The two phones belong to context phones and the two extensions are in context
internal. In context phones you need to include = internal so that context
phones knows about those extensions. Or put the two extensions in context
phones and not context internal.
--
Jim Dickenson
On Thu, 11 Mar 2010, Tommy Botten Jensen wrote:
Hi
I'm looking for a good phone SIP phone for conference room use.
My requirements are in order:
* Speaker quality
* External microphone support.
* Provisioning support / asterisk compatibility.
Does anyone have any experience on this
On 11 Mar 2010, at 12:43, Gordon Henderson wrote:
On Thu, 11 Mar 2010, Tommy Botten Jensen wrote:
Hi
I'm looking for a good phone SIP phone for conference room use.
My requirements are in order:
* Speaker quality
* External microphone support.
* Provisioning support / asterisk
On Thu, 11 Mar 2010 12:56:05 +0100, Tommy Botten Jensen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
Hi
I'm looking for a good phone SIP phone for conference room use.
My requirements are in order:
* Speaker quality
* External microphone support.
* Provisioning support / asterisk
I'd go with Polycom all the way. We have a number of different types of
phones in use, or that we've worked with, including Grandstream, SIpura and
Atacom, and the quality difference with the Polycom phones is astounding.
On 10/29/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
My apologies to
Michael Graves wrote:
On Mon, 29 Oct 2007 15:01:38 -0400, [EMAIL PROTECTED] wrote:
Well, just general office use. They are a real-state construction
company, so the phones will get some heavy use since most of the
phones are going to sales associates.
Now, one of the things we are most
My apologies to the list for not having entered a subject line in the
email.
Thanks
On Oct 29, 2007, at 1:42 PM, [EMAIL PROTECTED] wrote:
Hi all,
We have a client that needs to setup about 80 desk phones (about 50
in one location and about another 30 in 5 different locations). Which
Well, just general office use. They are a real-state construction
company, so the phones will get some heavy use since most of the
phones are going to sales associates.
Now, one of the things we are most interested in are:
1) Asterisk compatibility
2) Mass provisioning
3) Remote management
4)
On Mon, 29 Oct 2007 15:01:38 -0400, [EMAIL PROTECTED] wrote:
Well, just general office use. They are a real-state construction
company, so the phones will get some heavy use since most of the
phones are going to sales associates.
Now, one of the things we are most interested in are:
1)
snom320, snom360 and snom370 are supporting 12 different SIP identities.
Regards,
Sven
On Friday 16 March 2007 10:57, younss azzayani wrote:
Hi every body,
can someone please tell me about a SIP phone that support more than 10
extension (free or not free ;) ) wich will be used in my company,
ok thank you :) i'll look for this
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Try the Snom 360. The softphone version of it (a free demo) has 12 lines (I
presume the real thing has the same). You can find the softphone at
http://www.snom.com/download/snom360-5.3.exe
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of younss
azzayani
Rob Schall wrote:
This might sound like an odd question but here it is anyways...
We currently have Polycom 501 phones. We have Asterisk with
Realtime/MySQL running for our SIP/Voicemail/Extensions. When one phone
dials another, the receiving end does in fact see the callers ID. But...
check out rpid
Mark Johnson wrote:
Rob Schall wrote:
This might sound like an odd question but here it is anyways...
We currently have Polycom 501 phones. We have Asterisk with
Realtime/MySQL running for our SIP/Voicemail/Extensions. When one phone
dials another, the receiving end does
I set both the trustrpid and sendrpid to yes, but the calling phone
still doesn't show the caller id of the person they are calling.
Jason Fuermann wrote:
check out rpid
Mark Johnson wrote:
Rob Schall wrote:
This might sound like an odd question but here it is anyways...
We
try actually setting the rpid in the dialplan using
sipcalledrpid(name,number)
Rob Schall wrote:
I set both the trustrpid and sendrpid to yes, but the calling phone
still doesn't show the caller id of the person they are calling.
