Hello,
I have just picked up a pair of SoundPoint IP 300 phones for
testing purposes, and they work great. Really good quality units for the
price. I only have two complaints/issues with them;
1. The $40 additional cable for POE to work. That sucks.
2. I can't for the life of me figure
On Fri, 5 Nov 2004, Michael Giagnocavo wrote:
> >> >CoLinux works great for IAX to IAX or SIP to IAX where no Disk Access is
>
> >> >taking place.
> >>
> >> Thanks for clearing that up. I had been using it for IVRs. So if I
> >created a
> >> RAM disk for CoLinux and booted it from there... that
On Fri, 5 Nov 2004, Greg Boehnlein wrote:
> Hello,
> I have just picked up a pair of SoundPoint IP 300 phones for
> testing purposes, and they work great. Really good quality units for the
> price. I only have two complaints/issues with them;
>
> 1. The $40 additiona
On Fri, 12 Nov 2004, Jeremy McNamara wrote:
> Here is a selected portion of the strings output:
>
> [EMAIL PROTECTED] ~]# strings voipgw | grep Mark
> Written by Mark Spencer <[EMAIL PROTECTED]>
> [EMAIL PROTECTED] ~]#strings voipgw | grep CVS
> Asterisk CVS-05/30/03-20:39:27 built by [EMAIL PROT
On Mon, 8 Nov 2004, Rich Adamson wrote:
> > That's not right.
> > New phones come loaded with the current relevant firmware.
> > Upgraded f/w is only available to/from certified resellers.
> > Or look on the wiki for where it is freely available.
>
> The two new 500's that were purchased from a P
On Wed, 17 Nov 2004, Noah Miller wrote:
> I'm ordering some more phones - I have the Polycom IP 500's now and I
> like them. I need some less expensive phones, and I'd like to stay
> with all Polycoms for ease of administration. I've heard, though, that
> the IP 300's don't support PoE even t
On Thu, 18 Nov 2004, Kevin P. Fleming wrote:
> So, I ask again: given the choice between a sub-$100 16-port full-duplex
> 100Mb switch and external power supplies, and an over-$1000 12-port
> switch with internal power supply, which do you think is a better value
> for a small LAN? I can buy $2
On Sat, 20 Nov 2004, Wilson Pickett wrote:
> > Here's my current plan:
> >
> > Sounds like a plan?
>
> You asked for advice, here comes some that few will approve of :)
>
> FWIW I tried to get gnophone running and got no further than you did.
> What struck me though was that I have a very linu
On Fri, 26 Nov 2004, Rich Adamson wrote:
[SNIP]
> What if you have a single port T1/E1 card from digium? No decision to
> be made really; you're going to sync from the other end if it goes
> higher in the hierarchical chain. If the port goes to a box consider
> lower in the chain, then the dista
My copy of VoIP Business Weekly came today and the main article on the
front page is titled: "Penguin to Assault PBX Market". Their premise is
that Linux is driving the adoptation of Open voice standards in the
market, and as such is poised to disrupt the dominance of market leaders.
Third para
On Wed, 1 Dec 2004, Samudra E. Haque wrote:
> I would like to build my newest server based upon Fedora Core 3, and
> load up asterisk. I was all set to do so.. but then I read in Asterisk
> Users Digest, Vol 4, Issue 404:
I think you would be insane to run your production servers on Fedora Core
On Wed, 1 Dec 2004, Kanuri, Seshu (Company IT) wrote:
>
> > Tell me which one can get me access to the LinkSys Linux using SSH?
> > Does Satori has this feature? I am not so concerned with Voice Shaping
>
> > and QOS at this time, but more interested in converting this into a
> > Linux box tha
On Thu, 2 Dec 2004, Eric Rees wrote:
> Where only talking about 100 extensions. That is a lot to hard code by
> hand.
Just use app_queue and define a list of members as the SIP extensions. It
is a lot easier to maintain the queues.conf file than to worry about
adding 100 extensions into your d
On Sat, 21 Aug 2004, Ed Devine wrote:
> I've got a Dual Xeon system Digium Quad T1 and an Idle T1 circuit that
> I can experiment with. I've been wanting to use Redhat with software
> Raid 1 on an Asterisk server.
