similar to: Maybe OT: SIP - Missing 407 messages

who can tell me how to do TDM over enthernet ? pc a connect pc b only use TDM card? thank you John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?23? 11:47 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5, Issue 336 Send Asterisk-Users mailing list

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

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

--- hgaillac-sip@yahoo.fr a ?crit : > hello, > > How asterisk could support res_snmp even this module > don't help to monitor all asterisk features? > > monitoring asterisk with snmp would be a good > thing. > Which solution ? > > Harry > --- Kristian Kielhofner <kris@krisk.org> a ?crit : > > > hgaillac-sip@yahoo.fr wrote: > > > I

While not strictly Asterisk related this issue could certainly affect some of you: http://blog.krisk.org/2013/02/packets-of-death.html -- Kristian Kielhofner

That's great, But does any one know of a package that has capi as part of it. ?? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kristian Kielhofner Sent: Thursday, 16 December 2004 3:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] AstLinux - New Version - w/

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 everyone, Now that astlinux-trunk has been coming along very nicely, I thought I would try to add support for hard realtime capabilities to AstLinux. If everything works (and there are no problems with zaptel), with a little tweaking this should improve the audio quality on systems with high loads (and probably any system at that) - especially if it is finely tuned and has zaptel

Hello everyone, (I first sent this several hours ago, it appears it got lost. Thought I'd give it a second try). Now that astlinux-trunk has been coming along very nicely, I thought I would try to add support for hard realtime capabilities to AstLinux. If everything works (and there are no problems with zaptel), with a little tweaking this should improve the audio quality on systems

I've been spending some time looking at some of the significant changes Apple has made to Facetime in iOS 7. I'm far from an Apple fanboy but some of them are pretty interesting: - multiplexing everything over a single UDP port - deflate compression with SIP - various /slight/ protocol violations ;) More here: http://blog.krisk.org/2013/09/apples-new-facetime-sip-perspective.html As

Hi all, I try to make a call from my Openser(SIP Proxy) to the asterisk in different machine. I use my asterisk as a trunking gateway. I can make a call from my openser to some trunking gateway such as my cisco 5300 or welltech 5250. In the same method, I try to make a call to asterisk ( sip listen on udp 5060 ) I use ngrep on my asterisk machine and list as below. But I can't find any sip

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: ----

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'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, After checking out CVS HEAD from yesterday (for those new PLC/Jitterbuffer patches), I was affected by bug 3795 with my Polycom IP600's. After seing it resolved as of this morning (thanks Mark), I decided to try again... I can answer incoming calls. No problem there. Putting calls on hold, however, results in my Polycom IP600 indicating the call on hold, but the caller does

On Wed, Feb 4, 2009 at 4:00 PM, Gregory Boehnlein <damin at nacs.net> wrote: > Hello, > Is anyone running Asterisk 1.4 w/ RFC2833 to Level3's SONUS network? > We are unable to get reliable RFC 2833 DTMF working, and have instead had to > use G711ULAW w/ INBAND DTMF to get around the issue. Looks like an issue on > the SONUS side. > > Anyone else have this

Hello everyone, I have posted a new version of AstLinux (0.1.6). This one should be the "best" yet (relatively speaking). I have made a lot of init changes, added some applications, and of course - updated to asterisk-1.0.3, zaptel-1.0.3, and libpri-1.0.3. You may also like the "improved" website (don't expect much more) ;). Find more below:

Hello everyone, I have just finished up work on a new release of AstLinux, the embedded/live cd/minimal built from scratch Linux distro centered around Asterisk. Most of the work for 0.2.8 has been on the ISO image (live/install cd), the init system and further system customization with new rc.conf variables. - The ISO image now allows you to install AstLinux to a local disk/CF card,

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

Hey everyone, I came across a situation where I needed to use CDP to advertise a voice vlan to Polycom/Cisco (and other CDP capable phones) without a Cisco switch. Sure I have Cisco switches in places but I like my Polycoms to work out of the box and it isn't always practical to purchase a Cisco switch for every location. cdp-tools homepage: http://gpl.internetconnection.net/ So I