Jason Fuermann wrote:
check out rpid
Mark Johnson wrote:
I have some ethernet cable splitters I'm not using any more. They go in
pairs, one plugs into the wall socket in the office, the other plugs
into the other end of the same cable in the server room. each gives two
female ethernet sockets that represent two separate network cables, each
using
I was wondering if we could uplink small switches to the wall data ports
to
the switch, and connect the additional SIP phones to them to get them
connectivity to Asterisk?
Yes, we do it and it works fine, as long as you don't cascade more than 3
switches between two devices your latency
Shouldn't be a problem as long as you're using switches and not hubs, and the network is atleast 100Mb.-brandonOn 7/31/06, T. Shaw
[EMAIL PROTECTED] wrote:I have a client that is looking for a least cost solutionof providing
more SIP phones to an existing asterisk setup.The Issue is this: He has
Should be no problems at all. But keep in mind that you're getting what you pay for. There have been a few posts recently regarding issues with a Polycom 501 that turned out to be (partially) related to the networking equipment used. Make sure you get a good quality switch (and NOT a hub, as
2006/7/27, Nik Engel [EMAIL PROTECTED]:
User logs into any phone and the settings of the phone are always thesame. Meaning individual keyassignement is always the same.Hi,Do you mean :1. Without user logins, phones are unusable ? Or do you plan to offer default services (local calls for instance)
Hi Nik,
I like the Grandstream Budge Tone 102 VoIP Phones which you can find here:
http://www.voipsupply.com/product_info.php?products_id=40
and here:
http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-31609737728.htm
Also, the GXP-2000 is a very popular model too, although once
Hi !
Also, the GXP-2000 is a very popular model too, although once you
consider the capabilities of Asterisk the only real advantage this unit
has over the others (even in an office environment), is the Power over
Ethernet (PoE) feature:
which is supported be Snoom as well.
Anyway I would
On 7/27/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote:
Hello
Just use Snom or grandstream phones. They can be provisioned very easily via
HTTP.
You just setup a config URL on the phones, and they get their configurations
from there.
If you want to get more advanced, they can send along their
Bill Maidment wrote:
Hi
I've been using Asterisk for a while now with the TDM400 and it seems to
be working fine. I'm using version 1.2.2 and I've struck a problem when
I added a Budge Tone 100 SIP phone to the network. The phone rings when
calls come in and I can make calls but in all cases
The ClipComm CP101B has WIFI capability, they make
a bluetooth mini PCMCIA card for it. Or you can pick up a VXI BlueParrot
BP200, it's about the least expensive BlueTooth headset/base station I know of
that isn't junk, and it works well with just about any phone.
Downside is it does not
A lot of my customers have people who are in the office most of the time but
occasionally
wish to work from home. So they may have a sip
phone which is extension 208 in the office. When they work from home they
can of course
plug in a sip phone into their broadband
connection and work
Rana Dutt wrote:
Has anyone successfully had a SIP phone fail over from Asterisk Server A to Server B using DNS SRV?
If so, which phone worked for you? I'm assuming you set up your DNS SRV records
so that the IP
adresses of A and B are associated with the same name, and both
servers have
Has anyone successfully had a SIP phone fail over from Asterisk
Server A to Server B using DNS SRV?
Definitely we have been doing this for quite a while.
If so, which phone worked for you? I'm assuming you set up your
DNS SRV records so that the IP
addresses of A and B
Try www.SIPphone.com or www.terracall.com
Seshu
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ellafi
FituriSent: Tuesday, July 05, 2005 2:07 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
[Asterisk-Users] SIP PHONE
Hi All,
I just got
I use the polycom sip 500's with *. They are great. It also
has a services buttons for XML services. I haven't looked in to
using it just yet.
Peace out,
BrianOn 7/9/05, Mike Clark [EMAIL PROTECTED] wrote:
Pavel Jezek wrote: thank you Brian, but seems, that Polycom phones are not very good option
Brian Roy wrote:
On 7/9/05, Dan Perik [EMAIL PROTECTED] wrote:
PJ,
You should check out the Polycom 500/501/600. I'm quite sure it has all
that (although I don't use all of what you listed).