>
> Has anyone had any experience with software raid and Asterisk? Also, if
> t
On Sat, 21 Aug 2004, Lyle Giese wrote:
> I used software raids and want to get away from them. I really really
> like Microlite's BackupEdge tape backup software. BackupEdge does not
> work with a software raid, only a hardware raid.
>
> The second kicker was that the Promise (or any other) h
On Mon, 23 Aug 2004, William Boehlke wrote:
> Consider hot swappable SCSI RAID 1 instead of IDE. You'll appreciate it
> every couple of years when you lose a disk but the PBX stays up.
I hot swap on my 3Ware IDE RAID card all the time.
--
Vice President of N2Net, a New Age Consulting Servi
On Fri, 27 Aug 2004, Andrew Brown wrote:
>
> When using the callback feature on agents I notice that when the queue calls
> one of the agents and the agent picks up the call they hear nothing until
> pressing the # to accept the call.
>
> Only then does my announcement play back to the agent aft
On Sat, 28 Aug 2004, Michael Welter wrote:
> Since the Cisco 79XX phones preceded the PoE standard, they are
> different--polarity is reversed.
>
> IANAE, but as I understand the PoE devices, there are two types--one
> always applies -48VDC to the brown pair while the other senses (as per
> th
On Sat, 28 Aug 2004, Michael Welter wrote:
> Greg Boehnlein wrote:
> > On Sat, 28 Aug 2004, Michael Welter wrote:
> >
> >
> >>Since the Cisco 79XX phones preceded the PoE standard, they are
> >>different--polarity is reversed.
> >>
> >>
On Wed, 1 Sep 2004, Jay Milk wrote:
> Hello All,
>
> My asterisk installation has now been running for over two months
> without a hitch, and I've decided it's time to move things around a bit.
> It's currently installed on a 2.7GHz Celeron under RH9 installed on a
> 10GB "leftover" drive. Thank
On Wed, 8 Sep 2004, hank smith wrote:
> hello I am fitteling with the astwind-installer-0.1.1.exe asterisk for
> windows and am having trouble getting the thing to connect to the meers
> to download the updates and stuff. I looked at the wiki and set up
> networking and stuff with no success,
On 9 Sep 2004, khurram bhatti wrote:
> Well I wanted to test astwind and consulted * person
> he gave me this comment
> "lord help us all ... why would you want to simulate a linux system on
> top of a windows system in the first place?"
It's not a "simulated linux system". CoLinux is a kernel t
On Wed, 8 Sep 2004, Chris HARIGA wrote:
> I make it work!!
>
> My Astwind is up and running!
> Now is 11:53 PM and I'm going to bed. Tomorrow morning I will post how I fix
> the Ethernet connection.
I bet you followed the following directions! ;)
From: http://www.colinux.org/wiki/index.php/
On Thu, 9 Sep 2004, Chris HARIGA wrote:
[SNIP]
>
>
>
Yep.. Don't you hate the network driver names that these things pick?
Couldn't they just call it "Intel PRO 100" or something simple like "Pro
100"
CoLinux 0.6.2 will allow you to use the name that YOU assing to the
connection i
On Thu, 9 Sep 2004, arsal siddiqui wrote:
> dear khurram,
>
> i need to know the price for x100p. i've emailed convergence.com.pk
> and never get a reply. If you could help me in this regards, i'll be
> greatful. I need to know the price.
>
> send me an email off the list. if you can help me in
On Thu, 9 Sep 2004, Martin Mielke wrote:
> Hi all,
>
> due to the rather big email traffic regarding this issue, I decided to
> publish the script so people can download it at their own risk... :-)
>
> Please, visit:
>
> http://www.leals.com/~mm/asterisk
>
> for further information.
>
>
On Thu, 9 Sep 2004, Matt - Telcom Products wrote:
> Hello,
> Does anyone know how to conference a call on the SNOM 200 phone?
>Whenever I push the cnf/tran button it just hangs up on the active call.
>The manual says you have to push the cnf function key but it doesn't
>appear in the lcd o
On Thu, 9 Sep 2004, hank smith wrote:
> I have a SiS 900 PCI Fast Ethernet Adapter what do I put in there or is that
> what I put in the xml file?
