Hi List.
I've used 3x Snom 200 in a hotel environment for a year, all of them
have failed at some point (1 went up in smoke, 1 handset failed, 1
display failed). I have 110 grandstream phones there of which only 3
have failed (whereof 2 due to brutal abuse), and in another company I
have 75
Hi.
It matters hugely which version of firmware you're running on the phone.
The Cisco pages don't help you very much with this though.
I transformed 12 phones the other day from SCCP 3.something to SIP. Had
to upgrade to SIP 5 first and then SIP 7.
Well worth it, these are highquality SIP
At least in Call Manager 4 you can setup a SIP trunk between CCM and *.
Details are in the wiki.
Regards,
Maron
Mohamed Farid wrote:
Dear All :
We need to use the Conference Room Capability from Asterisk to use it
with our IPT Solution which based on Cisco Call Manager..
Also we need to use
Hello List.
I am setting up asterisk as a central dialplan, voicemail and conference
solution, connected to 12 Cisco 1760 Routers running Call Manager
Express IOS distributed around the world. This is all done over VPN.
These routers all have PSTN access in their respective country.
So far
Answering myself here, just thought of that I didn't put my version info
in there.
I'm running asterisk 1.05 on 2.6.9-gentoo-r13
Regards,
Maron Kristofersson
Maron Kristófersson wrote:
Hello List.
I am setting up asterisk as a central dialplan, voicemail and conference
solution, connected to 12
I'm seeing the same problem here, all SIP calls go to the default context.
Kelvin Chua wrote:
this is something i just recently noticed.
have you found any info on how to manage incoming calls through
chan_h323? it doesn't seem to match any entity you define, it always
uses the default context...
Kristofersson, Staffanstorp, Sweden
Tobias Jönsson wrote:
On Wed, 25 Aug 2004, Maron Kristófersson wrote:
Hmm, that raises a lot of questions for the script... How many
contexts do you have? Do they include each other. Is there any kind
of rule around the extensions... etc.
All the extensions
Hi Tobias!
Have you looked at http://voip-info.org/wiki-Asterisk+Dialplan+Patterns,
The extensions _123X0 would handle all the extensions below except for
the six digit extension.
Regards,
Maron Kristofersson, Staffanstorp SE
Tobias Jönsson wrote:
Does anyone have a program that could be used to
by strict rules or a
script), then approaching the task in chunks would be both faster and
probably less error phrone.
I have done a similar thing at one location, but that was a 500 line
extensions.conf.
Regards,
Maron Kristófersson, Staffanstorp, Sweden
Tobias Jönsson wrote:
On Wed, 25 Aug 2004
For those on a low budget compex (http://cpx.com) has some very low cost
switches that support QoS.
http://www.cpx.com/proddetail.asp?c=Switchese=109
Bought a few of these myself, seem to work well. They are only
manageable through an rs-232 console though, and don't have some other
features
I was even considering going further and writing a crossplatform or a
webapp for configuring. However I was thinking if someone has written
some notes on the config file specification that could save a lot of
time. I have no intention of competing with gsconfigure since I think
it's an
I'm very close to making this work in the crossover wine emulator on
linux. Currently I am getting an error when trying to download the
config directly from an ip address. See attached snapshot for details.
When installing the program I had to choose win2k as the emulated OS.
Regards,
Maron
and the attachment is here :)
Maron Kristófersson wrote:
I'm very close to making this work in the crossover wine emulator on
linux. Currently I am getting an error when trying to download the
config directly from an ip address. See attached snapshot for details.
When installing the program I
Also, I need a Linux tool to splice a series of gsm audio
clips together in order to use one 'get_data' instead of multiple
cat sound1.gsm target.gsm
cat sound2.gsm target.gsm
...
Regards,
Maron
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I had the exact same problem while installing a PRI yesterday. What I
did was changing to letting the telco handle the timing( digit nr. 2 in
span). However that didn't do the trick until I ran ztcfg -s and then
ztcfg, and then the PRI resynced and has been fine since then. However
it could
If you wan't to create a ringtone with makering.pl for firmware 1.0.50,
be sure to create it as ring.bin and then rename it to ring1.bin /
ring2.bin or ring3.bin. This seems to be the only change between the
format from 1.0.4.68.
Regards,
Maron
___
Hi!
I can highly recommend them, good quality and they seem to have very
competitive pricing as well.
Regards,
Maron Kristofersson
Iceland
Matt wrote:
Hi everyone
I'm interested in using the Telappliant/voip-talk offering as an alternative to my DDI
analog problem. (see [Asterisk-Users]
I think that's related to the 2.4 kernels, as they look at the HT CPU as
2 CPU's. I'm running Asterisk on Gentoo running kernel 2.6.5 and I'm
not having any problems.
Maron
Chris Bond wrote:
Are they any issues still with hyperthreading processors, I've read and been
told by a few people to
Not using any cards at the moment here, However I will have an E100 card
installed later this week.
Chris Bond wrote:
What cards you using currently I've just got one FXO card that I need to use
with it.
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: 01 June
of the format ( and a success
story) but I have tried it and it does not seem to work.
Maron
Duane wrote:
Maron Kristófersson wrote:
I guess a http scraper would be a legal way of mass-configuration.
Anybody created such a script and is willing to share?
Not my script, and I can't remember where I grabbed
Sorry, didn't see the attachment up until now, will try it out later
today... Thanks a lot.
Duane wrote:
Maron Kristófersson wrote:
I guess a http scraper would be a legal way of mass-configuration.
Anybody created such a script and is willing to share?
Not my script, and I can't remember
Running asterisk on gentoo 2004.0, kernel 2.6.5, 2.8 Ghz hyperthreading
CPU 1G RAM. I decided to use kernel 2.6 after reading about problems
with hyperthreading and asterisk in 2.4 on this list. So far I've only
connected to VOIP service providers and everything has been working very
well. I
Hello!
I've been reading through the archives on this list for the last 8-10
months. There are some reports on success with tftp autoconfiguration
with a given cfg.txt format but really vague. Has anybody successfully
done this without using GAPS, or has anybody got a correctly formatted
way of mass-configuration.
Anybody created such a script and is willing to share?
Maron
Stephen R. Besch wrote:
Maron Kristófersson wrote:
Hello!
I've been reading through the archives on this list for the last 8-10
months. There are some reports on success with tftp autoconfiguration
Well, I've been testing the service from Iceland for the last few days,
calling both UK based numbers and some numbers in Sweden, Can't say that
I've had any problems, actually, the phone calls all had excellent
quality, except one, but at that time I was downloading a linux .iso,
without QoS
Yes, the plan is to upgrade the wireless bridges to handle QoS.
Mike Machado wrote:
Would like wireless link contain only voice traffic? If not, it would
probably be a good idea to put some sort of minimum bandwidth guarantee
and prioritization.
I have * running over wireless with such bandwidth
wondering if there is any isdn based solution since there is a
possibillity of another staff member going abroad, and then I would like
to have 2 numbers, 1 for each user.
Is the phone on the user end software-based, ip-telephone or an analog
telephone?
Best regards,
Maron Kristófersson
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