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...ld have assumed none, but I can't even get replies on registration from any of my 3 VoIP providers. I tried defining the External IP and some other stuff, but I assume it's fully an issue with the firewall. Do I really need 5060 port forwarded just to register with remote hosts? Nicholas Blasgen Partner / Network Operations Refractive Dialer LLC (724) 252-7436 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100103/0374e44f/attachment.htm

I've set up plenty of Asterisk boxes but never one that had to deal with a proxy server to be able to use a line. Using "X-Lite" I have no issue with settings as follows: Display Name: Any Name User name: 00575000010XXXX Password: 00575000010XXXX Authorization user name: <blank> Domain: directnationalloan.com Checked "Register with domain" and "Send outbound

...ows "No Answer." (I would show you my CDR but it seems Originate doesn't log in the CDR like every other call for some reason). Any ideas to correct this issue? Or is there a better updated version of that list that would fix my understanding of what the error codes were? Nicholas Blasgen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090511/5fb05feb/attachment.htm

I'm running Asterisk without FreePBX or any of the other managers. I'm trying to figure out how to set the maximum number of channels allowed on a single line? I'd just rather not have Asterisk try the line when I know I'll recieve a CONGESTION back from the SIP phone provider (ViaTalk in this case). Is there a configuration option I can't find that sets the maximum number

I've got a very nice PHP AGI script but I want to be able to do some database cleanup when the user hangs up the phone. I wish everyone would hang up when they were suposed to, but some people don't. So what does Asterisk send to an AGI file when the line has been disconnected? If I remember reading somewhere correctly, I don't need to use DeadAGI. Instead I'm able to use

...'t tried passing "SIP/provider-08748db0" to ChanSpy, but from the documentation it seems that it shouldn't work. So the question is, how can I listen into a channel if I know either the channel or the unqiue id? And in the meantime I will play around with ChanSpy more. Nicholas Blasgen Partner / Network Operations Refractive Dialer LLC 408.395.2110 (w) 408.497.9796 (c) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090202/8f68ec23/attachment.htm

I've got 4 SIP phone lines with a call-limit of 2 for each. I've written a handy macro to allow my users to dial a phone number and the macro will figure out the next available line to use by first checking if the GROUP() is over 2 and then checking to see if ChanIsAvail() as a backup, and if it can't use the line for either reason it goes to the next line. The problem is that there

On Aug 7, 2007 'Mojo' wrote: Nicholas Blasgen wrote: > I've got 4 SIP phone lines with a call-limit of 2 for each. I've > written a handy macro to allow my users to dial a phone number and the > macro will figure out the next available line to use by first checking > if the GROUP() is over 2 and then checking to see if Cha...

I have 10 SIP trunks that I'd really like to round-robin load balance. Currently I have a macro that switches between available lines, but there really must be a function in Asterisk to do this on its own. So my question is just that, are there any easy ways for Asterisk to either balance between SIP trunks or even just a built in function to find the next available SIP trunk. I think using

I've heard about this, but I really can't seem to find anything on it. I've got a strange setup that exists only because of firewall issues, and everything about it seems fine. The setup: SIP clients -> Asterisk (office) -> IAX -> Asterisk (colocation) -> SIP PSTN Termination All the extensions I want to be able to dial are on the colocation box. What I'd really

I'm no longer on the DEV mailing list, but: # svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk svn: URL 'http://svn.digium.com/svn/asterisk/branches/1.4' doesn't exist http://svn.digium.com/svn/asterisk/branches/ -- /Nick -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there any way to see the number of AGI processes that Asterisk is handling? Either console, Asterisk Manager, or from within the AGI? I used to just count the number of running copies of my AGI process (ps aux | grep agi) but once in a blue moon one of my AGI processes will become a zombie or for some other reason not stop when Asterisk disconnects from it. I'd like to know, from

Hi All, I am relatively new to Asterisk. I have CDR enabled and successfully writing to MS SQL server. In my cdr table I am setting the userfield value with a line in my dialplan. If a call is placed to an invalid number (e.g. 12125551212), I see a cdr record created, however, my userfield value never gets set since the call never made it into the context of my dialplan. I am using AMI with the

hello, please any IAX2 ActiveX control that wrap libiax2 or libiaxclient? i want to develope my softphone in delphi thanks __________ Information from ESET NOD32 Antivirus, version of virus signature database 4345 (20090818) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

Hello, I'm running several asterisks in a carrier environment. The asterisks do mainly gateway business between E1 cards and IAX with some routing logic. On one key server I see issues of "Auto-congesting call due to slow response" coming every number of calls. The IAX peer is in the same subnet, the servers are not really loaded. Versions in use are 1.2.2 and 1.4.23-rc3, with rsa

I've got a macro that tries to find the first available SIP trunk to send outgoing calls on. It tracks the usage of the lines (since each trunk has a call-limit of 2) by using GROUP(). My problem is that once a call switched to ANSWER state, ``group show channels`` stops listing it and then my Macro starts screwing up because it's sending calls to a line that sometimes is full even

Hi everyone, I am running into wall today with simultaneous call limits. I have two Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to create a lot of sip calls from one machine to the other by issuing AMI Originate commands to one machine. The machine that makes calls plays a message (demo-intruct) upon the other machine answer. The machine receives the calls just waits for 40 seconds

It gets hard to read my logs when every time someone makes a phone call it displays long pages of "Dropping voice frame". Anyone encounter this before? Asterisk is bridging two SIP lines together, so the technology should be the same. Maybe I'll try allowing only ULAW. ************************************** Asterisk Standard debug (level 3)