Go read: http://www.colinux.org/wiki/index.php/coLinuxNetworking
Specifically:
"If in doubt, the name of the card can be found in colinux-daemon st
On Thu, 9 Sep 2004, hank smith wrote:
> is there going to be a gui for co linux and astwind?
No. AstWind is just a Debian GNU Linux distribution with a precompiled
Asterisk installation running under a CoLinux kernel.
> I will have to see if either there is going to be a gui or if yasr a screen
On Fri, 10 Sep 2004, Benjamin on Asterisk Mailing Lists wrote:
> However, you could use VMware on an Intel notebook to run both Windoze
> and Linux concurrently. This wouldn't be ideal for a real PBX for
> performance reasons, but since all you are going to use Asterisk for
> is to be a gateway fo
On Sat, 18 Sep 2004, Bill Seddon wrote:
> I've spent a couple of evenings writing a "presence" utility in C# so that a
> window, listing the currently registered SIP phones, can be displayed and a
> user can see who is on the phone and who is not. It uses the Manager API
> and works well, updatin
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.ne
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/asteris
On Thu, 23 Sep 2004, Gary Carr wrote:
> The RPMs had errors for me
>
>
> After installing RPMS and running modprobe zaptel I get
>
> /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol
> register_chrdev_R07a6f6f0
> /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol
> remove_wait_
On Fri, 24 Sep 2004, Greg Boehnlein wrote:
> On Thu, 23 Sep 2004, Gary Carr wrote:
>
> > The RPMs had errors for me
> >
> >
> > After installing RPMS and running modprobe zaptel I get
> >
> > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symb
OTECTED] On Behalf Of Greg Boehnlein
> Sent: Friday, September 24, 2004 8:47 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Asterisk 1.0 RPMS RH73 and RH9
>
> On Fri, 24 Sep 2004, Greg Boehnlein wrote:
>
> > On Thu, 23 Sep 2
On Sat, 25 Sep 2004, Florin Andrei wrote:
> On Fri, 2004-09-24 at 05:47, Greg Boehnlein wrote:
>
> > Anyone else having the problems that Gary is reporting?
>
> Um, well, not really. I'm rebuilding your package on Fedora 2 (kernel
> 2.6) and i had to add a "li
On Sat, 25 Sep 2004, David Troy wrote:
[deleted[
> I had a need for a much simpler proxy than his op_server.pl; to meet my
> need I re-worked and simplified his code. See below for this simplified
> proxy:
>
> http://www.popvox.com/simpleproxy.pl
Hehehehe.. I mentioned this in the Developer
On Mon, 11 Oct 2004, Pete Brown wrote:
> Greg,
> Which kernel are you using? I have two machines at home and the zaptel
> kernel module only runs properly on one of them...
>
> The P-3 box worked...
> kernel-2.4.20-30.9.i686.rpm
RH73 Zaptel Modules are built against:
2.4.20_28.7.i386
RH9 Zapt
On Mon, 11 Oct 2004, Mark Phillips wrote:
> I had a rather unpleasant bait and switch episode with Atacomm today.
> They advertise on their website (and indeed quoted me for) the Snom 200
> for $269 which, when I came to place an order for 15 of them, they
> didn't have but would like to replace w
On Mon, 11 Oct 2004 [EMAIL PROTECTED] wrote:
> Everyone:
>
> We are a Snom authorized reseller and the problem with the Snom 200 is the
> fact that Snom has EOL that model. It is being replaced with the Snom 190.
> The reason there are no Snom 200's is these unit were taken out of
> production ap
On Tue, 12 Oct 2004, Matthew Boehm wrote:
> Switching to DSL would require me to get a phone line, which kinda defeats
> the purpose of doing VoIP. =)
>
> Matthew
Matthew, for unparalelled hackability try the Linksys WRT54GS. It runs
Linux, and supports QOS if you use the Sveasoft (http://www.s
On Tue, 12 Oct 2004, Geoff Nordli wrote:
> Is this where we get to vote for our favorite router software? I choose
> Bering-uClibc
> (http://leaf.sourceforge.net/mod.php?mod=userpage&menu=910&page_id=36). It
> comes with a ton of packages, and you can easily configure it to boot from
> HDD, or C
On Mon, 18 Oct 2004, Claus Lavdal wrote:
> I would be very interested in the script that allow me to use
> saft_asterisk and non-root user on suse 9.1.
