similar to: Nobody there, continuing...

Hey all, I've been trying to get MoH to work from the line-in on my soundcard, but as of yet have had no success. I found this script that should allow for it to happen: http://www.sineapps.com/news.php?rssid=722 The script, when run as the asterisk user, works properly and streams sound to stdin. But when Asterisk starts MoH it stops it immediately afterwards with no explanation. Has anyone

I am creating some custom nagios plugins and I am scraping data from xen using the nagios command: /usr/sbin/xentop -b -i 2 -d1 I am trying to get the network tx / rx data, however I think I am missing something here. NETTX(k) NETRX(k) So total network bandwidth is 1048576 k if the above header from xentop is true. So I am wondering why the output for each vm exceeds the theoretical max, Is

Hello All Anybody had used ooH323 for asterisk i using ooH323-0.7.2 and asterisk CVS may 2005. OpenH323 1.17.1 and pwlib 1.9.0 and GNUGK 2.0.2 audio is very good, better than SIP and IAX, but i have problem. how to router call from openh323 to outside PSTN. my h323.conf setting ; Objective System's H323 Configuration example for tvcti ; ooh323c driver configuration ; ; [general]

Quick reminder before Astricon (from which we will be reporting from live): Tomorrow's guest will be VoIP author Alex Robar. Alex has worked with open source telephony solutions for the past four years, and has collaborated on the development and growth of an international Asterisk-based VoIP peering network. His book is FreePBX 2.5 Powerful Telephony Solutions and we'll be chatting with

Hi Since switching from 1.6.x to 1.8.4 I have noticed the following 1. When you do a 'core show channel <channel name>' the resulting information only shows data for "Frames In" , "Frames out" is always 0. 2. The rtptimeout option in the sip.conf no longer seems to work. I have this set to 60 seconds but have had channels which have not timeout when the rtp

Dears; I am facing a problem in disconnecting the calls, it is related to the rtptimeout (disconnecting if there is no RTP packets from both sides). My Asterisk Box is behind NAT but there is a static real IP address at the ADSL router. We call from the Mobile to the PSTN analogue numbers which are connected to Asterisk Analogue card (the telephone lines are analoge), and then we dial the

OOH323 Debugging Enabled -- Executing Answer("SIP/3513-090f7d40", "") in new stack -- Executing Wait("SIP/3513-090f7d40", "1") in new stack -- Executing DeadAGI("SIP/3513-090f7d40", "a2billing.php|1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php a2billing.php|1: line:58 - IDCONFIG : 1

Hi Samba List, I am finding a strange problem between a mount samba directory. Any clues why this is happening? The server side is WD MyBook World Edition II and the export directory is: /shares/internal/Music/ on the client site I am mounting the directory to /mnt/mybook-music the client is an ubuntu server $ uname -a Linux tsunami 2.6.24-22-generic #1 SMP Mon Nov 24 18:32:42 UTC 2008

Hi samba List, I am finding a strange problem between a mount samba directory. Any clues why this is happening? I have two servers. The samba server is a Western Digital World Edition II (2) server. The samba client (server) is running Ubuntu-kernel linux-2.6.24-22-generic The samba server is exporting the directory: /shares/internal/Music/ The client is mounting the exported directory to

Hi there, might somebody please recommend the best GUI rsync client for XP please? Cheers, Noah

Hey All, I am looking for an example of the proper config settings for background rb to run my rails app and my backgroundrb instance on separate machines (or more precisely, to have a rails cluster with one backgroundrb machine). Does anyone have such examples? Thanks! -Noah -------------- next part -------------- An HTML attachment was scrubbed... URL:

We have two small business that run out of our office. One business has 3 phone lines, and the other has only one. In a perfect world, Asterisk would indicate WHICH line (or group) the outsider caller called, so that we would know which way to answer the phone. The incoming calls would go through a voice mail menu first - different for each company. (Press one for sales, two for

Hi List!! Thanks for the colaboration, especially to Richard Cavanna who gave me the necessary support. I followed your indications and the comunication was better for the test users. The warning indication is no jumping anymore and the voice is not delayed. This is my sip.conf: [general] context=default ;allowguest=no ;realm=mydomain.tld bindport=5060 bindaddr=0.0.0.0 srvlookup=yes

Hi, Our discussions about 64 bit R has led me to another thought. I have a nice dual core 3.0 chip inside my Linux Box (Running Fedora 11.) Is there a version of R that would take advantage of BOTH cores?? (Watching my system performance meter now is interesting, Running R will hold a single core at 100% perfectly, but the other core sites idle.) Thanks! -- Noah

Hi, I want to create variable names from within my code, but can't find any documentation for this. An example is probably the best way to illustrate. I am reading data in from a file, doing a bunch of stuff, and want to generate variables with my output. (I could make a "list of lists" and name all the elements, but I really want separate variables.) ################# #This is

Hi, I use the max function often to find the top value from a matrix or column of a data.frame. Now I'm looking to find the top 2 (or three) values from my data. I know that I could sort the list and then access the first two items, but that seems like the "long way". Is there some way to access "max_2" or similar? Thanks! -- Noah

I am having difficulties with customer-bound DTMF being very short & clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset). My system set up as follows: PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE Asterisk is running Asterisk 10.4.0 on a

How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Thanx, Daniel Arohuanca Lagos +51 1 3594122 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080722/f9612f97/attachment.htm

Hi All - I noticed that the Polycom IP501's are now shipping. Has anyone gotten one yet, and if so, what's different about the phone? Any UI improvements, or is it just better hardware? Thanks, Noah

Hello, I was just assigned to perform a cost effectiveness study in healthcare. We are studying the cost effectiveness of a proposed diagnostic vs. current screening procedures. One of the team members suggest a commercial software package called "TreeAge Pro". Looking at the description, it appears to be a nice GUI to some very simple models that could be easily constructed in R.