similar to: [asterisk-dev] RFC 2833 DTMF w/ Level 3 Sonus

Hello, Straight from the floor of Astricon 2004, I am happy to release my updated Asterisk 1.0 RPMS for RedHat 7.3 and RedHat 9.0 platform. Current Release --------------- asterisk-1.0-0 libpri-1.0-0 zaptel-1.0-0 kernel-module-zaptel-1.0-0 RedHat 7.3 ---------- ftp://ftp.nacs.net/asterisk/rh73/RPMS/ ftp://ftp.nacs.net/asterisk/rh73/SRPMS/ RedHat 9.0 ----------

Neo: What are you trying to tell me? That I can dodge bullets? Morpheus: No, Neo. I'm trying to tell you that when you're ready, you won't have to. There have been over 500 downloads of the RedHat Asterisk RPMS since they were released 2 weeks ago, and I have received many comments to improve them. After some late night hacking this weekend, I have dropped 0.7.1 release 4 RPMS at

Hello all, It's my birthday today, so as my present I would like everyone possible to download and test my updated set of RPMS for Asterisk 0.7.1. By popular request, I installed and built a set of RPMS for RedHat 9.0, and in the process fixed a bunch of issues from the initial build. I have also updated and will be maintaining a page on the Asterisk Wiki located at:

This was so interesting I had to move it to its own thread! Is anyone using this script? How does it perform compared to the older WonderShaper script? -M- ================== Thanks Kristian I will checkout the new script and see how it goes! Jonn -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at

Hello everyone, I just ordered one of these: http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp Just over $110 with shipping but they are expecting the price to come down quite a bit: - 1.2Ghz ARM5 - 512MB RAM - Multiple flash storage options - Gigabit ethernet - USB 2.0 - 5 watt power usage They probably won't be shipping until late March but I

Hello all, Hot on the announcement this morning by Mark, I have updated and rebuilt new Asterisk RPMS for RedHat 7.3, 8, 9 and Fedora Core 1. Please feel free to install these and beat them up. The usual disclaimer applies.. These haven't been tested, not reccomended for production use, don't bug Digium for support of these, blah blah blah.... You can find the updated RPMS and Source

On Wed, Oct 29, 2008 at 1:28 PM, Drew Gibson <drew at oanda.com> wrote: > > I tried out the cdp-tools some time ago (it may have been on your > recommendation, Kristian) but with no success. > Is it possible to disable CDP on the 7940 (image_version : "P0S3-08-2-00")? > > regards, > > Drew > Hmmm... I guess I'd like to know why it didn't work

Hello everyone, I'm trying to turn up a SIP trunk with a Cisco UCM (Unified Communications Manager/Call Manager). It's currently configured for 3rd party call control (3pcc). The INVITEs show up without an SDP... Neither the Cisco admin nor myself can find any documentation on how to disable this feature (3pcc). Does anyone happen to know how to disable 3pcc on Cisco Unified

Hello everyone, I've had the time to examine some specific serial/parallel forking scenarios with Asterisk lately. Looking at chan_sip it appears that anytime Asterisk wants to tear down a call before it's brought up, it sends a 603 Declined: } else { /* Incoming call, not up */ const char *res;

Hello all, Per my last message to the list, and my promise to the Developers that I'd create RPMS if they released 0.7.0, I would like to announce the availability of experimental RPMS for Asterisk release 0.7.1. These are targeted at RedHat 7.3 systems, running the latest Kernel release (2.4.20-28.7). As the RPMS mature and people submit comments, changes, updates and patches, I will

Hey everyone, A while back I worked on a project to measure call quality. I've finally gotten around to releasing it and I'm calling it recqual (Real Call Quality). There isn't much to it and it should be considered alpha quality. I'm hoping some of the bright minds on the list can help me out with it. I'll include the intro text from the README in the tarball: ----

Hello, Please be conscious of Digium's bandwidth and use a Mirror when downloading 1.0. I have mirrored the tarballs at: ftp://ftp.nacs.net/asterisk/ Direct links: ftp://ftp.nacs.net/asterisk/asterisk-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/asterisk-sounds-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/libpri-1.0.0.tar.gz -- Vice President of N2Net, a New Age Consulting Service, Inc.

Hello everyone, Even though a lot of it was a bit last minute, several of us from the commnunity made it to Baltimore to help Digium with their booth at ISPCon. It was a great time. Gregory Boehnlein, Brian Capouch, Christian Savinovich, Kristian Kielhofner (me), and John Todd (not pictured) were there (as well as others), and some pictures were taken (the up close ones of me were very

Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. -- Your

Hello, At one of our locations, we have started to see Polycom 501s (running 1.6.7 firmware) randomly reboot. We have taken packet traces of the phones to determine if there is something odd in the Layer 2 or 3 of the network that might cause it, and have not seen anything strange. There are no errors on the ports. This appears to be affecting POE powered as well as AC powered phones. The Polycom

When we make calls out of asterisk to the PSTN via a SIP termination service provider the called party gets a slight echo of their voice. Here is the setup; analog phone <> Linksys ata <> asterisk <> sip provider sonus GSX 9000 <> PSTN <> called party. The caller on the analog phone connected to the ATA hears no echo at all. The called party has a slight

Hello, all. I've come across a nasty problem just as we are ready to put our first system into production. During our final testing, we were plagued with several "invalid extension" or "password incorrect" messages when we knew the information entered was correct. Upon investigation, we saw that DTMF signals were occasionally but not consistently duplicated. We might

Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. ? thanks in advance Mani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/c1b2ed9d/attachment.htm

Hello all, I am working on a creating some intelligent failover dial-plan logic and I'm running into something that I'd like some feedback on. Basically, it appears that if you place a call to an IAX2 peer that refuses the connection, or is unavailable, a NOANSWER dialstatus is returned. Example: -- Executing Macro("IAX2/cubix-19",

Wondering if anyone has experienced an intermittent one way audio (called party can not hear) problem in these conditions; Several IP501 phones local, same subnet. Remote asterisk No NAT anywhere Polycom IP501 ulaw only, canreinvite=yes Asterisk Call termination path is to a sonus GSX operated by the upstream carrier, ulaw only, canreinvite=no The idea is that if the Polycoms are