I guess the problem is that * sends the response to port 5060, while the phone
listens on port 2xxx for an answer.
CS
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Jason White
Gesendet: Montag, 29. Oktober 2007 07:46
An:
Hi,
Is it possible to have multi listening bindport in asterisk?
Now days mostly ISPs are Blocking the standard 5060 port so we want to keep
option if 5060 is blocked we can ask our customers to use another port.
Thank You
Abdul
__
Do You
On Mon, Oct 29, 2007 at 08:17:20AM +0100, Christian Stredicke wrote:
I guess the problem is that * sends the response to port 5060, while the
phone listens on port 2xxx for an answer.
That could be the problem.
The phone specifies port 2048 in its contact field. Is there a way to
configure
Well, the response should go to the port number provided in the Via header. If
there is a rport set, then to that port. Everything looks good in the log, the
only problem is that the response is sent to the wrong port.
The Contact port will be used later when the server wants to send a request
On Mon, Oct 29, 2007 at 09:22:21AM +0100, Christian Stredicke wrote:
Well, the response should go to the port number provided in the Via header.
If there is a rport set, then to that port. Everything looks good in the
log, the only problem is that the response is sent to the wrong port.
I
What you can still to is setting the port on the phone to port 5060 - just as a
little dirty workaround until there is a better solution available.
CS
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Jason White
Gesendet: Montag, 29. Oktober 2007
On 10/29/07, Christian Stredicke [EMAIL PROTECTED] wrote:
What you can still to is setting the port on the phone to port 5060 - just as
a little dirty workaround until there is a better solution available.
CS
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL
Just a little follow up here...
Missed a call from someone at Telewest on friday, so I don't know what
they were going to tell me.
However I've come in this morning and thought well, you never know,
perhaps he was phoning to say we've fixed it. Tried calling my mobile
and now it works. No idea
On Mon, Oct 29, 2007 at 10:19:57AM +0100, Christian Stredicke wrote:
What you can still to is setting the port on the phone to port 5060 - just
as a little dirty workaround until there is a better solution available.
Would that be the sip_port settings entry? It is documented as for internal
On Thu, Oct 25, 2007 at 07:13:52PM +0200, Vincent wrote:
On Thu, 25 Oct 2007 18:46:19 +0200, Vincent
[EMAIL PROTECTED] wrote:
I guess I should use this as a parameter to a function, but which one?
Never mind, I found how to use it:
exten =
Hi all, I use asterisk with realtime features for extension, sip and iax.
In extensions.conf I have put these lines:
[from-internal]
include = parkedcalls
switch = Realtime/@
[fromiax]
switch = Realtime/@
There is a way for put in my database the context also? Now if I want to
add a new
Hi
Sorry to use this public place, but IRC and emails to [EMAIL PROTECTED]
have not helped in the past.
I have several issues with using the files server downloads.digium.com,
which has replaced the simple ftp/http file server ftp.digium.com.
In downloads.d.c the directory listing is served
you can do it using iptables, port forwarding.
On 10/29/07, Abdul [EMAIL PROTECTED] wrote:
Hi,
Is it possible to have multi listening bindport in asterisk?
Now days mostly ISPs are Blocking the standard 5060 port so we want to keep
option if 5060 is blocked we can ask our customers to use
Tzafrir Cohen wrote:
Hi
Sorry to use this public place, but IRC and emails to [EMAIL PROTECTED]
have not helped in the past.
I have several issues with using the files server downloads.digium.com,
which has replaced the simple ftp/http file server ftp.digium.com.
In downloads.d.c the
Conall O'Brien wrote:
Hello,
I'm currently looking at FXO options to provide a POTS line to Asterisk to
trunk calls with.
Does anyone have any experience using the Linksys Sipura 3201 as an FXO
device for Asterisk?
I use one at home and can recommend it as functional and reliable.
HELLO ALL!
I followed a tutorial called DUNDi so easy to set up DUNDi peers.
Unsurprising it was not that easy hehe.
I have the following files up and running, peers are visible but when I
do a query e.g dundi lookup [EMAIL PROTECTED] I get the following error.
