Hi all,
Any good recommendation of IP phone in term of sound quality and
price (reasonable) using with asterisk?
ango
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Check out the snom 300 or the snom 820...
CS
-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Rilawich Ango
Gesendet: Mittwoch, 3. Juni 2009 09:45
An: Asterisk Users Mailing List - Non-Commercial
i quite like the aastra 55i phones, i find the sound quality is better than
the polycom sound stations on loud speaker, and handset quality is perfect.
2009/6/3 Christian Stredicke christian.stredi...@snom.de
Check out the snom 300 or the snom 820...
CS
-Ursprüngliche Nachricht-
On 03/06/09 08:37, Rilawich Ango wrote:
Hi all,
Any good recommendation of IP phone in term of sound quality and
price (reasonable) using with asterisk?
ango
Not sure where you are in the world, or what you really need but I like
the Siemens Gigaset IP DECT phones.
The S685IP is really
Christian Stredicke wrote:
Check out the snom 300 or the snom 820...
Good lord... talk about two extremes... :) The Snom 300 is pretty good,
but the 320 is much better and costs around a *third* of what the Snom
820 does.
Stick with the older model snoms. So far I've seen nothing about
On Wed, Jun 03, 2009 at 08:23:13PM +1000, Rob Hillis wrote:
Christian Stredicke wrote:
Check out the snom 300 or the snom 820...
Good lord... talk about two extremes... :) The Snom 300 is pretty good,
but the 320 is much better and costs around a *third* of what the Snom
820 does.
Hi,
I always get this Error :
-- SIP/1002-09525960AGI Script
/var/lib/asterisk/agi-bin/ast_staging/agi_answered completed,
returning -1
[Jun 2 21:38:17] ERROR[15299]: app_dial.c:1760 dial_exec_full: Could
not stop autoservice on calling channel
is anybody here get the same error, how to
On Wed, 3 Jun 2009, Rob Hillis wrote:
Christian Stredicke wrote:
Check out the snom 300 or the snom 820...
Good lord... talk about two extremes... :) The Snom 300 is pretty good,
but the 320 is much better and costs around a *third* of what the Snom
820 does.
Stick with the older model
Alex Samad wrote:
I have been looking at a snom 300, which seems okay. the display goes a
bit haywire occasionally - not sure why yet.
Are the 320 worth the extra money ?
IMO yes, though it really depends on what you want from the phone.
The Snom 320s handle transfers considerably better
On Wed, 3 Jun 2009, Rob Hillis wrote:
Alex Samad wrote:
I have been looking at a snom 300, which seems okay. the display goes a
bit haywire occasionally - not sure why yet.
Are the 320 worth the extra money ?
IMO yes, though it really depends on what you want from the phone.
The Snom
Thank you for sharing your solution,
However I need to be able to play different files when calling different
numbers dynamicaly from the dial-plan, this solution will still require the
creation of multiple static entry in musiconhold.conf
For example, if it was possible to specify
Kevin P. Fleming wrote:
[delted]
PCI-X (not PCI-E) slots are backwards compatible with PCI slots, by
definition.
Thanks to you both. I knew about the 5v cards, and IIRC the TDM400P
isn't available in 5v.
Some vendors PCI cards:
- don't work period
- work if you have the right rev of
Hi All;
I am looking to start develop an Softphone that has messanger feature (voice
and text, who is online also), anyone can advise for the best link to start
with it, so they have open source for softphone that we can start on it from
there?
http://www.qutecom.org (platform -
It turns out to be a pretty easy change, even potentially transparent.
Gotchas? YMMV, but if you're just doing vanilla asterisk functions, pretty
much no.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex
I have a potential client that currently has a T1 circuit that feeds into an
Adtran 750. Their phone sets are connected to the 24 ports on the 750.
I was wondering if I could take an Asterisk system with a Sangoma A102de in
it and plug the T1 into one port of the A102 and the 750 into the second
Jeff LaCoursiere wrote:
We are still talking about a $175 phone. How about the Polycom IP 320?
$85 at 888voipstore. Can't go wrong with Polycom for voice quality.
