similar to: Ser as IVR

I am curious if it is advisable to use implement Asterisk as a Session Border Controller for a VoIP reseller environment. Users will terminate calls SIP to my server, which will authenticate them via RADIUS, perform a LCR lookup, select an appropriate trunk (based on LCR), and terminate the call (update RADIUS accounting at end of call). All while acting as a B2BUA to prevent the users from seeing

hi! First of all i want to tell i have a dedicated server on layeredtech with direct internet connection and i currently dont use iptables, so this is not about network configuration =). well so, i install asterisk-1.4.2 on my server, and next install asterisk-gui from the digium repository. next i go to: http://pbxa.com:8088/asterisk/static/config/cfgbasic.html and install a default

set_format: Unable to find a codec translation path from ulaw to g729 Both endpoints are PAP2 set to G711 only I have 1.2.17 on Suse 10.1

Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? I use Asterisk now for my phone system. Thanks! Wiley E. Siler Director of Information Technology Education 2020 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:wsiler@education2020.com <mailto:wsiler@education2020.com>

Hello eveybody, I've been looking for a way to share trunks between two asterisk servers. I guest I have to use Dundi, but I've not found the exact method yet. I need a way to allow users registered in one server to use the another server's trunks in the case the first server's trunks were busy and vice versa. Is this possible? Thank you so much, any comment will be useful.

Hello, I'm looking to do the following, and I wonder if Asterisk can be used for it, and if yes, if anyone can point me to the relevant information (commands, sample config...): 1. Caller dials 111, 222 or 333. 2. Based on the dialed number, Asterisk queries an external MySQL table and retrieves alphanumeric data, plays/announces it to the user and deletes the row from the database: The SQL

I am not sure of the best way to do this, so I thought I would query the list. I have about 100 internal extensions ranging from 2000 - 2100. Each internal extension has a external DID number. For example: 2001 = 5552871620. As you can see the internal externsion and DID don't match in any way. What would be the best way to set the DID for when a extension dials out on the PRI? In

I was just curious to what your redundancy solution is. I have considered many options, so I thought I would share and get an idea for what others are doing. My setup is two different locations with a 10MB WLAN fiber link between the two. Each location has it's own PRI as well. I have considered and tested many options this last year or so. 1) Using hearbeat and drbd to monitor the

Hi....asterisk users... how to install asterisk on redhat ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070410/0e647e89/attachment-0001.htm

Hi List, I need to change my provider, at this time Asterisk box is on VOIP trunk. I have two options, T1 or 15 analog lines. I have some experience with analog and I have had two main issues with it. first is echo (I have not tried HPEC yet) and second unpredictable volume. The question is, if I use TE100 with PRI , will I have same issues? I would appreciate any comments and sample zaptel.conf

All, I have done research on VoIP for some time now. I'm a Cisco certified Network Engineer however Telecom is not my strongest suit. I've been a part of this mailing list for sometime but my delusions of grandeur in migrating our 25 year old phone system to a new platform have been on the back burner, until now. I have found my company is moving to a new location(building) and this

Does anyone know if is possible to purposely set red alarm status on PRI circuit for testing purposes (other than unplugging it). I have looked for a console command which might allow this.... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070411/985e8e64/attachment.htm

Hi there, We're trying to get IMAP voicemail storage working on an MS Exchange server - I would be grateful if anyone who has successfully done this could post the magic soup here, as extensive Google searching has yielded nothing other than tantalizing references to it being done without any specifics. -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web:

What wrong with this: [get-dnisinfo] ; sub-routine to get owner's password exten => s,1,Verbose( == ) exten => s,n,MYSQL(Connect connid localhost root password dax) exten => s,n,MYSQL(Query resultid ${connid} SELECT\ password\ FROM\ dnislookup\ WHERE\ dnis=\'${IVR-Exten}\') exten => s,n,MYSQL(Fetch fetchid ${password} password) exten => s,n,Verbose( == Password found

Can anyone tell me what the capacity is of 2 E1's in minutes. Ie how many minutes can 2 E1's take. Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070413/de59fcf5/attachment.htm

Hi all, lets say I've registered at several Sip-Providers. Provider A offers best rates but is often too busy to get a line. Sip Provider B is stable (but more expensive). The asterisk box has a high call volume therefore problems at provider A will get obvious after a few calls stalled. In this case astersik shall switch temporarily to provider B but shall test periodically for selected

Hello all, I currently have all outgoing calls set to mask the caller id so it will always appear to be coming from our main number. The problem I'm having though, is with both the call detail in mysql and with the automon (recording) feature. It shows the originating number as the number I masked it to, rather than the actual person calling. How can I go about having both the destination see

Hello, Could somebody tell me is it possible to use asterisk without RTP proxying in SIP<->H323 calls? I mean exactly what canreinvite=yes option do in SIP<->SIP calls. I don't need a transcoding, only a signaling conversion, and this is possible with some softswitches, so i wondering what about asterisk. Same question about H323<->H323 calls I'm using NuFone

Hi all, Is using a Citel gateway with Asterisk a good solution for reusing of the old Nortel digital phones? Would love to get some input from actual users. Any/all opinions welcome. robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070510/bc5fc18f/attachment.htm

Hi, MoH volume is uncomfortably high and I want to bring it down. Its mpg123. How can I do it? -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070514/2a40a5d4/attachment.htm