similar to: Best way to limit outgoing calls per trunk

Hi Guys, Currently, I have a Sangoma A400 installed with 20 ZAP PSTN analogue lines. The first line is giving me problems due to rain (probably coroded line). My server using FreePBX dials out with g0 (group 0 which includes all 20 lines) and it happens that the bad line is the very first line. Can I simply put ; in zapata.conf like this to seclude the first zap line from getting calls in or

Hi Guys, Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When trying to dial a number, I get this: tel*CLI> Use of uninitialized value in hash element at /var/www/html/panel/ op_server.pl line 3367. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3372. Use of uninitialized value in pattern match (m//) at

How many simultaneous conference meetme setups can be supported in the same time on Asterisk, and what are the corresponding server's specs for this. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100414/ba26a927/attachment.htm

I am trying to use the GotoIfTime function and get a busy signal. What I am trying to accomplish is to have the system tell callers that we are closed after 5:00pm. Here is the code below. ; If we're open, then go to the open context ; We're open from 9am to 6pm Monday through Friday exten => 3200,1,GotoIfTime(09:00-17:59,mon-fri,*,*?open,3200,1) ; ; We're also late on Tuesday and

Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing "sip set debug peer PROVIDER": Sending to 123.123.123.123 : 5060 (no NAT) ^^^^ That is ALL I am getting with sip debug turned on. With Allow Anonymous SIP set to YES, then the call comes in properly and you see

Hi, When I was testing an IVR, I realized I miss a function I would call GotoIfTimeWithOffset. Today, this IVR is using function AEL GotoIfTime in several places. The problem is if it's 11pm at the moment I'm testing this IVR, I can't nicely test the 9am or 2pm branch. GotoIfTimeWithOffset would get 2 incoming arguments : - the first is a time range (just like GotoIfTime), - the

Hi Everyone, Sorry, if it's not directly related to Asterisk. Some of people on this list might have PBX deployed for their clients. What software do you use to invoice them so the invoice looks like a proper telecom invoice maybe? Prefer: -opensource with Windows binary available. -able to create .pdf invoices rather than printable ones. Thanks -------------- next part -------------- An

Hi Everyone, I understand that there are a few warnings about using "cp" to move .call files to /var/spool/asterisk/outgoing as they might acted upon before copy is done. So, using PHP, What is the equivalent of "mv" command? Would it be rename() in php or is there a better method? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Embarrassed as I am to write this, I am hoping for some advice. One of our very first PBX installs, now six years old, was "taken advantage of" over the past few weeks. A victim of sipvicious, I assume, that managed to guess one of the SIP passwords. 4000 calls to various middle eastern destinations have been placed, which ended up being sent over our customer's PSTN

Hi All, I know I can do this pretty easily with one of the SIP Proxy/Routers, I already do this using OpenSER as a load balancer. I have a special requirement that insist an Asterisk server, 1.6.1.x, is used. I will have 2 SIP trunks coming into the server and I will have to send calls to these SIP trunks with a round robin distribution pattern. I was thinking of using a group count

Hi Guys, I have a need to alter the general timeout in Asterisk. I am wondering if this is something that is hard coded into Asterisk code or if there is a parameter that can be set somewhere. For outbound, I am using x. and hence unless I append a # sign, I would have to wait maybe 5 seconds or so for the call to go through. Is there anywhere in Asterisk that I can change this 5 seconds to

Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3234 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090226/a46e68fa/attachment.bin

Hi Guys, I am making another module for Voicemail. I have three fields in a POST form that have to be connected together to make it a single 10 digit number but there is something wrong in my syntax probably. $npaa = "('$_POST[anpa]')"; $nxxa = "('$_POST[anxx]')"; $blocka = "('$_POST[ablock]')"; *$grplist = $npaa.$nxxa.$blocka;*

Hello, Has anyone already implemented some sort of call preemption in Asterisk ? I am trying to achieve something like this : - I want to limit the number of calls on a given SIP peer to 10 - on the other hand, some calls have higher priority than others - when the ceiling of 10 calls is reached and a call with a high priority is attempted, I would like to drop a call with a lower priority

Hi Everyone, I am accustomed to PUTTY and it's very nice as in it allows many many SSH profiles to be saved and allows tunneling etc....but it's not very good when it comes to scrolling up and down, colors, text size, and specially it doesn't give a title to the opened instance. Maybe giving the IP address as the title of the window would help a lot if you have many different servers

Hi Everyone, There are situations when internet connection is lost, SIP provider fails, or even authentication to SIP provider fails, and we want to use the backup Dahdi channels (PSTN). As simple as it may sound but with the many different situations and error messages it seems like it's not so easy to predict all the errors. Is there any single parameter value that can be changed to send

Is there a way to limit the number of simultaneous outbound SIP calls made by Asterisk? We've tried using the 'Asterisk sip call-limit' parameter but that doesn't seem to be working and one of our engineers says that parameter has been depreciated. Thanks, Deric.Page at nisc.coop -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list! Our company is currently using 3 asterisk boxes in 3 locations connected through iax2. Our main office makes and receives many more calls than the other two. I'm looking for a way to check within the dialplan how many channels are in use at the main office so if it reaches a threshold outgoing calls can be iax'ed to one of the satellite locations. Is there a command I could use

Hi list, For a client I am setting up a system which will use T1 PRI from Primus, who offer only NI-1 and NI-2 protocols for D-Channels. Previousely I have only used switchtypes euroISDN and National. Although the documentation says Asterisk does support NI-1 ans NI-2, but wanted to get your opinion if you have used these protocols on an Asterisk box and if there were any things to consider. If

Hi, I've an PRI coming to my asterisk ,calls are coming fine and my agents are able to answer no prob. but I've an agreement with my telco with some incoming no if the no of calls on these no are more then 3 then send to another no. they use busy signal to divert call on another number so I'm sending the call to Congestion() if no of calls in this group are more then 3. But my