True, Polycom's are brilliant for voice quality, but unlike the Snom, a
Polycom /will/ reboot on the drop of a hat /and/ take
Yes, that should work fine, just remember you need a crossover cable to go
from the a102 to the legacy system
2009/6/3 Jim Dickenson dicken...@cfmc.com
I have a potential client that currently has a T1 circuit that feeds into
an
Adtran 750. Their phone sets are connected to the 24 ports on
I will agree with most of what you say about the polycoms, but here is one
thing to help you out. The Polycom 501 has an HTTP configuration interface;
I set up mine thru http://127.0.0.1:8088/asterisk/static/phones where
/var/lib/asterisk/static-http/phones contains the polycom TFTP directory
Michael C. Cambria wrote:
Kevin P. Fleming wrote:
[delted]
PCI-X (not PCI-E) slots are backwards compatible with PCI slots, by
definition.
Thanks to you both. I knew about the 5v cards, and IIRC the TDM400P
isn't available in 5v.
The TDM400P (and all other PCI Digium cards except the
I'm trying to isolate the IP address of inbound calls to my switch over
IAX2. Is the proper way to get that information as follows:
${IAXPEER(IP)}
If the caller was inbound via SIP, this works:
${SIPCHANINFO(PEERIP)}
So I'm looking to return the IP address of the caller via IAX2.
I don't quite understand what you're trying to achieve, but if it's a
firewall wouldn't using something like iptables make more sense and be far
more secure?
Cheers
2009/6/3 Lee Spenadel spena...@gmail.com
I’m trying to isolate the IP address of inbound calls to my switch over
IAX2. Is the
Hello,
I try to connect Qutecom in my Asterisk Server but without success.
What field I need to complete?
Username;
Password;
Realm (asterisk IP Address);
Server (asterisk IP Address);
Proxy (asterisk IP address);
It's correct?
Thanks,
CS
-Mensagem original-
De:
I connected without Proxy filled out.
I used name and not IP address in Realm and Server
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: César Sequeira cesar.m.seque...@gmail.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
I try to connect with name and without Proxy field but when I check again
the Proxy field I have the same name than Realm and Server.
I don't have any information in asterisk's console, so the softphone is not
reaching the server.
-Mensagem original-
De:
On Thu, 4 Jun 2009, Rob Hillis wrote:
Jeff LaCoursiere wrote:
We are still talking about a $175 phone. How about the Polycom IP 320?
$85 at 888voipstore. Can't go wrong with Polycom for voice quality.
True, Polycom's are brilliant for voice quality, but unlike the Snom, a
Polycom
What you say...Allan Oepping (al...@pacificwebworks.com):
I'm not sure if this posting will go to the correct thread or not, as I
am subscribing to make this post, and don't have a message to reply to.
Hose, if this does not end up in the thread can you post in in there?
I am getting the
On 06/03/2009 11:47 AM, Jeff LaCoursiere wrote:
On Thu, 4 Jun 2009, Rob Hillis wrote:
Jeff LaCoursiere wrote:
We are still talking about a $175 phone. How about the Polycom IP 320?
$85 at 888voipstore. Can't go wrong with Polycom for voice quality.
True, Polycom's are brilliant for voice
Hello Philipp and All,
My scenario is a bit different than the one I had explained before. I'm
sorry.
Let's suppose I have someone calling one of my Asterisk clients. This
asterisk client has CFB (Call Forward Busy) activated. The forward number is
a Voice Mail System, however is not a
The only thing I know about the T1 is that it uses wink start signaling.
Wink Start? That is an analog protocol used by DID or EM trunks. If that
is what it is using, then the T1 must be a digitized set of DID analog
trunks. A wink is a hook-switch-flash used to tell the originating side
I personally find the snom phones to be generally ugly and
un-finger-friendly, in terms of reliability and quality, never had any
trouble, good phones all in all, i just can't get past the tacky look and
feel so don't buy them.
2009/6/3 Darrick Hartman dhart...@djhsolutions.com
On 06/03/2009
We have a bunch of SNOM 360's we are not using. I agree they are not
intuitive to the user. They work ok in general. I would part with 15 or so
at an attractive price, one or more,
I like the Grandstream 2000 series. Easy to use, easy to set up, good web
page. Priced nice on the wholesale
I'll put in my $0.02CAD for Polycom. We use the 330s here.