>
> Regards Claus
Claus,
Sorry it took a couple of days to get this information back to
you, but I wanted to let you know that you can
On Tue, 19 Oct 2004, Olle E. Johansson wrote:
> > I wish it were in v1.0 as well. Would creating a patch for 1.0 be pretty
> > simple, or do the code changes run deep?
> It will eventuall get into a release.
>
> But please, as a community, we have to refrain from temptation of adding
> new functi
On Thu, 2 Feb 2006, John Todd wrote:
[SNIP]
> 3) Nobody else has thus far taken the bait and made any comments
> about their systems. I appreciate Signate's comments; they seem to be
> the only ones to publicly claim large-scale throughput using Asterisk
> in a public forum. Most other people
On Wed, 29 Jun 2005, Paul Fielding wrote:
> I have indeed already done so - I use G729 quite a bit since I travel alot
> in adverse conditions. Interesting thing is, I can get less choppy audio
> sometimes from my Vonage device using (what I suspect to be) Ulaw, while
> either ulaw or G729 wil
Hello,
I have a client that has a fairly small installation (20 SIP
Phones) that is running Stable. Asterisk appears to be consuming large
quantities of memory, and growing uncontrollably to the point where after
about 6 weeks the box starts to swap itself to death. I've been keeping my
On Wed, 20 Jul 2005, Olle E. Johansson wrote:
> Brian West wrote:
> > Kristian Kielhofner, the lead developer of the AstLinux project, will
> > be speaking at ClueCon. His latest AstLinux Version 0.2.6 is a complete
>
> Kristian will also be speaking at Astricon 2005 in California
> http://www.
On Fri, 22 Jul 2005, Thomas Christie wrote:
> * If you can get the song from this flash animation converted to MP3, then
> it might be good (bad):
> http://www.ebaumsworld.com/flash/spacepeople.html .
http://damin.umlcoop.net/spacepeople.mp3
--
Vice President of N2Net, a New Age Consulting
On Mon, 25 Jul 2005, Terry Moore-Read wrote:
> I'm relatively new to Asterisk and I'm hoping attending Cluecon will be
> a good way to get up to speed on the project and hear about what others
> are doing with it.
>
> We currently use a Cisco IP phone system at my office, although I just
> added
On Mon, 25 Jul 2005, Terry Moore-Read wrote:
> I'm relatively new to Asterisk and I'm hoping attending Cluecon will be
> a good way to get up to speed on the project and hear about what others
> are doing with it.
>
> We currently use a Cisco IP phone system at my office, although I just
> added
On Mon, 25 Jul 2005, Brian West wrote:
> I'm going to be speaking about how to use valgrind, gdb and strace to
> help debug issues... it can be applied to more than just asterisk.
Given the following from one of my Client's boxes...
pbx*CLI> show memory summary
[DELETED]
7084 bytes in
On Mon, 25 Jul 2005, Brian West wrote:
> I'm going to be speaking about how to use valgrind, gdb and strace to
> help debug issues... it can be applied to more than just asterisk.
Given the following from one of my Client's boxes...
pbx*CLI> show memory summary
[DELETED]
7084 bytes in
On Wed, 20 Jul 2005, Greg Boehnlein wrote:
> Hello,
> I have a client that has a fairly small installation (20 SIP
> Phones) that is running Stable. Asterisk appears to be consuming large
> quantities of memory, and growing uncontrollably to the point where after
> about
On Wed, 20 Jul 2005, Greg Boehnlein wrote:
> Hello,
> I have a client that has a fairly small installation (20 SIP
> Phones) that is running Stable. Asterisk appears to be consuming large
> quantities of memory, and growing uncontrollably to the point where after
> about
On Thu, 11 Aug 2005, Trevor Peirce wrote:
> Ing. Marlo R. Beltran G wrote:
>
> > I am about to buy ip pbx asterisk system but what ip phones do you
> > recommend? Are polycom ip all functional with the ip pbx system???
>
> We just got a Polycom IP501 for testing and have thus far been
> unsuccess
On Fri, 12 Aug 2005, Matt Florell wrote:
> Short answer: NO
>
> Long answer: you have to send it to Digium for them to do an upgrade,
> they don't have an official process for this yet and won't give you a
> price, I have called and asked them many times. They also mention
> upgrades from your 40
On Fri, 12 Aug 2005, Bruce Leetch wrote:
> Am I banging my head against at Windows/VMware/Linux/Asterisk
> incompatibility? Or can this work and I'm just doing something stupid
> (always a possibility with me).
