similar to: Video Conference

I'm replacing a SoundPoint IP 600 with a SoundPoint IP 650. I attempted to simply reuse the existing config files for the old phone on the new phone, but the new phone would lock up on the 4th digit when attempted to dial out certain numbers. So, I downloaded the newest firmware and config templates from Polycom, and attempted to migrate the settings. Seems I'm missing something from

Hello All, I have a requirement to setup a predictive dialer for a customers call center. I am asking for pros and cons of the different dialers available for Asterisk. If you are going to send marketing material send it to my e-mail directly please and not to the list. I was hoping to get the opinions of any one using any of these dialers and what they liked and didn't like, ease of

I am looking for pros and cons on the Intel Atom cpu. Has anybody been using these in production? I am looking at an Atom D510 (dual core 1.6GHz, 1M cache) to run maybe 25 to 30 extensions, 4 or 5 calls at once(maybe as high as 7 simultaneous calls), g729 all the way through except voicemail will be wav format for email purposes(requirement). I will be tying 3 of these together to route

Hello All, I have Asterisk 1.4.5 running on a SuSE 10.3 x86_64 2.6.18.2-34 I upgraded from 1.4.2 to 1.4.5 on sunday the 24 of june and since have been getting: Internal RTCP NTP clock skew detected: lsr=1402479300, now=1402675136, dlsr=196500 (2:998ms), diff=664 I see an entry in Mantis that Russell fixed code so that this will not show when it shouldn't. Would i be correct in

I am trying to setup one time caller id block on my system(activated when an incoming call matches *811XXXXXXXXXX), and I have had little to no luck. Could you take a look at my context/macro definition and help me figure out what I am missing? Here is my context for my dialplan: include=default plancomment=user-default

As I did two years ago, "I'm posting a new thread with the "Top Posting" subject" rather than hijacking the "Paging for Praying" thread. Two questions: 1. Steve K: What do you mean by "/coat"? 2. How do we change rule #5? --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax -------------- next part

Hello all, OS OpenSuSE 10.3 * ver 1.4.26.2 zaptel ver. 1.12 Digium TE122 I have a request for remote users to be able to dial through the system so that the sales managers can barge/chanspy on the sales force. I have the DISA part working with authentication(rather straight forward) but what I can not figure out is how to enable the supervisors to be able to barge on these calls. Is there a

Hello all, Was hoping to get a sanity check along with a question. Below is the output from top run with normal defaults, except to show both CPU's, on a SuSE 10.2 box with Asterisk v1.4.15. top - 10:00:58 up 3 days, 5:54, 4 users, load average: 0.15, 0.05, 0.01 Tasks: 110 total, 2 running, 108 sleeping, 0 stopped, 0 zombie Cpu0 : 0.2%us, 0.2%sy, 0.0%ni, 97.3%id, 2.2%wa, 0.1%hi, 0.0%si,

Hello All, I have a need for a wireless solution and have been looking at the Aastra 57i CT phone that have the wireless handset with them. Aastra says they will cover "up to 300,000 square feet". I am finding this hard to accept. I was also wondering about the "secure WDCT cordless technology" Could this be a form of DECT? Any one using these that can shed some lite?

Hello, I am using a perl script to pull call info from a DB and place calls via telnet and AMI, all on local machine of course. My problem is that I need to capture any response from the carier, such as this taht appears in the CLI: [Mar 1 12:55:50] == Using SIP RTP CoS mark 5 [Mar 1 12:55:50] -- Got SIP response 503 "No Circuit Available" back from xxx.xxx.xxx.xxx:5060 [Mar

Hello, I have a situation where a remote worker dials in to the asterisk server, enters the "secret code", then dials out via Disa on a PRI. This was all working great until this morning. Now the calls is placed out, connected but there is no voice from/to either phone. This is what shows on the CLI when the calls is bridged at a verbose of 4 and a debug of 1: [Jan 25 17:51:40] --

Hello All, I have a setup of ABE on rPath linux,Sangoma A101D, and a T-1 line (Not PRI) which is all happily coexisting and all lights are green. The T-1 comes in from the world into a "Shark Box" which splits the T into 384K data and 6 channels voice. The data side is working great. The voice side, not so great. It was originally broken out to 6 pots line and Verizon came back

Hi all! I am looking for some software that would work as a proxy between a SIP device (SIP phones and ATA boxes) and the Asterisk system running IAX. The reason is that I can only get IAX trow the firewalls, and can't change the settings. One solution I am using are getting several Asterisk system communicate trow IAX bout in this case would I rater have every persons computer act as a proxy

Hi All; Any one can advise for a good stable open source video conference or video server? Regards Bilal ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

I am failing to get compactPDF to make any change to a pdf file that, a/c to the message from the CRAN upload site, can be very substantially compacted. Any ideas what may be wrong? I have also tried recreating the pdf file. I also tried R CMD build --resave-data --compact-vignettes DAAG The data files compact alright (but I get the 'significantly better compression' warning message

Hello all, I have been searching thru all docs that I can find on wiki and such but can not get an answer. I am trying to collect a date from user input in the form of digits dialed from the phone to use in an agi script to do a database look up. I have tried to use "Get Data filename, timeout, maxdigits " in the agi script. In * console I get message saying playing filename but it

Hello all, Ok it is Sunday afternoon and I am going crazy. I have been running in circles so long that I can't think straight. As an example, I sent this message to the wrong address the first try, AAARRRRGGGGGGHHHHH. I have Asterisk 1.6.0.6 and DAHDI Tools Version - 2.1.0.2, DAHDI Version: 2.1.0.4, OpenSuSE 10.3 x86_64, tdm422 at the end of installing dahdi-linux and dahdi-tools I get:

I have a doubt... is it posible to do Video Conferencing using asterisk? -- Facundo Ameal. fameal<at>gmail<dot>com Linux User #395088 Open your mind, use open source.

Hello all, I have seen some chatter again about DTMF. I see most of the talk about DTMF around not being able to get an external IVR to recognize digits, not a big issue for me at this time but sill interesting. My issue though, is with talk off on a zap channel. It seems to be getting worse or maybe my patience is thinning. All my calls go out and come in pstn through an FXO as I do not

I find that if start R-2.6.0 in a workspace with no .RData file, load one of my workspaces from R-2.5.1 or earlier into R-2.6.0, and then before doing anything else type postscript(file="z.ps"), I get:: > ls() character(0) > load(url("http://www.maths.anu.edu.au/~johnm/r/test5.RData")) > postscript(file="z.ps") Error in postscript(file = "z.ps")