similar to: New To Asterisk

Hello again, Here's the next big issue, I thought I'd let you munch on. We are utilizing Cisco 7960's and the following entries in our extensions.conf file: Exten => 1637,1,Dial(SIP/100) Exten => _NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) Exten => _NXXXXXXXXX,2,Congestion Exten => _1NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) Exten => _1NXXXXXXXXX,2,Congestion These

I don''t like this: i = mock(Integer) i.should_receive(:asdfasdf).and_return(''foo'') puts i.asdfasdf Shouldn''t rspec check to see that :asdfasdf is a valid message to be sending Integer? Joe

Howdy y'all, I am using an ATA-186, through Vonage, connected to my X100P. It works well - except, hanging up... The hangup doesn't register, most of the time, with Asterisk. Are there any known tweaks out there that I should look into? I've noted some have altered dsp.c, and rebuilt. Just curious to see what y'all think. Below is my zapata.conf file:: [channels]

Can anyone provide me with a current config for recieving calls with Iconnecthere? I'm having some difficulty with it... Regards, Phillip -- Phil Jackson, President & CEO The Jackson Group - Intelligent IT. (TM) www.jacksongrp.com

Thanks for taking a look greg and hank. This seems to be getting bettre everyday..help please My sjphone is running on the same box as asterisk...i believe then the red hat firewall should not be a problem. Whenever i dial from CLI i get ######### Executing Goto("OSS/dsp", "default|s|1") in new stack -- Goto (default,s,1) -- Executing Wait("OSS/dsp",

Hi all, I'm using R to find duplicates in a set of 6 files containing Part Number information. Before applying the intersect method to identify the duplicates I need to normalize the P/Ns. Converting the P/N to uppercase if alphanumerical and applying an 18 char long zero padding if numerical. When I apply the pn_formatting function (see code below) to "Part Number" column of the

> > If we do end up creating ARM binaries for the general public, your input > > and expertise will be greatly appreciated! ;) > I will be happy to provide some Debian & Ubuntu ARM packages. I just > need access to ARM server(s). I don't know if there is such server available. If so, I also would like give help. :) Regards, chenwj -- Wei-Ren Chen (陳韋任) Computer

In taking a cursory browse at Vonage's site today, I noticed they are now offering a soft phone. Has anyone had any experience using this? And does this possibly open new opportunities for using Vonage with Asterisk? Just thinking outloud on the list, soliciting thoughts and experiences from others. AJ

I've not been able to register with iaxtel.com for the last couple of days. Is anyone else seeing this, or did I miss something?

On 03/04/2013 11:07, Renato Golin wrote: > On 1 April 2013 22:05, Bill Wendling <wendling at apple.com > <mailto:wendling at apple.com>> wrote: > > We would like to support ARM again. > > > Hi Bill, > > Glad you asked! ;) [...] > Sylvestre, > > If we do end up creating ARM binaries for the general public, your input > and expertise will

Dear All. I have installed Asterisk-1.0.6 on FreeBSD-5.4 RELEASE via port, so the chan_h323.so modules already included. When i try the SIP channel, asterisk works fine. In this case, i use the XLite softphone as client, i can hear the voice transfered through clearly. Asterisk is good!! :D The problem is when i try the H323 channel, the voice cannot be transferred through. I have

Hi, I would like to replace the last tree characters of the values of a certain column in a dataframe. This replacement should only take place if the last three characters correspond to the value "/:/" and they should be replaced with ""(blank) I cannot perform a simple gsub because the characters /:/ might also be present somewhere else in the string values and then they

configuration ISDN BRI card : ISDN Olitec PCI 128 (hisax gazel) internet connection by ISDN 64kb/s dynamic IP nom de domaine registered to : dyndns.org avec ddclient to register IP par ddclient asterisk (on internet gateway) configuration pour ISDN BRI par virtual modems /dev/ttyI* (modem.conf) logical telephone SIP "SJPHONE" on 2 local stations "windows" (i don't succeed

Hi All, I've been experimenting during this weekend with asterisk as a softswitch, talk about me being completely lifeless, but let not talk about that. I've been conducting some really funny tests, trying to get an optimal SoftSwitch functionality. Here is my current setup: Source: Windows XP Pro + SJphone Box 1: Asterisk running in PassThorugh mode Box 2: Asterisk running in

Dear all, I would like to know how can I compute the length of a string in a dataframe. Example: SEQUENCE ID TGCTCCCATCTCCACGG HR04FS000000645 ACTGAACTCCCATCTCCAAT HR00000595847847 I would like to know how to compute the length of each SEQUENCE. Best regards, João Fadista [[alternative HTML version deleted]]

I am having a problem with SJphone registration, having read the list and wathced it for a while for similar problems. I just can't seem to figure out the problem. I tryed to follow a tutorial from http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+sjphone, but in SJphone (SIP tab), I can't find the following setting. Use local outbound proxy - checked. Proxy IP Address:

Dear all expert, (i posted this question one time, but i couldn't reach the answer -so allow me to post here) 1)I download asterisk realse version 1.2 beta1. After that i compile it successfully and run it with: asterisk -vvvc 2)I follow the instruction in http://www.asteriskguru.com/tutorials/asterisk_voip_ipphone.html in sip.conf: i add two account: [ivan] type=friend username=ivan

Hi there, I tried to get a few "Optipoint 400 SIP" working with *, but it refused to work properly. In my testing-network i have two Sjphones (they are working really fine) and three optipoints. I?m able to dial the number of a Sjphone on all of the optipoints. The call is signalled at the Sjphone with the right number of the optipoint and an connection can be established. But when I

Hi, I notice that SJPhone is registering to asterisk with an expires of 120 secs. However, when I invoke the command "sip show peer [sip id]". I notice that the output indicates the expires 427 and the expiry is 900. Can someone explain these numbers to me? I also notice that just before SJPhone re-register, when I try to make a call to the SJPhone, asterisk will complain that