It's not going to work. Vmware presents a complete Virtual PC, so unless
EMC / Vmwar
On Fri, 12 Aug 2005, Tom Rymes wrote:
> > VMWare is a virtual machine and has nothing to do with the physical
> > layout of the box (which is why you can migrate vmware images
> > across machines for example).
> >
> > If you want to run Asterisk under Linux setup a box to run it.
>
> Agreed.
On Wed, 28 Dec 2005, Goran Skular wrote:
> Hi,
>
> Anybody have some experience and did some testing with ipVolution E1/T1
> cards?
I chalked these up to VaporWare...
--
Vice President of N2Net, a New Age Consulting Service, Inc. Company
http://www.n2net.net Where everything clic
Hello,
I know that Mog was trying to get the bug-tracker cleaned up as
the number of bugs has increased substantially over the past few months. I
figured that I would do my part to bring attention to a couple of bugs
that are interesting and have some wide reaching impact. That being sai
Hey there,
Just wanted to drop a line and let people know that I'll be
heading to San Francisco for O'Reilly's Etel. If you are interested in
attending, there are some free passes floating around. If anyone is
interested in getting together for a beer, let me know!
Info on the conferenc
On Wed, 11 Jan 2006, stevanus wrote:
> Hi,
>
> As I've dealt with asterisk 1.0.10 successfully, I wonder what the
> benefit I will get from upgrading to 1.2.1..
> [Of course I know there're lot of new interesting stuffs in 1.2.1, but
> are they stable already?]
>
> Does the 1.2.1 need more re
On Thu, 12 Jan 2006, Steven wrote:
> If it is a toll free number, it may be related to
> http://bugs.digium.com/view.php?id=5266 .
You may also want to investigate:
http://bugs.digium.com/view.php?id=5970
as well as:
http://bugs.digium.com/view.php?id=6027
There are several people that are p
On Sat, 14 Jan 2006, Brian Capouch wrote:
> Rich Adamson wrote:
> >
> >
> > Since there does not seem to be anyone else complaining about the same
> > problem, there must be something in your config that is causing it.
> > Without specific copy/paste samples of what you've configured, no one
>
On Mon, 16 Jan 2006, Koopmann, Jan-Peter wrote:
> On Sunday, January 15, 2006 12:21 AM Tony Mountifield wrote:
>
> > In article <[EMAIL PROTECTED]>,
> > Pisac <[EMAIL PROTECTED]> wrote:
> >> I've found something here: http://bugs.digium.com/view.php?id=5374
> >>
> >> but I don't understand how t
On Tue, 17 Jan 2006, Kib Eki wrote:
> Hi Karsten,
>
> I have the same problem. MusicOnHold sounds awful. The PRE-1e does not have
> this
> problem. I have two identical systems (hard-/software). One system has the
> problem the other does not. I thought i could be timing problem or interrupt
On Tue, 17 Jan 2006, Jean-Michel Hiver wrote:
> Dov Bigio a ?crit :
>
> >Ok.. but I don't use Real Time at all.
> >I just use cdr_mysql. It would be smarter if it simply ignored MySQL outages
> >or at least just logged, but without stopping.
> >
> >
> What would be even nicer would be for * to
On Tue, 17 Jan 2006, Alexander Lopez wrote:
> Buffer! For how long? How big of a buffer? If I can buffer 10-20 calls
>that might work if I have a light use PBX but 100-2000 buffered calls
>may not hold a busy PBX. OK so make it configurable, With any luck you
>won't know how much to put so yo
On Tue, 17 Jan 2006 [EMAIL PROTECTED] wrote:
> Buffers don't have to be in memory. My suggestion on the solution would be
> to buffer the CDR info into a backup file based database (configurable
> filename/path) on the local filesystem (or NFS mounted system for
> redundancy) and then when the SQL
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
On Sun, 22 Jan 2006, Steve Totaro wrote:
> I have a T3 coming from my carrier. From there I want to use an Adtran
> mx2800 T1 Mux to break the T3 into 28 T1/PRI which feed into seven quad
> T1/PRI equipped servers.
