similar to: Am I missing something really basic here????? help with Asterisk@home

Ok, I thought the point of asterisk@home was that it automatically detected the X100P board and configured it correctly. Is this incorrect? You still need to modify /etc/zaptel files? And not just using the AMP configurator. There is no mention of this on the Asterisk@home webpage. Can anyone who has actually used ast6erisk@home confirm this one way or the other? Thanks, Dean

Hello All, I have 4 X100P cards. I was hoping to have card (line) go to separate ext. i.e. Card 1 (XXX)555-0001 My Ext Card 2 (XXX)555-0002 Wife's Ext Card 3 (XXX)555-0003 Fax Ext Card 4 (XXX)555-0004 My and Wife Ext. This is what I have now and all incoming line rings this one extension. exten => s,1,Dial(SIP/300,10) So what is "s" . Thanks, David -- This message has been

Hello, I'm trying to patch Asterisk for uses wth BroadVoice. I'm running Asterisk@Home. Here is the Error: [root@pbx1 asterisk]# patch < broadvoicesip2.txt can't find file to patch at input line 8 Perhaps you should have used the -p or --strip option? The text leading up to this was: -------------------------- |Index: channels/chan_sip.c

Actually, I love my install of AAH 0.6. When something is not available in AMP I just dive into the configs and correct it. Most of the little things ARE available in AMP though so those times are few... W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of David Sent: Monday, March 21, 2005 9:03 PM To:

I am running Asterisk@Home Voicemail works fine but does not email out the voicemail attachments. Any suggestion? ----------------------------------- Voicemail.conf [general] #include vm_general.inc #include vm_email.inc [default] 201 => {password},Jeff G Laptop,jrglass@columbus.rr.com,,attach=yes --------------------------------------------------------------------- Sip.Conf [201]

Hello, Information: gcc -v: gcc version 4.3.3 (Debian 4.3.3-3) os: Debian/Testing Pulled latest release from asterisk site, compiled, installed it. I have a barebones configuration: $ ls -l asterisk extensions.conf modules.conf sip.conf users.conf voicemail.conf You can see them here: http://home.comcast.net/~jpiszcz/20090418/extensions.conf

Hello, I have been attempting to get the Monitor function to accept a loal variable substitution in order to use the same filename later in the same context. Monitor does not appear to like it, as it attempts to use wav|filename as the recording type, as opposed to just wav. Here is what I get if I just supply a filename directly (it works fine): --context----------------------------- exten

Hi, I'm currently trying to connect an Aastra 57i to our Asterisk Server. The strange thing is, that altough I have definitely entered the correct IP address of the server, the phone doesn't even attempt to register. Here is the configuration file (local.cfg) of the phone: firmware md5: dee6e938b469e217a87138076f47fe41 boot count: 1 tone set: Germany language 1: German time server1:

Anyone using Simpletelecom with *? I had a nice working system with them, my credit can out so I apply another $5 to continue testing. Since then nothing has worked. I always get: -- Executing SetCallerID("SIP/line1-74ac", ""myname"|<>|a") in new stack -- Executing Dial("SIP/line1-74ac",

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Arrggghh. Tearing my hair out here. I'm trying to set up the spa3000 in the UK for my home, and want * to control the dial plan I've googled to no avail. I've read the manual to no avail. Can someone, please let me know what the parameters is the spa and * are to a) receive a call from the pstn b) make a call to the pstn from the phone attached I can make sip to sip calls (i.e. I

Has anyone had any success with getting SLA going between 2 SIP phones? (Particularly a set of Cisco 79xx's) The SLA document that comes with the asterisk source is about as clear as mud. Does anyone have a working sip.conf, sla.conf, and extensions.conf that I can use for reference? The part I'm most confused about is how to build the lines in sip.conf and how the phones should

Hi I looked at few emails related to this subject. And still not sure how to solve the loop detect problem for my case iqbala at improvise:/etc/asterisk$ cat sip.conf [general] context=line1 [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 [line] type=friend context=line1 secret=anothers33cret bindport=5061 host=192.168.1.106

Hello All, Yep I'm a newbe. I'm just started to play with asterisk. What I have Redhat Fedora Core 2 (New install) 3 X100P cards. I installed zaptel-1.0.3 libpri-1.0.3 asterisk-1.0.3 Where should I start?? -- Thanks, David -- This message has been scanned for viruses and dangerous content by KE6UPI, and is believed to be clean. KE6UPI thanks MailScanner for their support. Please

I figured I''d post here before submitting a ticket, but I''m seeing some confusing stuff when dealing with YAML now. I was using it to freeze objects in my database, and so i had some data already around to mess with. I upgraded both Ruby (1.8.3 -> 1.8.4) and Rails ( -> 1.1RC1) and this junk started. I''ve outlined the problem in two pastes, which I''ll

I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway working with asterisk. The Idea is the following. PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} -- Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic system has FX0

Hello, Let's say we have the following function: foo <- function(x) { line1 <- x line2 <- 0 line3 <- line1 + line2 return(line3) } And that we want to change the second line to be: line2 <- 2 How would you do that? The two ways I know of are either to use fix(foo) And change the function. Or to just write the function again. Is there

I've tried several different ways to accomplish this, but as yet to no avail. My y-axis for a plot has a rather long label, and thus I have been using "/n" to break it into two lines. However, to make it technically correct for publication, I also need to use superscript in the label. For example: par(oma=c(0,0,2,0),mar=c(5,6,0.25,2),lheight=1) plot(1:10,

I am trying to implement a rollover of extensions. exten => 3000,1,GotoIf($[${line1} = Congestion]?3:2) exten => 3000,2,Dial(${line1},15,rt) exten => 3000,3,GotoIf($[${line2} = Congestion]?5:4) exten => 3000,4,Dial(${line2},15,rt) exten => 3000,5,GotoIf($[${line3} = Congestion]?7:6) exten => 3000,6,Dial(${line3},15,rt) exten => 3000,7,GotoIf($[${line4} = Congestion]?1:8)

Hi, I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxs (for the fxo cards), which worked fine. ztcfg is showing two channels configured, but when I start asterisk and do show channels, i see no active channels. zapata.conf has: signalling = fxs_ks