IIRC, the 500's browser is crippled. I think you have to go up to the
600
PJ,
You should check out the Polycom 500/501/600. I'm quite sure it has all
that (although I don't use all of what you listed).
- Dan
Pavel Jezek wrote:
Still looking for cheaper (under $250,-) alternative to cisco 7940
with features needed for corporate use, mainly:
- shared phone book
On 7/9/05, Dan Perik [EMAIL PROTECTED] wrote:
PJ,
You should check out the Polycom 500/501/600. I'm quite sure it has all
that (although I don't use all of what you listed).
IIRC, the 500's browser is crippled. I think you have to go up to the
600 to get that functionality.
-Brian
thank you Brian,
but seems, that Polycom phones are not very good option for general
corporate use and even not for use with asterisk (* explicitly
unsupported!), look:
from voipsupply.com:
Please Note: Polycom phones are not supported under Asterisk Open Source
PBX.
from Polycom FAQ:
Can
Pavel Jezek wrote:
thank you Brian,
but seems, that Polycom phones are not very good option for general
corporate use and even not for use with asterisk (* explicitly
unsupported!), look:
from voipsupply.com:
Please Note: Polycom phones are not supported under Asterisk Open
Source PBX.
I believe that [EMAIL PROTECTED] will create cisco files.
- Original Message -
From: Max Clark [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, June 28, 2005 4:32 PM
Subject: [Asterisk-Users] SIP Phone Config
We use a mix of Cisco/Polycom phones. I just keep generic template files and
make a copy of them for each new phone. It would be fairly trivial to put a
webserver on the same server with the tftpd. In php or perl it would be fairly
trivial to make a webpage that copied template files,
Snom make good gear. Not cheap though.
PaulH
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson
Sent: Thursday, 19 May 2005 3:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Phone
On Wed, 18 May 2005 22:29:40 -0500, Kristian Kielhofner wrote:
Ariel,
It's probably not a good idea to reccomend the IP 500/300 anymore.
They are being phased out by Polycom because they (and the IP 300) only
have 2mb of flash, and Polycom is looking to standardize on 4mb for
their
Kristian Kielhofner wrote:
It's probably not a good idea to reccomend the IP 500/300
anymore.
They are being phased out by Polycom because they (and the IP 300)
only have 2mb of flash, and Polycom is looking to standardize on 4mb
for their firmware (which the IP 600 has had since day
Discussion
Subject: Re: [Asterisk-Users] SIP Phone Recommendations?
On Wed, 18 May 2005 22:29:40 -0500, Kristian Kielhofner wrote:
Ariel,
It's probably not a good idea to reccomend the IP 500/300 anymore.
They are being phased out by Polycom because they (and the IP 300) only
have 2mb of flash
This may not be what you are looking for, but I have had pretty good success
with the X-Lite phone. I am not sure if you are looking for software SIP
phones.
Thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Mensel
Sent: Wednesday, May 18,
The receptionist phone is going to be a hard one. We use Flash Operator
Panel. Works great.
Now about the phones for all around great phone we are using the Polycom
IP-500 which is in my view one of the top of the line phones.
For el cheapo well we are using one that is yes cheap but also pretty
Ariel Batista wrote:
The receptionist phone is going to be a hard one. We use Flash Operator
Panel. Works great.
Now about the phones for all around great phone we are using the Polycom
IP-500 which is in my view one of the top of the line phones.
For el cheapo well we are using one that is yes
On Wed, 18 May 2005, John Mensel wrote:
Hi all. I'm in the process of putting together a new Asterisk system as a
proof-of-concept, and wanted to see which SIP phones all of you had the best
luck using with Asterisk. I've just come off a very trying experience with
some Cisco 7960s, and
The Sipura SPA-841 has everything except memory buttons but has a directory
and speeddials so I don't think that's so important. Cheap and well made,
although if the speaker phone is very important, get Polycoms, it's the
business they are best in.
