similar to: RC1 still broken with Cisco 7960?

It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box

I have two 1-700 numbers from IAXTel. Both get registered from the same Asterisk server. I can make and receive calls on each without any difficulty. What I can't figure out how to do is route the incoming calls differently based on which 1-700 number is dialed. I must be missing something obvious. Thanks -brian -------------- next part -------------- An HTML attachment was scrubbed...

Anybody figured out how to close a VoicePulse Connect! account? As bad as their web site is at most other things, the notion of actually closing an account doesn't appear to have even been contemplated. -brian

In the absence of "The Definitive Guide to Asterisk Dial Plans" book, I'd like to do something possibly unique with the formatting of extensions in my dial plan, and am having trouble. We use VoicePulse connect, which gives us local DID for inbound and outbound calls (even though DTMF tones are not working in Voice Pulse Connect at the moment). To dial local numbers, you have to

Hi, Searchng the archives for the messages containing: error & making & R-1.8 error & make & R-1.8 I found two possibly relevant messages. Both concerned compression routines, and the one reply that I could find indicated that the local libraries might be out of date or corrupt. I think these are the relevant libraries: /usr/lib/libz.so.1 /usr/lib/libz.so.1.1.3

Trying on my Arch machine I get one step further. The splash screen pops up. Then it dies and this is the console output: Code: WINEPREFIX=/home/thom/.wine-poker wine ICMTrainerLight.exe Could not parse file "/home/thom/.local/share/applications/Hattrick Organizer.desktop": Invalid key name: Path[$e] Could not parse file

I've just upgraded my Linux (RedHat v7.2 + v2.4.17 kernel) box to Samba v2.2.3a. Now I'm ready to set up winbindd so that this box may act as a PDC. First, a little background. I have previously been using Samba 2.2.2 as a master workgroup server for Linux and Win98 clients. Now I want to add support for use as a PDC with Win2K clients. I've set up my /etc/nsswitch.conf per the

What you and so may others on this lise seem to forget is that Cisco is a company offering bsuiness products for businesses. Businesses typically pay by check and wire transfer, especially for items such as this. If you want home-user pay-by-credit-card service, buy products from Belkin's home line and similar. Oh...what's that? None of these cheesy Stocked-at-Costco hardware

Hello List, I'm trying to prepare some lecture notes on non parametric methods, and I can't manually reproduce the results of the wilcox.test function for ordinal data. The data I'm using are from David Howell's website, available here http://www.uvm.edu/~dhowell/StatPages/More_Stuff/OrdinalChisq/OrdinalChiSq.html If I run the wilcox.test function on the data I get a p-value of

Hello, I'm considering using Asterisk with some type of Cisco phone, and currently considering either the 7940 or 7960 because of its more-complete functionality (compared to the 7905). I'm currently wondering: Do all the expected functions (transfer, conference, voice mail, message waiting indicator, etc.) work normally with Asterisk over SIP? What caveats are known about using

Anybody know an easy way to adjust audio level of recordings made in Asterisk (using the 'record' application)? I've noticed that recordings using the "wav" format are about twice the level of those made using "WAV" or "wav49". Unfortunately, the "wav" recordings are uncompressed and about 10 times the size of the other formats. -brian

Dear All, X.509 certificates support for OpenSSH version 6.0p1 was published. I brief new version include : - support for Android platform; - engine implementation is now considered stable; - various regression test improvements including fixes for OpenSSL FIPS enabled 1.0.1 stable release and korn shell Yours sincerely, Roumen Petrov -- Get X.509 certificates support in OpenSSH:

Boy after really digging into this, I have discovered that there is more information about each of these topics than I previously realized. Strangely, searching the wiki on "iax" returns exactly nothing. But searching on iax2 does start to dig up some good stuff. Sorry for the hassle. Tough day. -brian -------- Original Message -------- Subject: Re: [Asterisk-Users] IAX config

Hi all, We have recently implemented an Asterisk system using Trixbox (asterisk v1.4.4 at the moment, yet to move to 1.4.9) but are getting pressure to switch back to our old key system unless we fix two major issues. So please help me avoid switching back! An overview: We have about 12 Linksys SPA941 SIP phones connected on a private switched network to our asterisk box which is a

I am getting ready to install Asterisk and I was looking into the Polycom IP600 phones. I spoke with Polycom sales to verify the multiple line appearance and they said it would work. More specifically, if lines 1-3 all contain the same SIP registration info, the Polycom will only send out 1 SIP registration to the server and then handle the calls ringing on multiple lines. I was wondering if

All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk server from VM machine to dedicated machine More than enough bandwidth Setting 802.1p = 7 Set Dedicated voice traffic 35% of bandwidth. Not sure

Hey all. This morning I found that my audio playback is randomly sprinkled with sound skips and dropouts. I went to /var/log/yum.log and found this: Feb 09 20:18:22 Updated: lame-3.99.4-2.el6.rf.i686 I'm not saying that caused the problem but it's all I could find that changed. When I ls in /dev there is no dsp entry. That would explain why mplayer Can't open audio device

Hi all, I am a biostatistician and I have developed my own ranking system for clinical data. I would like to test the efficiency of it w.r.t. to other ranking systems. I would like to simulate the data and after assigning ranks to my observed scores(after neglecting dropouts), observe the type I error. If I want to do a Kruuskal Wallis type of test, what test statistic should I use to test for a

Hello, I am currently pricing out various T1 and PRI options for a client of mine. We need voice and data - we want T's. Whether it be two seperate T's, two superate fractional T's, or one combined fractional T, we need it done. We are getting pricing and one provider is telling us that they have quality issues with the "Integrated Access" product. From what they

Hi! The university where I work just bought four Cisco 7960G IP phones (they didn't ask, just came across the door and gave me a box and told me: "Can you make this work with the Asterisk PBX we have?"). According to what I read, there is no much hope, because I have not the SIP firmware (too bad). Has anybody succesfully got an answer from cisco?, or does anybody happend to