CAUSE: NOAUTH: Unsupported
In article [EMAIL PROTECTED],
Tzafrir Cohen [EMAIL PROTECTED] wrote:
Sorry to use this public place, but IRC and emails to [EMAIL PROTECTED]
have not helped in the past.
I have several issues with using the files server downloads.digium.com,
which has replaced the simple ftp/http file
Hi,
I'm currently looking at FXO options to provide a POTS line to Asterisk to
trunk calls with.
i've had some problems setting the disconnect tone correctly to my
country. As a matter of fact, i still do, as the calculated values does
not always hang up the phone.
Other than this i
Hello
I'm learning more about dialplans and have a couple of questions:
1. Am I right in understanding that the actions that can be performed
in extensions.conf can be of two types only:
- internal commands (Dial, Wait, etc.)
- calls to external scripts throught AGI?
2. I'd rather write scripts
Tzafrir Cohen wrote:
Furthermore, I cannot follow links directly. Links are redirections.
For instance, the link marked with aadk points to:
http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/telephony/aadk
$ HEAD
Hi all,
i need an XML file format which is used in remote provisioning of different
spa devices. Please somebody tell me the format or tell me where can i find
it on the internet. I also need a list of parameters which are configured
using auto-provisioning.
--
Best Regards
Rizwan Hisham
Dave Fullerton wrote:
Tzafrir Cohen wrote:
Let's look at http://downloads.digium.com/pub/telephony/
I get a list of items. I have to guess which of them is a file and which
is a directory. There is no proper date of change.
Not sure I completely understand what you mean by I have to guess
On Mon, Oct 29, 2007 at 09:02:14AM -0400, Dave Fullerton wrote:
Tzafrir Cohen wrote:
Hi
Sorry to use this public place, but IRC and emails to [EMAIL PROTECTED]
have not helped in the past.
I have several issues with using the files server downloads.digium.com,
which has replaced
Rizwan Hisham wrote:
Hi all,
i need an XML file format which is used in remote provisioning of
different spa devices. Please somebody tell me the format or tell me
where can i find it on the internet. I also need a list of parameters
which are configured using auto-provisioning.
For SPA-921
Hi,
I have asterisk installed.
When a connection comes from the outside one of our phones rings for about 45
seconds.
Is it possible to another phone fetch the call while it's ringing on the first
phone?
I don't want to use ringgroups because the second phone would be ringing also.
Thanks
I started out a few years ago with some SPA-841 sets, because the
Grandstream 2000 I thought I wanted was perpetually delayed. The GS had more
call appearances, and I didn't want just the 4 max that the SPA offered. As
it turns out, with the greater flexibility of VOIP, I don't need 'dedicated'
Hi Friends!
I need help! I'm still Asterisk rookie, so please
forgive me.
My Asterisk is no longer answering incoming call on
the phone line. I call the phone and it rings but
asterisk is not picking it up. The phone line is
attached to port 4 (FXO) on my digium TDM411P card.
I am running
i have spa 2100. tried to access the file but got 404 not found. Any clues
why?
On 10/29/07, Per Jessen [EMAIL PROTECTED] wrote:
Rizwan Hisham wrote:
Hi all,
i need an XML file format which is used in remote provisioning of
different spa devices. Please somebody tell me the format or
Am Montag, den 29.10.2007, 15:54 + schrieb Gordon Henderson:
On Mon, 29 Oct 2007, Abdul wrote:
Hi,
Is it possible to have multi listening bindport in asterisk?
Now days mostly ISPs are Blocking the standard 5060 port so we want to
keep option if 5060 is blocked we can ask our
I'm confused about something.
It's the way Asterisk handles the A leg (ie the first party dialed) on an
originate command via the Manager Interface.
Lets say our originate commands looks like this:
ACTION: Originate
Async: yes
Timeout: 6
Exten: callback
Channel: SIP/[EMAIL PROTECTED]
Gordon Henderson wrote:
On Mon, 29 Oct 2007, Abdul wrote:
Hi,
Is it possible to have multi listening bindport in asterisk?
Now days mostly ISPs are Blocking the standard 5060 port so we want to
keep option if 5060 is blocked we can ask our customers to use another
port.
