similar to: Asterisk CALLING CARD

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hello all, does anyone has experiencie using asterisk with a digium CARD using G729 managing 120 concurrent calls with SIP and/or H323??? I wanna know if Asterisk is stable doing this....because we wanna implement it in some locations!! Thanks All!! Sebastian. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does asterisk support using an H.323 provider for outgoing calls? From everything I have found, it looks like it does. However, I have had no success in getting it to work. I would really appreciate if somebody could give me a hand. I am using the channel that comes with asterisk. I have also tried using the channel from inaccessnetoworks but have not had any more success. My provider

I downloaded the astcc calling card program. Thanks, it is very easy to setup and works Excellent. Anyway, it says to use DeadAGI to run it rather than AGI. I don't know what I am doing wrong. I just updated my asterisk from cvs and rebuilt and reinstalled. I do not have an application called DeadAGI. I have searched the source, google, etc. but have not been able to find anything.

is h323 per user based working??? I have setup this: [User1] type=user host=xx.xx.xx.xx context=international incominglimit=30 But all calls from xx.xx.xx.xx are not routed to context international, it is working????? I am using chan_h323 Thanks!! Sebastian Nocetti. --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date:

Hello; I've been playing with a nifty Open Source java based report writer called Datavision (datavision.sourceforge.net) and I've managed to write enough logic to calculate phone bills at different rates from the MySQL cdr's. (cdr_addon_mysql) Eventually I want to have sets of rate structures for each user of the system - so I can bill client A at 3 cents a minute and client B at 2

I've been asked to recommend a solution for a one-E1-port PSTN gateway supporting SIP. I've never set up a Cisco 5300 or equivalent, but I know they work. I use the Asterisk software in a couple of places and would like to use the E100P. My question is whether anyone out there has any installations using this and what their opinion is about it (does it work? how's the audio quality?

Just wondering if anyone has tried MAC OS X and panther. I will like to do SIP to H323, not sure if this will be possible on the MAC because of the Libraries PWlib and OPenh32 for Linux.. Just curious.. Anyway, anyone has an easy guide (step by step) to setup oh323 with asterisk. I saw a guide but i am not very savy on linux. thanks, Francisco ----- Original Message ----- From:

Hi list, We're trying to set up a phone number that customers can call to get a hold of anyone of a group of sysadmins (and not their voice mails!). We found the findme example ([1]) that makes the callees press 1 to accept the call. It almost works, but it doesn't work correctly when one of the callees, the sysadmins, hangs up after accepting the call. We're using this

Hi, I have this scenario Cisco 5300 (public ip. 200.47.xx.xx) <---> Asterisk (public ip: 64.76.xx.xx) <--> Cisco 3600 (public ip: 64.76.xx.xx , same network than * ) When a calls comes in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome message and resend call to Cisco 3600 that have 4 analog lines connected... but after cisco play welcome message and when

Hi, I have read quitea bit of the available resources and have this idea of asterisk. Would someone kindly answer these briefly 1-) Asterisk does not need a sound card...but if i am to record voice into an extension or dial from CLI ( basically use asterisk itself as a softphone ) then i need a sound card. : Yes/No a-) If yes creative soundblaster pci 128 is my best bet. Yes/No 2-) Which is

https://bugzilla.mindrot.org/show_bug.cgi?id=3017 Bug ID: 3017 Summary: ExitOnForwardFailure=yes doesn't work for local forwards (-L) Product: Portable OpenSSH Version: 7.6p1 Hardware: Other OS: Linux Status: NEW Severity: normal Priority: P5 Component: ssh

hello all, I am having a trouble with Audio using h.323 channel... I am doing this Call comes into cisco 5300 and is sent to Asterisk, asterisk catch call with h.323 driver and send call to a SoftSwitch that routes the call, I can see log debug telling me, CALLED XXX, and then RINGING, and I can hear ring tones... but when call is answered, I DONT HEAR ANYTHING... I am using lastest ASTERISK

Hi, I have a client who has asked me to look into the delivery of 30 second audio messages to a list of opt-in customers. Probably looking at about 5,000 messages a week over a 6 week period. I know that this would be a piece of cake to have someone develop but I thought I would ask here first if someone is already doing this and what they would charge to take this on as a hosted solution

Hi, I''m trying to run the "validates_inclusion_of" method in a before_save hook, because the range is dependant on the related data. But I can''t figure out how to call it. When calling it normally, it says it can''t find it. I''ve tried several combinations like "ActiveRecord::Validations.validates_inclusion_of", but I can''t seem to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello I'm using the Local channel in an app of mine and I'm finding that the app is being cut out of the call path. You used to be able to avoid this using the \n command but that doesn't seem to work any more. This is on a recent version of Asterisk. Any comments/suggestion? Darren Wiebe darren@aleph-com.net -----BEGIN PGP

Just testing the gateway... plz ignore. -- Posted via http://www.ruby-forum.com/.

What has been changed at the ASTCC from the previous head to the current one? How to use the PIN? Can I avoid it? bye Ronald

Scenario: Here I am working away in R. I've got results that prove global warming is anthropogenic and also the solution for producing limitless carbon-neutral energy from nuclear fusion. Its been a good day. So, I want to save my work. I don't want to overwrite my current .RData, so I save it to another file: save(file="prize.RData") # just need to email this to the Nobel

Good Day Has anybody here looked closely at the call cost calculation in ASTCC? Can you duplicate the way the cost of a call is calculated? I believe that there is an error in the code. I have fixed it, I think and submitted a patch but we need user comments. I would appreciate if anybody involved would slip over to chech out this link on the bugtracker and provide feedback.