similar to: Qestion about TDM over enthernet

Displaying 20 results from an estimated 4000 matches similar to: "Qestion about TDM over enthernet"

2004 Dec 20
7
'I'nvalid extension handling problems, even with workaround
Hello folks, I'm having trouble configuring Asterisk to play an "invalid extension" message to anyone dialing an undefined extension. First I tried using the 'i' pseudo-extension, but it didn't work at all; searching the wiki I found that page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension where it basically says that the 'i'
2005 May 28
1
Pictures of the Digium booth at ISPCon 2005
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
2008 Nov 01
1
VoIP traffic shaping
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
2004 Sep 19
1
Timing source on SMP system - Disable RTC forzaprtc
Kristian, I have 2 X100P cards but neither work on my Compaq DL360 G2. The system will not even boot! Take a look at my initial post and let me know if you have any other advice. Regardless, thanks for your post! ------------------------------------------------------------------------ - I need a timing device for the DL360G2 for conferencing and meetme. For a timing device I have 2 X100P cards
2005 Feb 24
2
Brainstorm: Running Asterisk as cool as poss ible - AKA solid state.
Hi Kristian, Anywhere I can read about this Soekris/AstLinux project? ... Regards, Hans -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Kristian Kielhofner Sent: Thursday, February 24, 2005 6:02 AM To: jim@vanmeggelen.ca; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]
2006 Jun 07
1
asterisk-1.2.9 / res_snmp.so
--- 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
2004 Sep 19
0
Timing source on SMP system - DisableRTC forzaprtc
Yes, I tried adding an adaptec that I thought was uhci. This didn't work. I didn't check the specs but figured that they would be uhci given the history of ohci. Regardless, I will confirm with adaptec. Thanks, Chad -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kristian Kielhofner Sent: Sunday,
2006 Mar 07
0
R: Capturing DTMF during a call
Thanks Kristian, but i just answered to call, how can i use the Read application? Thanks Giordano Grandis -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Kristian Kielhofner Inviato: luned? 6 marzo 2006 18.15 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Capturing
2004 Dec 22
2
Why use 'Answer'?
Why is it that newcomers always feel like inserting 'Answer' is a necessary step in their extension.conf entries? >[voiptalk.org] >;forwards any calls starting with an "8" thru voiptalk.org >exten => _8.,1,Answer >exten => _8.,3,SetCIDNum(55555555) >exten => _8.,4,SetCIDName(My Name And Surname) >exten => _8.,5,Dial(SIP/${EXTEN:1}@voiptalk.org,,g)
2006 Jun 01
4
G729, voicemail, no codec_g729
I am trying to create a %100 g729 (with no transcoding) system (using a Soekris, of course). I am running AstLinux with the native sounds, g729 is the only codec allowed, %100 SIP (g729 only allow=) - I think I am covering all of my bases. I have only "format=g729" in voicemail.conf. On an incoming call to a mailbox, everything goes well until recording the message. When the
2006 Jun 01
1
G729 + Native (files) MOH
Hello everyone, One more little problem with a %100 g729 setup. Native moh: musiconhold.conf: [default] mode=files directory=/mnt/kd/moh/default random=yes ; Play the files in a random order ls /mnt/kd/moh/default fpm-calm-river.g729 fpm-calm-river.ulaw fpm-sunshine.g729 fpm-sunshine.ulaw fpm-world-mix.g729 fpm-world-mix.ulaw Place a call on hold: Jun 1 14:55:30
2004 Dec 15
0
AstLinux - New Version - w/ 1.0.3 what about capi!!!!
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/
2005 Jul 04
0
SV: Epia C3 Linux
Hello AstLinux seems quite suited for my use. Can you configure more incoming port via a web interface? I'd like to install it to a "normal" hdd. Can that cause any problems? BR Amund Nygaard -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Kristian Kielhofner Sendt: 4. juli 2005 03:23 Til:
2006 Nov 01
0
AW: Which IP phones have best voice quality, preferably under $150
snom 300 :"> CS -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Im Auftrag von Kristian Kielhofner Gesendet: Mittwoch, 1. November 2006 12:01 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] Which IP phones have best voice quality,preferably under $150 Zeeshan Zakaria
2006 Nov 30
0
Voicemail callback bug?
Which version? Similar issues parsing callback number in 1.2.12 > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Kristian Kielhofner > Sent: Thursday, September 28, 2006 10:27 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Voicemail callback
2013 Feb 06
2
Somewhat OT: Specific SIP packets can cause ethernet controller reset
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
2004 Dec 08
0
Zaprtc seems unsupported, Asterisk in productionenvironment without Digium cards
I'm having a similar issue to this in that the USB ports on the system are ohci based, not uhci, therefore ztdummy will not work. My system is also running the SMP kernel so zaprtc will not work. To me it looks like the only good solution is a hardware timer, even if it's as simple as an x100p card. Kelly PS. If you read Jon's message again, the answer given is not acceptable for
2009 Feb 25
2
SheevaPlug Development Kit
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
2008 Dec 31
2
Friday VUC 12 Noon ET with Kristian Kielhofner: Identifying Asterisk Quality Issues
Happy New Year in advance by a few ticks for the northern hemisphere. Here's the first topic and guest for 2009: In any voice path there are several potential sources of quality problems, ranging from echo to voice dropouts and everything in between. With VoIP systems the potential for quality problems increases dramatically, often times making it very difficult to identify the source of
2005 Mar 19
2
More HEAD wierdness (chan_sip, jitterbuffer/PLC problems)
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