Hi All,
Does anyone know of a good virtual appliance for Asterisk under VMware?
I am very interested in the JEOS concept for reducing the attack surface
of a machine, so I think an appliance might be a good way to do this. BTW,
I'll be using this with direct SIP Trunking and Snom 370/360 IP
Zaheer Master wrote:
Hi All,
Does anyone know of a good virtual appliance for Asterisk under VMware?
I am very interested in the JEOS concept for reducing the attack surface
of a machine, so I think an appliance might be a good way to do this. BTW,
I'll be using this with direct SIP Trunking
Tzafrir Cohen wrote:
On Fri, Oct 26, 2007 at 04:52:07PM -0800, Mojo with Horan Company, LLC
wrote:
I don't have T1 but it seems that the first time I run ztcfg (or in
fact, the zaptel startup script runs it for me) it fails.
What distribution is it?
RHEL4 / CentOS4 has an
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
brand/model would you recommend. We were personally thinking in
recommending either Cisco, Aastra, Polycom, or Snom, for we've heard
great
Why dont you make 2 separate Originate actions, one for each call leg.
Then call Bridge manager Action whenever you want.
Moy
On 10/29/07, Douglas Garstang [EMAIL PROTECTED] wrote:
I'm confused about something.
It's the way Asterisk handles the A leg (ie the first party dialed) on an
What is the use case?
Linksys, Polycom, Snom, and Aastra all have their strengths and weaknesses.
--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of
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
Lyle Giese wrote:
Philipp Kempgen wrote:
Lyle Giese wrote:
I had a working 1.0.x Asterisk setup using:
SetVar(ALERT_INFO=http://127.0.0.1/Bellcore-dr2)
Which used the short quick rings.
In Asterisk 1.4, I have tried several things, but I think the correct
syntax is:
We used to have this problem with 1.2, too. I think it was some timing
thing that resulted from the caller hanging up at just the right (or
should I say, wrong) moment, like after the min-message-len timer. I
won't tell you what we did to fix it, because you don't want to hear about
upgrading to
On Mon, 29 Oct 2007, Abdul wrote:
Hi,
Is it possible to have multi listening bindport in asterisk?
Now days mostly ISPs are Blocking the standard 5060 port so we want to
keep option if 5060 is blocked we can ask our customers to use another
port.
Really?
What country?? What ISP?
This
MFC/R2 on AsteriskNOw!! How?
Please!!!
Thanks!!
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
This question is about 1.2.x asterisk. Please no flames, or you should
upgrade to 1.4.
Does anyone know what might be the cause for 'stuck voicemail's in
1.2.6asterisk? By stuck, I mean the phones show a voicemail, and if
you log in
you get you have 1 new voicemail, and if you delete it it says
Our features.conf let us set *8 to pick up a ringing line elsewere in
the system. I believe it can be extended to *8x, to pick up a specific
group.
moj
Nuno Fernandes wrote:
Hi,
I have asterisk installed.
When a connection comes from the outside one of our phones rings for about 45
Nuno Fernandes wrote:
Hi,
I have asterisk installed.
When a connection comes from the outside one of our phones rings for about 45
seconds.
Is it possible to another phone fetch the call while it's ringing on the
first
phone?
I don't want to use ringgroups because the second phone
Matt wrote:
This question is about 1.2.x asterisk. Please no flames, or you
should upgrade to 1.4.
Does anyone know what might be the cause for 'stuck voicemail's in
1.2.6 asterisk? By stuck, I mean the phones show a
I would suggest at least upgrading to the current 1.2.x series,
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)
We have that problem here with Asterisk 1.2.9.1. There is a fix in later
versions of the 1.2 branch, but I couldn't tell you which one. You can just
delete the .txt file from the user's voicemail folder and it should clear
the MWI on the phone.
On 10/29/07, Matt [EMAIL PROTECTED] wrote:
This
Stay away from Cisco they just don't work for the price, if it would
be in the price range of a Grandstream phone I would tell you go for
it, but at the current price its just not worth it. Aastra, Polycom or
linksys all work for me. Never tried Snom before.
