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.conf
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 same
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 concurrent calls this way using the SIPP
program
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 worked. Never went that far
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 you can afford
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 flaming, you could accept
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 the
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
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
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?
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
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
Hello,
I was wondering if anyone out there is successfully running
Asterisk 1.2 svn w/ Centos 4.3. I had an experience over the last two
weeks that has me scratching my head and muttering strange things in the
wee hours of the morning. I am going to try and be as descriptive as my
On Mon, 22 May 2006, Greg Oliver wrote:
Have you tried compiling statically on CentOS 4.2 and running on 4.3?
No. Not really in the plans either. Standard policy w/ Asterisk around
here is to compile on the box it is going to be running on, under the
distro it's running on.
I am assuming
On Mon, 22 May 2006, alist wrote:
Greg,
When I upgraded to 4.3 I experienced problems with some non-asterisk
RPM's that were compiled on earlier versions of CentOS 4. Once they were
recompiled on a fully updated 4.3 system they worked fine. Have you
tried recompiling everything?
We
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 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
channel per quad (I know Asterisk can do
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 for
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 wonder
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
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
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
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
is going
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 this can be
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 buffer it
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 you
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,
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
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
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 clicks
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 authors
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:
On Mon, 21 Nov 2005, Jonathan k. Creasy wrote:
I've thought about doing that as I have a few spare also. I would use
the raq4 I think.
Let me know if you have any trouble with it.
What you may want to do (I have several of these) is see if you can
re-install the new Centos + BlueQuartz
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 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 President
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
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
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 ever
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 in some CASE
statements where
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 that
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 you
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 features:
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 405/410
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 / Vmware
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. You would
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
unsuccessful at
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 6 weeks the box
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 6 weeks the box
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 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 an
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 an
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
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
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 will
On Tue, 28 Jun 2005, Brian West wrote:
Ok I have to get a vote of all the people that are going to come to
Cluecon so we order the beer keg's for the developers board room.
Anyone have any preference? (if you haven't registered for ClueCon
now is the time to register!)
Choices...
yOn Tue, 21 Jun 2005, Matthew Boehm wrote:
We have a TE110P (single span PRI) and are having tons of echo on all calls,
both incoming and outgoing. We didn't have any echo at all yesterday and
nothing in any of the configs has changed.
All of all calls follow this pattern:
Cisco 7960
Hello,
If one is using Dundi, and it returns multiple, weighted routes to
a destination, how is that applied in the dial-plan?
asterisk*CLI dundi lookup 1588XXX
1. 400 IAX2/dundi:[EMAIL PROTECTED]/1588XXX
(EXISTS|NOUNSLCTD|NOCOMUNSLTD)
from 00:30:48:71:26:70, expires in
On Wed, 1 Jun 2005, Michael Stearne wrote:
On 6/1/05, Russell Bryant [EMAIL PROTECTED] wrote:
I am on IRC as drumkilla and also available by email if anyone has any
questions or comments.
Please test and report any issues on the Asterisk issue tracker, even if
it is just a note
On Mon, 30 May 2005, Chris Mason (Lists) wrote:
I thought I saw a Soekris embedded in the Digium booth photos, can you run
Asterisk on one of these? How? I'd be interested in it for a back pbx, given
the reliability. In fact, might want to move my home pbx to this also.
On Sun, 29 May 2005, Kristian Kielhofner wrote:
Dean Collins wrote:
Great booth guys, looks really interesting - can you cull out some of
the more lousy photos though.
Anything else you've seen at the event that's looks interesting?
Dean,
I could cut out some of the more
On Thu, 19 May 2005, Tim Simms wrote:
Have a link?
http://www.ispcon.com
If you are interested in manning the booth, talk to Rick Segrest.
--
Vice President of N2Net, a New Age Consulting Service, Inc. Company
http://www.n2net.net Where everything clicks into place!
On Thu, 19 May 2005, John Todd wrote:
[DELETED]
Err well, that last presentation doesn't have anything to do with
Brian or Greg at the moment, but of course SIP CPE is fairly relevant
to service providers or PBX vendors/consultants working with Asterisk.
Jon,
Glad to see that
On Mon, 9 May 2005, Vikram Rangnekar wrote:
[Deleted]
Can dundi or the switch statement help me get out of this mess ?
Dundi will make this trivial.
--
Vice President of N2Net, a New Age Consulting Service, Inc. Company
http://www.n2net.net Where everything clicks into place!
On Mon, 9 May 2005, David Choo wrote:
Actually, this is whats facing me right now. I think Dundi will resolve the
problem, but I've never really placed it to the test. Anyone tested Dundi?
Best Regards,
I run it in production on CVS-Stable and it works without a problem.