Chris Mason
www.anguillaguide.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason
(Lists)
Sent: Wednesday, April 20, 2005 3:36 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SIP Phone Compatability
The Sipura SPA-841 has
:36 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SIP Phone Compatability
The Sipura SPA-841 has everything except memory buttons but has a
directory
and speeddials so I don't think that's so important. Cheap and well
made,
although
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama
Sent: Wednesday, April 20, 2005 9:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Phone Compatability
The SPA-841 doesn't seem to have conference call feature
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
programmable buttons for those features.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Daniel Salama
Sent: Wednesday, April 20, 2005 9:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Phone
, April 20, 2005 2:37 PM
Subject: Re: [Asterisk-Users] SIP Phone Compatability
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
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
If you're using SIP I think what you want is canreinvite=yes which means the
two remote user clients can talk directly to each other. Asterisk disappears
from the loop which means no accounting. I think NAT causes problems in this
scenario also.
More details on the wiki
Regards
Cameron
-
X-lite for Linux.
http://www.xten.com/apps/xprolinuxbeta/
--- Klaus Peras [EMAIL PROTECTED] wrote:
Hey there,
does anybody know a SIP-Client that I only have to
unpack and can run it
on Linux just like SJPhone, except SJPhone??
I need a Softphone for a Levigo Thin-Client, wich is
not
Thibault Lamy wrote:
Do anyone have any experience with SIP phone that support
a headset ? We have Budgetone phones but we need headsets.
I am using Snom 190's with Snom head sets and like them a lot.
On my list of things to do is using the Snom with a Labtec PC headset
and see if that works as
Grandstream are supposed to be releasing a BT103 ? Its a 100 series
phone with headphone jack... when, I couldn't say though.
Thibault Lamy wrote:
Hi there,
Do anyone have any experience with SIP phone that support
a headset ? We have Budgetone phones but we need headsets.
What would you advise
On Thu, February 24, 2005 11:59 am, Thibault Lamy said:
Do anyone have any experience with SIP phone that support
a headset ? We have Budgetone phones but we need headsets.
Just deployed a batch of Sipura SPA-841's. Headset jack is standard so a
few of us are using our cell-phone headsets and
The Uniden UIP200 is a decent phone with a headphone jack if the Sipura
doesn't appeal to you.
-Original Message-
From: Thibault Lamy [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 24, 2005 11:00 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP Phone with
Cisco 7940 :)
- Original Message -
From: Phil Siegrist [EMAIL PROTECTED]
Date: Wed, 22 Sep 2004 10:15:57 -0400
Subject: [Asterisk-Users] SIP Phone
To: [EMAIL PROTECTED]
Hi All,
I am look for recommendations for a good SIP phone, specifically with
a good speaker phone. I have tried
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Phone
Cisco 7940 :)
- Original Message -
From: Phil Siegrist [EMAIL PROTECTED]
Date: Wed, 22 Sep 2004 10:15:57 -0400
Subject: [Asterisk-Users] SIP Phone
To: [EMAIL PROTECTED]
Hi All,
I am look
On Wed, 22 Sep 2004 16:40:04 +0200, Michael Bielicki [EMAIL PROTECTED] wrote:
Cisco 7940 :)
I'll concur with that.
The Cisco 7940 and 7960 phones have great speakerphones :)
As for ones to stay away from - the Grandstream BT-100 series. The
sound is fine on the local end, but is very low for
) 804.400.3686
[EMAIL PROTECTED]
-Original Message-
From: Shaun Ewing [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 22, 2004 11:04 AM
To: Michael Bielicki; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] SIP Phone
On Wed, 22 Sep 2004 16:40:04 +0200
On Wed, 22 Sep 2004 07:56:48 -0700, Wiley E. Siler [EMAIL PROTECTED] wrote:
Do you have a price range?
I don't know about pricing in the US, so I'll skip this (I buy mine in
Australia).
I use Polycom IP500s and the speaker phone is awesome. It picks up
speakers in the room very well at 5-6
) 804.400.3686
[EMAIL PROTECTED]
-Original Message-
From: Shaun Ewing [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 22, 2004 11:04 AM
To: Michael Bielicki; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] SIP Phone
On Wed, 22 Sep 2004 16:40:04 +0200
lines hooked up to it.