On 10/29/07, [EMAIL PROTECTED] [EMAIL
just install chan_unicall.so
On 10/29/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
MFC/R2 on AsteriskNOw!! How?
Please!!!
Thanks!!
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To
Matt wrote:
This question is about 1.2.x asterisk. Please no flames, or you
should upgrade to 1.4.
Does anyone know what might be the cause for 'stuck voicemail's in
1.2.6 asterisk? By stuck, I mean the phones show a voicemail, and if
you log in you get you have 1 new voicemail, and if you
Hi,
On Mon, 2007-10-29 at 10:29 -0700, Douglas Garstang wrote:
I'm confused about something.
It's the way Asterisk handles the A leg (ie the first party dialed) on
an originate command via the Manager Interface.
Lets say our originate commands looks like this:
ACTION: Originate
Async:
Matt wrote:
This question is about 1.2.x asterisk. Please no flames, or you should
upgrade to 1.4.
Does anyone know what might be the cause for 'stuck voicemail's in
1.2.6asterisk? By stuck, I mean the phones show a voicemail, and if
you log in
you get you have 1 new voicemail, and if
Abdul wrote:
Hi friends,
We have are getting SIP/2.0 407 Proxy Authentication Required on
Invite pakcet once Nokia E65 trying to dial number. But it can recive
well from other caller.
We tried to disable secrete and it worked fine. But we have lot of
users and disabling secrete is risky.
No:
register = abc:[EMAIL PROTECTED]
[peer]
host=zzz
Its possible to make mistakes and typos you know. Maybe you can post
your config file and we can help you.
On 10/26/07, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi Pablo;
How the IP address will be wrong, and asterisk able to
do
Take a look at http://spc.pifiu.com there they have the spc.exe (
Linux variant) which will generate the sample XML file for your
firmware version. There is also in PDF format the admin guides that
explain all the parameters.
On 10/29/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
i need
I suppose the VMware image of AsteriskNow is a good place to start? I just
found this and I think it answers my question :)
Regards,
Zaheer K. Master
President, Adamant Security Inc.
___
--Bandwidth and Colocation Provided by
All,
I run a bunch of (well 20+ actually) Asterisk boxes at home, work,
friends and the lie with our own dialplan in the form 8EE where 'EE'
is the exchange number and '' is the extension number.
This arrangement has been in for 2+ years and worked well with a central
box
On Mon, Oct 29, 2007 at 03:44:13PM +, Jeng Yu wrote:
Hi Friends!
I need help! I'm still Asterisk rookie, so please
forgive me.
My Asterisk is no longer answering incoming call on
the phone line. I call the phone and it rings but
asterisk is not picking it up. The phone line is
doesn't look legit to me.
It's got CE/FCC emblems, but no ID #'s ?!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kyle
Sexton
Sent: Monday, October 29, 2007 5:35 PM
To: Asterisk Users Mailing List
Subject: [asterisk-users] Mystery phone!
Does anyone
Does anyone know who really makes this phone:
http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/
Large pictures are at the bottom:
http://www.hybsys.bg/img/ipph/IP5000_1.jpg
http://www.hybsys.bg/img/ipph/IP5000_2.jpg
--
Kyle Sexton
___
--Bandwidth
Hi:
Iam using Fedora core 5 .
Thanks in advance;
Date: Mon, 29 Oct 2007 10:23:28 +0530 From:
[EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re:
[asterisk-users] Realtime Mysql error On 10/27/07, wassim darwish wrote:
Hi: Iam using an asterisk
I've had experience with Linksys and Polycom. Either one is easy enough
to provision. Took me a while to understand how to provision Polycom.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, 30 October 2007 3:42 AM
To:
On Mon, Oct 29, 2007 at 10:19:49AM +, Steve Davies wrote:
snom phones have been using ports in the 2000+ range since the dawn of
asterisk without any problems, so I suspect that this will be an
Asterisk configuration error, or a change to the asterisk SIP stack
that is causing problems.