>
> Everything seems very straight forward with the exception of the D
> channels
On Sun, 22 Jan 2006, Steve Totaro wrote:
> Thanks for some answers, that is what I thought.
>
> Asterisk is NFAS capable so I am looking at seven D channels on the T3 I
> guess. I don't want to put a D channel on each T1 or I will lose
> several channels that could be used for calls.
>
> I
On Mon, 23 Jan 2006, Kevin P. Fleming wrote:
> Greg Boehnlein wrote:
>
> (Steve Totaro wrote:)
>
> >>What I would really like to do is have one D channel coming in on the T3
> >>and have it split between each of the T1/PRI or even better one D
> >>channe
On Sat, 13 Aug 2005, Lull, Rick wrote:
> This is one of the main reasons that AstWind has stagnated. The timing
> granularity of the virtual machines is not acceptable for doing anything
>
> IO related.
>
> Just since I am curious, what version of VMWare did you use and what kind of
> box where
On Thu, 18 Aug 2005, Holden Hao wrote:
> > I don't mind buying an appliance to get something solid but IP Cop just
> > looks better than he appliances I see out there.
>
> Astaro has been getting good reviews from Linux World. They have an
> appliance solution or a self-install solution. It fea
On Thu, 18 Aug 2005, Hadar Pedhazur wrote:
> First, many thanks to Greg Boehnlein for his patch to chan_agent.c
> for adding a "preackannounce" option.
>
> I am running CVS HEAD from 2005/07/31, and the patch failed in a
> few hunks, since the code was refactored to add
On Fri, 26 Aug 2005, Hadar Pedhazur wrote:
> Hmmm. I am often surprised when I don't get a response to a post that I
> think would interest at least _one_ person in the community. This one
> surprised me a little more, since I offered some code ;-).
>
> This morning, I just got a bounce notice
On Mon, 29 Aug 2005, Kristian Kielhofner wrote:
> Matt,
>
> It sure is! You should be testing it! :) Test it and see, but 1.2
> will be "STABLE" pretty soon here...
No. No NO NO NO NO! :)
1.2 will never be called Stable, based on the controversy surrounding the
naming moniker. Do you
Hello,
I recently updated some Polycom 501 phones to the new 1.6.7
firmware, and have lost the ability to do "One Touch" voicemail access via
the messages button.
I've verified that I have the correct XML tags set in the phone config,
I.E.:
msg.bypassInstantMessage="1"
mwi msg.mwi.1.su
On Sun, 30 Jul 2006, Peter Johnson wrote:
> How about up.oneTouchVoiceMail="1" in your sip.cfg
>
> Peter
Ahhh... that tag wasn't in my config generator script, so I must have set
it by hand in the old ones. That does the trick!
I owe you a beer!
--
Vice President of N2Net, a New Age Cons
On Sat, 29 Jul 2006, Tom Vile wrote:
> Did you look on the site?
>
> http://www.4psa.com/products/voipnow/demo.php
Ughh.. it's PLESK! Looks like the entire thing is written w/ the PLESK
user API. NEXT!
--
Vice President of N2Net, a New Age Consulting Service, Inc. Company
http://
On Sat, 29 Jul 2006, Douglas Garstang wrote:
> You have a config generator script for the Polycom XML files? What did you
> build that with?
Bash scripts and some sed logic. It isn't pretty, but it works.
# This creates the actual SED script that we use to modify the template
echo "s/reg.1.disp
On Fri, 4 Aug 2006, Steve Totaro wrote:
> > If your M13 is coming up clean, I'd double check the continuity to
> > those ports from the DSX connections out to the patch panel.
>
> That was exactly the issue. The amphenol cable was loose on one end.
> What is the best way to fasten these things?
On Tue, 8 Aug 2006, mitcheloc wrote:
> >
> > On Sat, 29 Jul 2006, Tom Vile wrote:
> >
> > > Did you look on the site?
> > >
> > > http://www.4psa.com/products/voipnow/demo.php
> >
> > Ughh.. it's PLESK! Looks like the entire thing is written w/ the PLESK
> > user API. NEXT!