Later,
Paul Hales
IT Support
Adairs
-Original Message-
From: Phil Stevens [mailto:[EMAIL PROTECTED]
Sent: Friday, 17 September 2004 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] SIP Phone - PBX Phone
Hi Paul,
I
What type of existing PABX do you have (Make and Model)
What interfaces can you use to connect to your PABX, ie
analog tie lines, E1/ISDN, anything else?
Cheers,
Peter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of P J
Sent: Friday, 17 September
Good to see another Australian user on the list!
You could set up a card with some FXO ports (TDM400?) and use those lines to
hook up the Asterisk box to your existing PABX. But I am sure someone else
will come up with a _much_ more clever solution.
Later,
PaulH
Melbourne
-Original
-certified card? Or, is
there a cheaper option (1 port? for testing purposes) that will satisfy
Austel requirements?
Thanks.
-Original Message-
From: Paul Hales [mailto:[EMAIL PROTECTED]
Sent: Friday, September 17, 2004 2:02 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] SIP Phone
,
Paul Hales
IT Support
Adairs
-Original Message-
From: Phil Stevens [mailto:[EMAIL PROTECTED]
Sent: Friday, 17 September 2004 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] SIP Phone - PBX Phone
Hi Paul,
I have yet to find out the make
.
-Original Message-
From: Paul Hales [mailto:[EMAIL PROTECTED]
Sent: Friday, September 17, 2004 3:17 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SIP Phone - PBX Phone
The TDM400 is used for both PSTN and PABX - PABX connections, from
Of
Jeremy Bogan
Sent: Sunday, August 22, 2004 7:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Phone recommendation for
Receptionist
I've got an installation where there's 12 POTS line
incoming into *,
and am trying to get some
Try the netweb-301 Hard IP Phone attached as a Pic
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of James H.
Thompson
Sent: Sunday, August 22, 2004 11:54 PM
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP Phone
I've got an installation where there's 12 POTS line incoming into *,
and am trying to get some insight as to which VoIP hard phone would be
most suitable for this scenario.
What would you guys recommend?
A Cisco 7960 with the 7914 expansion module [
Jeremy Bogan wrote:
I've got an installation where there's 12 POTS line incoming into *,
and am trying to get some insight as to which VoIP hard phone would be
most suitable for this scenario.
What would you guys recommend?
A Cisco 7960 with the 7914 expansion module [
el Flynn wrote:
Hi there,
I've got an installation where there's 12 POTS line incoming into *,
and am trying to get some insight as to which VoIP hard phone would
be most suitable for this scenario.
Other than the incoming lines, the receptionist would need the normal
keyphone type stuff
What phone do you have?
On Fri, 16 Jul 2004 11:59:39 +0500, atif [EMAIL PROTECTED] wrote:
I am configuring a sip-phone, receing calls, excellent voice quality. but it does
not place calls, please, can some one sort out.
here is my debug output, and below that is sip-debug,
Jul 16
--- Antonio Diego [EMAIL PROTECTED]
escribió: Hi,
First you need to upgrade to the latest CVS and
then
insert a second /
third priority line with hangup in the dialplan.
Regds
Vivian Alan
Hi Alan, thanks for your help.
My hardware is:
-5 budgetone 100
-2 handytone-286
-Asterisk
Hi,
First you need to upgrade to the latest CVS and then insert a second /
third priority line with hangup in the dialplan.
Regds
Vivian Alan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, May 25, 2004 7:48 AM
To: [EMAIL
Hi,
First you need to upgrade to the latest CVS and then
insert a second /
third priority line with hangup in the dialplan.
Regds
Vivian Alan
Hi Alan, thanks for your help.
My hardware is:
-5 budgetone 100
-2 handytone-286
-Asterisk server running on Pentium IV, RAM 1GB,
RedHat 8.0.
I've just
On Tue, Apr 06, 2004 at 05:52:51PM -0600, Rich Adamson wrote:
download ethereal and take a peek at the packets on the wire. Without
something like that, no one is really going to be able to help you.
