On Fri, Sep 26, 2008 at 6:00 AM, Darrick Hartman
[EMAIL PROTECTED] wrote:
Dean Collins wrote:
I'd also like to know what happens when someone 'chats' to the account
connected to the Asterisk server.
Good question
Lots of questions about this one. There's definitely a demand for it so
I can
Paul Hales wrote:
Just to check - have you got the right modules plugged into the right
sort of lines?
Yes - if you have a look at my zapata.conf snippet the zap chanell has
signalling=fxs_ks
Also - some analog phone interfaces are NOT standard. :(
This could be the problem. However,
On Thu, 25 Sep 2008, Philipp Kempgen wrote:
Jon Weisman schrieb:
I'm planning on getting a Dell PowerEdge 1950.
All I can tell is that I have bad experiences with those Dell
PowerEdges. A standard Debian Etch install (2.6.18 kernel I think)
didn't even have the driver to run the network
yes i have ztdummy loaded. i assume that is what i want.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Gibson
Sent: Wednesday, September 24, 2008 8:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re:
Its really just a very minor system I am running, its sole purpose is a
vm basically. Well a VM that can redirect calls based on number.
I would prefer to just run it on this windows machine doing nothing most
of the time. Id rather not buy an appliance, maybe if its $100 but I
would rather
Shotgunning the use of IP addresses is foolish at best and lazy
programming at worst. Imagine if the poeple writing browsers did that!
The internet could end up with double or triple the traffic for no
extra benefit not to mention the additinoal load on web servers etc.
It's not particularly
On Sep 25, 2008, at 9:00 PM, Darrick Hartman wrote:
Dean Collins wrote:
I'd also like to know what happens when someone 'chats' to the
account
connected to the Asterisk server.
Lots of questions about this one. There's definitely a demand for
it so
I can see why Digium would be
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Michael J. Liberatore wrote:
Its really just a very minor system I am running, its sole purpose is a
vm basically. Well a VM that can redirect calls based on number.
I would prefer to just run it on this windows machine doing nothing most
of
On Thu, Sep 25, 2008 at 05:25:52PM -0700, Fred Posner wrote:
On Sep 25, 2008, at 11:06 AM, Steve Anness wrote:
So what a minute. They will charge us to use Skype with our Asterisk
servers? Yes, I think I shall move along.
Steve
I talked with both Skype and Digium today at
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Arturo Ochoa wrote:
Ok, so it's clear now that this feature is missing on Asterisk, but as
Russell states, it's on the roadmap.
So, Can you guys give an alternate idea on what to do on this scenario:
One customer has this situation:
The
On Thu, 25 Sep 2008 18:00:00 +0100, Tim Panton wrote:
It's essentially a channel driver.
Licensed per channel in the same way that the g729 codec is.
which would mean that us freebsd folks are going to be left out. oh well.
--
Regards, /\_/\ All dogs go to
On Thu, 25 Sep 2008 12:21:41 -0400, Jason Aarons \(US\) wrote:
A lot of places you still can't get GSM in the US.it has
improved...but GSM 3G coverage is lacking compared to EVDO/CDMA.
which isn't usually a problem as all 3G phones i've seen also use GSM, and
the phones switch to GSM when
Hi,
there is http://bugs.digium.com/view.php?id=13405 updated version of fax
(T38) gateway.
Your bug reports and questions are welcome.
Thank you in advance.
Best regards
Daniel.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
I have tryed skip2pbx 580€ yeastar 60 €, the quality is the way behind of a
good sip provider, thay are simply not suitable for business, i hope it
would not be the case of asterisk addon. Also i wonder if skype auto relay
will be disabled (bandwith), wait and see...
Hi,
if you are interested in t.38 gatewaying you may try fax gateway that has
been posted recently: http://bugs.digium.com/view.php?id=13405. I'm looking
forwards seeing any reports.
Best regards
Daniel.
On Fri, Sep 26, 2008 at 11:10 AM, Adam Goryachev
[EMAIL PROTECTED] wrote:
-BEGIN PGP
Hello,
I would like to push presence from one asterisk to another.
Here is my scenario:
Office A has 3 users: extension 100,101,103
Office B has 3 users: extension 200,201,203
Now 200 would like to see on his phone (BLF) when user 100 is on the
phone.
Asterisk of Office A and Asterisk of Office
Hi,
I'm using Asterisk 1.2.