Our usage of it is
Hello,
Haven't had a chance to test this configuration yet, so I can't
really answer my own question, but wanted to get a couple of eyeballs to
look at it and verify that I'm going to get the behavior that I expect out
of it. Basically, I have a TE405P w/ 3 PRI to the Telco and one
P3 1Ghz under Tao Linux 1.0 (2.4 Kenrnel) cvs-stable w/ X101P
--- Results after 66 passes ---
Best: 1.024461 -- Worst: 1.024420 -- Average: 1.024447
And on our new gateway box...
P4 3.0 Ghz under Tao Linux 4.0 (2.6 Kernel) cvs-stable w/ TE405P
--- Results after 106 passes ---
Best: 1.023967 --
On Wed, 4 May 2005, Andrew Kohlsmith wrote:
When the span was 0, I NEVER got that message. I haven't heard any
complaints from the other office mates that use the PRI for voice, but the
error just bothers me.
What is the real difference between 0 and 1 on the span timing?
all that
On Thu, 5 May 2005, Charlie Watts wrote:
Manjit Riat wrote:
Out of curiosity what's the reason? Why would they not sell phones to
asterisk users? Do they not trust asterisk or their phones to work
with each other?
My guess: They don't want to compete with the folks that OEM Polycom
On Thu, 5 May 2005, Colin Anderson wrote:
I like easy questions!
Add to /etc/rc.d:
modprobe wctdm
ztcfg -vv
su yourasteriskuser /usr/sbin/safe_asterisk
where yourasteriskuser is the user that you normally run Asterisk under.
Omit the su yourasteriskuser if you are running Asterisk
On Sun, 1 May 2005, David John Walsh wrote:
what sort of level of PC is required for 300 concurrent calls?
Doing what? Ulaw? That could probably be done on a single P4 2.8 Ghz. If
you want to transcode from Ulaw to something else, you need to scale the
hardware appropriately. Every case is
On Sat, 29 Jan 2005, Greg Boehnlein wrote:
Hello,
I'm looking at building a couple new PRI Gateway boxes using
TE405P cards, and was wondering if anyone has had any experiences (good or
bad) with the Intel SE7210TP1_E motherboards from Intel. General Technics
builds some really nice
On Fri, 22 Apr 2005, Chris Coulthurst wrote:
Is there a specific SIP or IAX phone that truly shines above the rest
where it comes to 'happy' compatibility with Asterisk? I guess I'm
talking about feature sets, like early-dial, off hook call announcing,
conferencing, echo suppression, etc
Hello,
I've been asked to build a couple of Gateway servers for a client
w/ TE405P hardware, and have been looking around at various 1U options.
I've been looking at SuperMicro and Tyan barbones boxes as possible
platforms, but then was directed to Dell's SC1425 by a friend. Short
On Fri, 22 Apr 2005, William Boehlke wrote:
SC1425 is great value but note it does not have high availablility
configurations.
In our opinion, telephony requires dual NICs, dual power supplies and RAID 1
to have any hope of achieving five nines.
William Boehlke
What box would you
On Sun, 17 Apr 2005, Dave Weis wrote:
On Sun, 17 Apr 2005, Greg Boehnlein wrote:
On Thu, 14 Apr 2005, Rod Bacon wrote:
I have been frustrated by a variety of zyxel issues/products and have
found
the best solution for all of them lies in a cylindrical receptacle that
sits
On Wed, 20 Apr 2005, Daniel Salama wrote:
Every once in a while I read messages about people having problems with
certain models of SIP phones, some of them being well known models.
I'm interested in purchasing new SIP phones for my office and wanted to
know which brand/model is most
On Wed, 20 Apr 2005, snacktime wrote:
Not sure. I'm unable to open a ticket with Zoom technical support on the
issue.
I had an interesting experience with the zoom. Their SIP
implementation doesnt' expect to see SIP traffic on the internal lan,
and running * on the internal lan would
On Wed, 20 Apr 2005, Daniel Dziubanski wrote:
Greg,
Are you using AMP?
No.
And If so, you have any tips and tricks on how to easily manage phones via a
amp plugin/fix?
No. The Polycom phones will provision themselves via FTP using XML files.
It probably wouldn't be hard to write. In
On Wed, 20 Apr 2005, Aza wrote:
[DELETED]
I seem to have found a solution to the problem I had with a ZyXEL Prestige
2000 series ATA. It looks like these things just can't cope with NAT no
matter what you do with Asterisk, STUN servers or SIP Proxies. Specifically
the voice quality of the
On Thu, 14 Apr 2005, Rod Bacon wrote:
I have been frustrated by a variety of zyxel issues/products and have found
the best solution for all of them lies in a cylindrical receptacle that sits
beside my desk...