If you go to linksys's website and click on partners then apply for
partnership you will be able to get access to the documents programs
you need
John
[EMAIL PROTECTED] wrote:
Take a look at http://spc.pifiu.com there they have the spc.exe (
Linux variant) which will generate the sample XML
Or you can download them at http://spc.pifiu.com and not have to go
through that bullshit.
On 10/29/07, John Mason Jr [EMAIL PROTECTED] wrote:
If you go to linksys's website and click on partners then apply for
partnership you will be able to get access to the documents programs
you need
Hmm the shape looks like an Aastra but the buttons down the side look
like PlayStation buttons to me. Maybe it's a Sony Cisco joint effort.
Joel.
On Mon, 2007-10-29 at 16:35 -0500, Kyle Sexton wrote:
Does anyone know who really makes this phone:
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)
Hi all, I use asterisk with realtime features for extension, sip and iax.
In extensions.conf I have put these lines:
[from-internal]
include = parkedcalls
switch = Realtime/@
[fromiax]
switch = Realtime/@
There is a way for put in my database the context also? Now if I want to
add a
Smith, Rick wrote:
doesn't look legit to me.
It's got CE/FCC emblems, but no ID #'s ?!
If that is a mark of legitimacy, then most equipment must be fake. :-)
Steve
___
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Could anybody help?
Can you show a CLI session? The error you get is not familiar.
Otherwise your configs look ok, did you make the keys priv or dundi?
There was an error in the howto, the example was to make the keys
named priv but in dundi.conf the keys were named dundi, double check
that as
That's really a question for [EMAIL PROTECTED]
The short and generally not very helpful answer is that there are a lot
of poorly packaged software releases out there that don't play well with
cross-development environments.
-Philip
Douglas Garstang wrote:
I'm trying to build an Asterisk rpm
Has anyone had any experience in getting Asterisk to interoperate with a
Coppercom switch using SIP, either as subscriber lines or else as a
trunked configuration?
And if so, do you have any configs you could share (for both ends)?
Thanks,
-Philip
I'm trying to build an Asterisk rpm from the supplied asterisk.spec file.
Made numerous changes to get it to work.
The architecture of the system I am building on is x86_64. I'd like to build
for i686 though.
I added a --target i686 to the rpmbuild line in the Makefile, but it looks like
it's
Since I'm executing a 'make rpm' from within the Asterisk 1.4.13 distribution
source, I'd say it's an Asterisk question.
- Original Message
From: Philip Prindeville [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
Mojo with Horan Company, LLC wrote:
Lyle Giese wrote:
Philipp Kempgen wrote:
Lyle Giese wrote:
I had a working 1.0.x Asterisk setup using:
SetVar(ALERT_INFO=http://127.0.0.1/Bellcore-dr2)
Which used the short quick rings.
In Asterisk 1.4, I have tried several
I don't know that I would want to download an unauthorized copy of a
program to run on my computer without means to verify it's authenticity.
And even if the programs and docs are valid, why not sign up and get
them from the source, might even be beneficial.
John
[EMAIL PROTECTED] wrote:
Or
On Oct 29, 2007, at 5:35 PM, Kyle Sexton wrote:
Does anyone know who really makes this phone:
http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/
Large pictures are at the bottom:
http://www.hybsys.bg/img/ipph/IP5000_1.jpg
http://www.hybsys.bg/img/ipph/IP5000_2.jpg
I don't know who makes
The web site is Russian (Serbian I think). Company is Hybird Systems
(Hibridni System AD). Best I can tell which probably does not help much
except to say it is a legit company that has been around a long time making
computer stuff since the 60's.
On 10/30/07, Kyle Sexton [EMAIL PROTECTED]
Maybe I'm not following the problem here ... couldn't he just rework his
extensions in a way that uses macros so he doesn't have to change 50 things?
On 10/30/07, JR Richardson [EMAIL PROTECTED] wrote:
Hi all, I use asterisk with realtime features for extension, sip and
iax.
In
Read all the options of the Dial() function. There are options you can mess
with to play something while the call is ringing (music on hold feature if I
recall). Check out all the Dial options.
On 10/29/07, Douglas Garstang [EMAIL PROTECTED] wrote:
I'm confused about something.
It's the way
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