similar to: two X100P cards, different context

Hello, I am new on asterisk and do some tests on freepbx. I have 2 SIP provider: Provider1: In-/Out- Flatrate, only 1 Number Provider2: Incoming Flatrate, Outgoing Cost depend on destination, 3 numbers On Asterisk site i have 3 phones (branch ??, don't know how its called in asterisk) Is it possible to do something like: Phone 1: Incoming Call: Number1/Provider1 Outgoing Call:

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 asterisk-users-bounces at lists.digium.com wrote on 12/14/2017 09:36:06 AM: > From: "basti" <mailinglist at unix-solution.de> > To: asterisk-users at lists.digium.com > Date: 12/14/2017 09:36 AM > Subject: Re: [asterisk-users] Rewrite Outgoing Number > Sent by: asterisk-users-bounces at

I need to create a legend for a simple scatter plot in the following format. This is Blah1 number1 number2 This is Blah2 number3 number4 . . . This is Blah6 number11 number12 I looked up these help pages and found the following solution. lStr<-c(Blah1, Blah2,....Blah6, number 1, number2, ...number12) legend(x="topright",lStr,ncol=3) So this creates the tabular format I am

Hi, I am having trouble passing variables via the call files, here is my call file via the php: fputs($oSocket, "Action: login\r\n"); fputs($oSocket, "Events: off\r\n"); fputs($oSocket, "Username: $strUser\r\n"); fputs($oSocket, "Secret: $strSecret\r\n\r\n"); fputs($oSocket, "Action: originate\r\n"); fputs($oSocket,

Hello, I am calculating 'Betweenness' of a large network using R. Currently, I have the node-node information (City1-City2) in an excel file, present in two columns where column A has City1 and column B has City2 that city1 is connected to. These are the steps that I go through to calculate betweenness of my network. a) Convert the City1-City2 (text) into Number1-Number2 in the excel

Hi guys, I have an asterisk box with sip 20 internal extensions and 100 lines registered on a external voip provider. For most part of time, it work fine, but in few moments it act ignoring sip packets becouse my ip phones can't register in asterisk and asterisk can't register his 100 lines in external voip provider. I have log's only for external registration error: [Apr 16

Hi List, Could some kind soul post me a quick config that makes use of a Wildcat X100P when I do a show channels nothing is there lsmod shows the wcfxo and related drivers loaded and with no errors zap show channels is blank as well mucho thanks

Hey everyone, I have a more of an opinion question then a technical question. The asterisk server I am setting up is going to host 3 different businesses. Each business is in the same building, and on the same network. My question is regarding calls coming in and going out. We are a small ISP and have a lot of numbers that are forwarded to our phone system. The other companies have about 3

So I'm not getting caller ID via the X100P card. I've confirmed the PSTN is sending me caller id by plugging in a little third party box. Ideas, tips, greatly appreciated

I have the following: Call -> PSTN -> * -> GrandStream 101 1.0.3.81 The GS displays "ohn ro n2600" when the call is past to the GS. If I pass the call to a XTEN client, Caller ID shows up. Any ideas ??

Hey everyone, I have noticed a few questions close to the issue I am having but I haven't seen any that quite match the problem I am seeing. I have 3 queues. Some members share one queue and some are completely separate. Some members have a higher penalty then others. I am using addqueuememeber and removequeuemember for the login and log out and I verify members with their password for

Hi, I have two loosely related questions which could make my live again a bit easier: 1) Is there a simple way to select a range of columns in a data frame using column names? I am thinking of something like mydf[1,"col4":"col8"] 2) I have a data frame with many columns and they all have short variable names which is good in most cases but sometimes it would be nice to have

As I've said several times on this list[insert usual apology here], I still haven't received the last 20 of 100 phones I ordered over 2 months ago. If you get a hold of them please let me know MATT--- -----Original Message----- From: mikeu [mailto:mikeu@execpc.com] Sent: Saturday, January 10, 2004 12:41 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Chagres

I have a couple of X100P's in my system and while on calls they just randomly hang up for no reason. I have tried messing with the busydetect and callprogress setting them to yes and no same and still random hangups. Is there another setting I should be looking at? My zap config looks like. context = inbound-work include => extensions signalling

Hi, I'm currently running asterisk 1.0.9-r1 and zaptel 1.0.9_p1-r1 (the current gentoo ~x86 versions), with the UK CallerID patches from http://www.lusyn.com/asterisk/patches.html applied. The Zap interface itself seems to work fairly well - although it's a little quiet, there is no echo. Unfortunately, there's also no CallerID. My zapata.conf is as follows: [channels]

Grandstream and Global IP Sound have inked a deal in which Global IP Sound will provide its royalty free iLBC codec to Grandstream. GS will integrate this codec into the BT and HT product lines

anyone around here have the ability to terminate a .NL phone number to IAX or SIP ?? if so please contact me off list. thank you

Hi, When I try to dialout over chan_capi everything works fine when I settle for msn=* in my capi.conf and use the primary msn of my ISDN-line. But trying to configure a different MSN the chan_capi doesn't dial and comes with: No one is available to answer at this time What can be the prob? -- Thanks, Marc aka IzNoGood

Hi Folks, I want to use key() to position the legend on top of the page in the following example. Prefer not to use locator()> par(mfrow = c(2,2))> plot(1:4, title="Plot Number1")> plot(rnorm(10), title="Plot Number2")> plot(1:10, title="Plot Number3")> key(x,y, text=list(c("Alpha","Beta")), text=list(c(1,2)))> > > x,y

Hi guys, I have a customer with asterisk registering 100 lines from my Voip Provider. In some times during a day we receive this messages: [Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request