I've had pretty good luck with the Zoom X5V Voice Modem so far. It has a
built in
On Wed, 13 Apr 2005, Kevin P. Fleming wrote:
[ DELETED]
Realistically, how cheaply can you put together a box with a T-1 card
and a channel bank with 24FXS ports (even disregarding G.729
transcoding, which would add to the cost)? $700? $800? more? I can't say
for sure, but if you wanted
On Mon, 11 Apr 2005, Noah Miller wrote:
This this may sound ridiculous, but we've had problems with this when
the
users did not plug the handset cord in completely. 8 out of our 12
employees
made the mistake, as the plug on the IPX00's appears to be all the way
in
when it is
On Mon, 4 Apr 2005, Juergen K. Zick wrote:
Hi Dalon,
I have it running including VMAIL, 3 SIP and one IAX2 account AND OPENPVN ...
Incoming and outgoing connections are OK, both in nat'ed local 192.168.x.x
and external real IP adresses ...
--Juergen
Juergen,
Did you build
On Wed, 23 Mar 2005, Joseph wrote:
My DTFM is not working in current CVS-stable *-1.0.6 and *-1.0.7 but it
works in version 1.0.5 (was working with 1.0.3).
I'm using SPA-3000 and dtmfmode=inband
I noticed the exact same behavior w/ my upgrade to 1.0.7 using Polycom
SPIP phones w/
On Thu, 24 Mar 2005, David Gomillion wrote:
Problem:
Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that
Asterisk can detect and use. It worked in 1.0.5, but has not worked
since. This has been verified on SoundPoint IP 300's and SoundPoint IP
600's.
Workaround:
is that a call was initiated, but you have no idea
how long the call was, or what the rate/billing should be.
On Thu, 24 Mar 2005 00:17:31 -0500 (EST), Greg Boehnlein [EMAIL PROTECTED]
wrote:
On Tue, 22 Mar 2005, Matt wrote:
Hi,
The reason I didn't look into that is it says:
You CAN
On Tue, 22 Mar 2005, Matt wrote:
Hi,
The reason I didn't look into that is it says:
You CAN NOT:
Accounting:
* generate Start or Alive records, which is doable easily for
connected calls, but
If I can't generate a start record... what good is it for CDR recording?
You don't care
On Sun, 20 Mar 2005, Rob Gillan wrote:
Hi,
Having all sorts of troubles getting mysql cdr support under OS X.
Mysql, DBI and DBD all installed and running ok, privileges all set
correctly (I think). Latest asterisk-addons checked out of cvs. Keep
getting error on make install
On Thu, 17 Mar 2005, Christopher Jacob wrote:
[Deleted]
So, the moral of this story
While Polycom may not offer configuration type support for asterisk, they
stand by their hardware. With Cisco you have to shop around to find a decent
deal, and who know how you're going to get support.
On Fri, 18 Mar 2005, Jose R. Ortiz Ubarri wrote:
Jose R. Ortiz Ubarri wrote:
Hi:
I had asterisk with RealTime database working perfectly in a RH 9.0
machine. I used the sip cache so I even had MWI working. The problem
is that I decided to move to Fedora Core 3. I installed the
On Sat, 19 Mar 2005, Ronald Wiplinger wrote:
Russell Bryant wrote:
Hello everyone,
Version 1.0.7 of Asterisk, Zaptel, libpri, and Asterisk-addons has now
been released. Libpri and -addons have not changed, but have been
updated anyway to keep the version numbers consistent. All
On Sat, 19 Mar 2005, Paul Fielding wrote:
Make this another vote for Zap and IAX2 monitoring :)
Paul
Seconded! As cool as FOP is, this looks pretty awesome!
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On Sat, 19 Mar 2005, Scott Gruby wrote:
On Mar 18, 2005, at 10:47 PM, Russell Bryant wrote:
Hello everyone,
Version 1.0.7 of Asterisk, Zaptel, libpri, and Asterisk-addons has now
been released. Libpri and -addons have not changed, but have been
updated anyway to keep the version
On Thu, 10 Mar 2005, Kevin P. Fleming wrote:
http://host-a.starnetworks.us/Members/kpfleming/spring_von/photoalbum_view
Enjoy!
Anyone have pictures from the Heart show? :( My camera phone just wigged
out. I thought I had like 60 pictures right from the stage, but apparently
it didn't save
On Thu, 10 Mar 2005, Kevin P. Fleming wrote:
http://host-a.starnetworks.us/Members/kpfleming/spring_von/photoalbum_view
If you look closely, you'll see me at the booth doing some troubleshooting
for Digium during one of my session breaks. We actually setup an IAX2
connection from the main
On Fri, 21 Jan 2005, Greg Boehnlein wrote:
Hello,
I've got a mixture of SPIP 300 and 500 phones in production for
various clients. I've got the XML settings configured for local
conferencing, but I'm not seeing the expected behavior from the phone when
I attempt to conference two
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