span Digium board). Damon

Hi, I use link_to_remote to create a link to trigger an ajax, things work fine, a new rhtml is created, however, the newly created rhtml doesn''t replace my old zone, it shows instead on top of my old zone...Does someone know why??? my code is like <table> <tr><td colspan="2"><a href="#" onclick="new Ajax.Updater(''zone1'',

...here is a totally different issue I am overlooking? About 3 to 5% of all Polycom to PSTN via asterisk>SIP peer calls are impacted. I have not set the Polycom canreinvite=no yet, hoping to not have to do that as the wan link is a t1 that is also used for data. Thanks for any help! Damon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060405/2169bee3/attachment.htm