>
> Instead of flamin
On Tue, 8 Aug 2006, Matthew Warren wrote:
> Yes it is an addon of Plesk, thats stating the obvious. But while your
> complaining about people writing stuff to use what are you doing. If your
> not a developer don't critisize the developers. I see nothing more than you
> displaying that you are
On Fri, 11 Aug 2006, Senad Jordanovic wrote:
> [EMAIL PROTECTED] wrote:
> > Hi
> >
> > After a month or so using Asterisk we've had or first downtime period
> > due to a faulty RAM chip on the server, so we're starting to think
> > about the possible high-availability solutions.
>
> Hi
>
> If y
On Sat, 26 Aug 2006, Kelvin Williams wrote:
> If Asterisk was used to set up and tear down calls, and using canreinvite
> allowing the RTP to pass from end-point to end-point, how many calls could
> Asterisk handle at once?
I've pushed over 1,000 concurrent calls this way using the SIPP program
On Mon, 28 Aug 2006, Andrew Kohlsmith wrote:
> On Monday 28 August 2006 13:02, Greg Boehnlein wrote:
> > I've pushed over 1,000 concurrent calls this way using the SIPP program
> > for SIP performance testing. There was some tuning that needed to be done,
> > but it work
On Tue, 29 Aug 2006, Nick Hoffman wrote:
> On Tue August 29 2006 04:39, Greg Boehnlein <[EMAIL PROTECTED]> wrote:
> > On Mon, 28 Aug 2006, Andrew Kohlsmith wrote:
> > > On Monday 28 August 2006 13:02, Greg Boehnlein wrote:
> > > > I've pushed over 1,000
On Tue, 29 Aug 2006, Benjamin Lawetz wrote:
> Hello all,
>
> we're having an issue with DTMFs being sent to Sipura's. Calls are
> originating from a Cisco AS5300 being sent to asterisk which in turn sends
> it to the Sipura. Connected to the Sipura is a legacy PBX (or actually shows
> the s
On Sat, 2 Sep 2006, Diego Quintana Cruz wrote:
> Hi everybody,
> I'm trying to load-test my Asterisk PBX using SIPP, but I always
> getting errors, I followed the instructions given in [1] which mainly
> was to create the user sipp in sip.conf and the dialing plan for his
> context in extensions.c
Anyone heading out to Ohio Linuxfest? Digium is going to have a booth
there. I would appreciate people checking this out and digging it if they
feel it would be appropriate. Nothing like a FREE Open Source conference
with live penguins! :)
http://digg.com/linux_unix/Ohio_Linux_Fest_2006
--
Hello,
Rumor has it that the TDM2400 series cards will be available in
the next week or so. If you are a distributor that has pricing /
availability information, please contact me offlist. I am putting together
a solution for a client that will require a TDM2420E (8 Port FXS w/ Echo
Can
On Tue, 11 Oct 2005, Tom wrote:
>
>
> I currently have a single PRI however we are getting a second PRI, and the
> provider (qwest) wants to know if our PBX supports GSAS (they say its a
> redundant d-channel technology but searching on google for GSAS reveals less
> than nothing). I've set som
On Wed, 2 Nov 2005, Matt Darnell wrote:
> Well that didn't take long!
>
> He was a really nice guyI bet it would be a blast to go have a beer with
> him.
>
> We met him at the Internet Telephony Expo.
Read his bio on Rotten.Com. I'm surprised to see him posing with Women.
--
Vice Pres
On Fri, 18 Nov 2005 [EMAIL PROTECTED] wrote:
> I couldn't find his bio on rotten.com
http://www.rotten.com/library/bio/hackers/captain-crunch/
--
Vice President of N2Net, a New Age Consulting Service, Inc. Company
http://www.n2net.net Where everything clicks into place!
On Tue, 22 Nov 2005, Lenz wrote:
> I also have never found anybody running an Asterisk system using app_icd.
> Maybe app_queue is now after all flexible enough to be used in most cases.
> Anybody else using different apps for Asterisk call centre applications?
I suspect that since the author
On Mon, 21 Nov 2005, Bob Knight wrote:
> Just pulled a v1-2 onto a system that was running a v1-0.
>
> Zaptel and libpri, build and install just fine.
> Building asterisk is fine.
> But when I try to do a make install on asterisk, it goes into an
> infinite loop doing on .depend doing: build_tool
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