Singer
Geraint Lee wrote:
I personally find the snom phones to be generally ugly and
un-finger-friendly, in terms of reliability and quality, never had
any trouble, good phones all in all, i just can't get past the tacky
look and
I personally find the snom phones to look quite good compared to the
american and chinese brands, might be a european thing though :)
Zoa
Geraint Lee wrote:
I personally find the snom phones to be generally ugly and
un-finger-friendly, in terms of reliability and quality, never had
any
What Lee is trying to do is DETERMINE the IP of a specific caller, then
take an action on that.
Perhaps you misunderstood his use of the word isolate
John Novack
Geraint Lee wrote:
I don't quite understand what you're trying to achieve, but if it's a
firewall wouldn't using something like
Hasn't this religious argument/discussion gone on long enough??
zoach...@securax.org wrote:
I personally find the snom phones to look quite good compared to the
american and chinese brands, might be a european thing though :)
Zoa
Geraint Lee wrote:
I personally find the snom phones
On Tue, Jun 02, 2009 at 04:40:53PM -0500, Erick Perez wrote:
I totally agree with you Jeff, however some of us do not actually sell
viagra over the phone.
This is a campaign to spread a message to the population about the health
prevention steps that should be taken in order to prevent
I think it is a DID trunk. I am having problems getting the clients telco to
tell me much about the T1. For sure 24 analog channels in a single T1.
Would I be able to use this type of T1 with a Sangoma A102de?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Wilton
On Wed, 3 Jun 2009, Tzafrir Cohen wrote:
On Tue, Jun 02, 2009 at 04:40:53PM -0500, Erick Perez wrote:
I totally agree with you Jeff, however some of us do not actually sell
viagra over the phone.
This is a campaign to spread a message to the population about the health
prevention steps that
On Wed, 3 Jun 2009, Jim Dickenson wrote:
I think it is a DID trunk. I am having problems getting the clients telco to
tell me much about the T1. For sure 24 analog channels in a single T1.
Would I be able to use this type of T1 with a Sangoma A102de?
--
Jim Dickenson
On Mon, Jun 01, 2009 at 12:47:22PM -0700, Terry Nathan wrote:
G'afternoon everybody,
I'm having a problem with consistently being able to ring our extensions
from an outside line. I don't have a problem reaching the number, but
during our calls to Background(msg) that I am having a
Hi all,
I am trying to make decisions in my dialplan based on {DIALSTATUS}. I am
creating calls using AMI (rawman with parameters via URL) with
action:Originate. I am using SIP and an outside voip provider for the calls.
If I define the number to call in the Channel parameter (e.g.
you could say it has, and you're right, it probably has :)
but i personally find these threads help make the day pass a little faster
2009/6/3 John Novack jnov...@stromberg-carlson.org
Hasn't this religious argument/discussion gone on long enough??
zoach...@securax.org wrote:
I personally
How did I miss thins gem?
Polycom /will/ reboot on the drop of a hat /and/ take a damned
long time
to do it (~45-60 seconds) In addition, the web interface should
be
taken away and shot - the only real way to configure them is
through (T)FTP.
I didn't say
Edwin Quijada wrote:
Thanks for your helpful reply.
I am not so good in coding.
simply all i need is as follow
When a call comes, goes into an IVR, and then depending on the entry option
it will connect to a remote SQL Database, to check the pre-existed data,
and in the end of
I finally got the provisioning for the T1. It is:
T1 Service Type Robbed-Bit Signaling (RBS), four-wire
Signal Protocol: EM Wink
Line Coding: AMI
Frame Mode: D4
Channels:24
That seems like something that the Sangoma card can support looking at web
sites and such.
I did see
They way I do dialing is with this AMI packet:
Action: Originate
Channel: Local/dial_num...@cfmc_cdi_private
Exten: 1322
Context: default
Priority: 1
Variable: CfMC_ActionID=callE1321
Variable: CfMC_DialInfo=Dahdi/G1/8881231234
Variable: CfMC_RingTimeout=30
ActionID: callE1321
Async: true
And
Hi All,
I am looking for an option in Meetme or similar which will enable to
skip to next priority (a voicemail) if the person in Meetme conference
is alone and if he is there for some time (say 3 minutes)? Any hints on
this? Thanks in advance.
Regards,
--
Kurian Mathew Thayil.
(GPG KeyID:
I have a single server running asterisk 1.6.0.8 with a few sip voip providers
and a tdm card for redundancy. It has a caching name server and the sip
providers
are hard coded in the hosts file.
When the internet connection dies, it fails over to the dahdi channel as it
should, but slowly the sip
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