search for: natekapi

I have been trying to make Broadvoice match incoming Caller ID and do specific things based on the number received, but due to Broadvoice requiring the "s" to start off an incoming extension, I cannot get this to work. Has anyone been able to do this? Here are some examples of my setup: from sip.conf: context=broadvoice-incoming from extensions.conf [broadvoice-incoming] exten =>

I currently have my astcc databases card lenghts at 7 digits long. I would like to expand this to 10 digits now though. Will I screw things up if I leave the old 7 digit long pins in there and start using/generating 10 digit pins? Thanks

I've been having a lot of problems with Broadvoice lately. Anyone else been without service for extended periods of time this week?

Does anyone know what happens if you try to have 5 concurrent outgoing channels with VoicePulse Connect? Does it give you an error message or a reorder or something? I'm worried about using them as my primary carrier if this is the case. I noticed that they supposedly only allow 4 channels for free and then you have to pay $20 a month extra per channel. I'm guessing this is for inbound

I'm trying to setup a PBX with Asterisk and the SIP Express Router. Asterisk will handle PSTN incoming and outgoing calls, voicemail and conference rooms, while SER needs to handle SIP registration and SIP to SIP calls while forwarding requests for other features to the Asterisk box. Does anyone have any good examples of using Asterisk and SER together? Thanks in advance.

When using the dial command and the D option to send DTMF digits when the channel is answered, is there a way to allow for some dead air, and then send more DTMF digits? I would like to automate a call, and it requires entry of a few short dtmf digits all a couple seconds apart from each other. Thanks!

I've been getting these really weird errors that scroll almost nonstop across the screen when someone attempts to place a call and it starts ringing. Jul 19 17:55:01 ERROR[23246]: utils.c:509 tvfix: warning negative timestamp -536835.-70000 Does anyone know what this is all about and what I can do to fix it? Thanks for any help in advance, this is causing huge problems!

I have a really good high speed connection on my dedicated server, but I still get jitter even if there is only 1 call going on at a time. Will tweaking my iax.conf's jitter buffer settings really make any significant difference? Does anyone have a really good example that they have found to work good, besides the default one? Thanks for any help in advance!

I saw someone mention on the security lists that IAX2 encryption has been around for about a month now. Is this true? Where can I find more info on how to set this up? Thanks!!

Is there a way to rewrite the agi_callingpres variable that is displayed when you have AGI debugging on? I'm guessing 0 means present the caller id and 1 means block the caller id. I would like to be able to easily block the caller id number from showing, and was hoping that I could just reset this variable. Thanks!

I know that the wiki has an extensive list of European VoIP providers out there....but there's so many that it's kind of hard to sort through. So I was wondering if anyone could recommend some reliable SIP/IAX termination providers in Europe? Something like VoicePulse Connect, NuFone, Vitelity, or Junction Networks based out of Europe. I really don't trust a US VoIP company for