Do you mean then that my SIP trace displayed at kphone looks otherwise OK --
that the REGISTER
On Tue, Apr 06, 2004 at 05:52:51PM -0600, Rich Adamson wrote:
download ethereal and take a peek at the packets on the wire. Without
something like that, no one is really going to be able to help you.
Do you mean then that my SIP trace displayed at kphone looks otherwise OK --
that the
On Wednesday 07 April 2004 09:24, Richard Airlie wrote:
On Tue, Apr 06, 2004 at 05:52:51PM -0600, Rich Adamson wrote:
download ethereal and take a peek at the packets on the wire. Without
something like that, no one is really going to be able to help you.
I thought I'd chime in here with a
First pass through the trace indicates all udp packets originating from
194.200.209.137 have incorrect checksums. However, the asterisk machine
acknowledged the initial register packet with a 100 Trying, therefore
it must be ignoring udp checksums. (Still curious why incorrect checksums
are
download ethereal and take a peek at the packets on the wire. Without
something like that, no one is really going to be able to help you.
I am clearly doing something ridiculously wrong.
Running Asterisk 0.7.2 on FreeBSD 5.1, I have SIP soft phones which are
unable
At 7:54 AM -0600 3/31/04, Rich Adamson wrote:
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk-a-users-list [EMAIL PROTECTED]
Subject: [Asterisk-Users] Sip phone with push display?
Anyone know of a business class sip hard phone that includes a quality
display capable of supporting push data
Hello, all
Sorry to correct you on this Matt, but I am currently doing this with
the Polycom 600 phones. You need the latest version of both the SIP
software and bootrom to do it, (and that stuff ain't easy to get) but it
is workable.
The latest version of software provides for distinctive ring
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] SIP phone as intercom
Hello, all
Sorry to correct you on this Matt, but I am currently doing this with
the Polycom 600 phones. You need the latest version of both the SIP
software and bootrom to do it, (and that stuff ain't easy to get
[mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 31, 2003 7:57 AM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] SIP phone as intercom
Cool, haven't looked that in depth into the new firmware(is that the 2.4.1
firmware?) I'll have to try that.
I'll post your instructions on the Wiki page
Wow! Thanks John for the detailed information.
This is such an awesome system... and great support here, too.
On Dec 31, 2003, at 12:07 AM, John Baker wrote:
Hello, all
Sorry to correct you on this Matt, but I am currently doing this with
the Polycom 600 phones. You need the latest version of
) on setting the
ALERT_INFO variable in Asterisk?
Thanks,
MATT---
-Original Message-
From: mattf [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 31, 2003 7:57 AM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] SIP phone as intercom
Cool, haven't looked that in depth
Yes, of course. However, that would be a feature of the SIP phone, not
Asterisk, since Asterisk isn't providing the dialtone on your SIP phone,
the phone is doing that.
On Tue, 2003-10-14 at 16:28, Chris Hariga wrote:
Hi,
si posible on SIP phones to have the dial tone after 9 like on the FXS
Peter Pauly wrote:
If Asterisk registers with a SIP long distance provider and
I make a call from an IP phone through Asterisk to that
LD provider, does the RTP (audio) traffic flow between the two
end points directly (normally the IP phone and the LD provider) or
does it flow through
I'm asking because I have Asterisk running behind a NAT firewall
along with an IP Phone (software) and I'm trying to get it
working with Iconnecthere (ICH). I am able to register, connect
, but no audio. I have ports opened up on the firewall, but
they point to the Asterisk machine and not
Sean,
Any recommendations on a hardware based SIP phone to use with *?
I'm looking for something that would be common, as well as quick and easy
to source, somthing relatively quick and easy to configure.
I'm very new to this as well, but with 20+ years of telephony and data
network
Hi Olaf,
I've just started working on a SIP and RTP proxy to handle exactly this.
I'm really just in proof of concept at the moment but just one hour ago
I got a completely successful connection out over NAT in which both
endpoints thought they were talking to the proxy. I should have the code
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