I have to redirect a call coming from a line with DIDs to an ATA devices
but keeping the DNID just as Asterisk would be DNID-transparent. I
need this because the machine connected to my ATA needs to know which
DID was called from outside.
Anybody knows if DNID can be
Hi
I am using
show cannels verbose
to get info about my current sip calls. However, the time displayed is
always zero.
Any hints ?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix,
On 26 Sep 2008, at 11:17, Grygoriy Dobrovolskyy wrote:
I have tryed skip2pbx 580€ yeastar 60 €, the quality is the way
behind of a good sip provider, thay are simply not suitable for
business, i hope it would not be the case of asterisk addon. Also i
wonder if skype auto relay will be
Hello,
I use an Asterisk box with the following configuration:
Operating System : linux Fedora Core 4 (2.6.17-1.2142_FC4smp #1)
Asterisk 1.4.18
I use the following asterisk command to send url to client :
Dial(IAX2/ciwww/[EMAIL
Gordon Henderson schrieb:
However, given the past history of problems I've seen people writing about
on this list, I'd be very suspicious of using Dells with plug-in cards.
Dells themselves are fine, but it seems there are IRQ issues with some of
their systems... (Search the archives)
On 26 Sep 2008, at 04:36, Dean Collins wrote:
I'd also like to know what happens when someone 'chats' to the account
connected to the Asterisk server.
I asked Mark about that.
They expect to have text to work right, when associated with a voice
call.
It is less clear what happens it it is
Loic Didelot schrieb:
I would like to push presence from one asterisk to another.
Here is my scenario:
Office A has 3 users: extension 100,101,103
Office B has 3 users: extension 200,201,203
Now 200 would like to see on his phone (BLF) when user 100 is on the
phone.
Asterisk of
On Fri, 2008-09-26 at 08:43 +0100, Grey Man wrote:
It's not particularly difficult to determine the best IP address for a
piece of client software to use.
Oh?
Check the local machines default
gateway, apply the subnet mask and then compare it against all the
local IP's.
Yeah? And if
i!
Not about this directly, but an alternative. If you need the length of
finished calls, work with the system. Use a specific call to the date command,
so it's easy to evaluate the time info or some other tool to give you an
absolute of time. Then at the end of the call use another system
I'm a little surprised no one wants to say anything on IRC this
morning about this. I know many of you here are interested. Mark was
talking about this three years ago and it was exciting news then as it
is now (IMO).
Maybe Mark will join us, although I believe he's got a long flight
today or
Giorgio Incantalupo schrieb:
I'm using Asterisk 1.2.
Anybody knows if DNID can be modified?
Not sure about 1.2 but at least in 1.4 you can set CALLERID(dnid).
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566
Get Olle to call in for once in his life!
Mark did say IM and video, IM first. It's all gonna happen. (just not
right away)
On Fri, Sep 26, 2008 at 3:34 PM, Tim Panton [EMAIL PROTECTED] wrote:
On 26 Sep 2008, at 04:36, Dean Collins wrote:
I'd also like to know what happens when someone
i upgraded to 1.4.21.2-2 and set the autofill to on and it solved the
problem.. yet i kept the failover settings incase it happens again.. so if it
happens.. the fail over will redirect the caller to the same queue but the
conditions will apply like it was a new call.
i found out that there
Hi
Thanks for the hint, however I do already have a cdr tool for finished
calls.
core show channels verbose
does show the duration of calls in real time. However, it does not work all
the time, I.e. at times it works great other times it just displays 0 for
the call duration, although the call
On Fri, Sep 26, 2008 at 2:36 PM, Brian J. Murrell [EMAIL PROTECTED] wrote:
Check the local machines default
gateway, apply the subnet mask and then compare it against all the
local IP's.
Yeah? And if more than one matches? Then what?
Use one of them!
And if the network set up is too
On Fri, Sep 26, 2008 at 03:44:01PM +0200, randulo wrote:
Get Olle to call in for once in his life!
Mark did say IM and video, IM first. It's all gonna happen. (just not
right away)
On the topic of #pidgin they say, amomng others, Pidgin does NOT
support voice or video. Likewise we should
On Fri, 2008-09-26 at 14:54 +0100, Grey Man wrote:
On Fri, Sep 26, 2008 at 2:36 PM, Brian J. Murrell [EMAIL PROTECTED] wrote:
Yeah? And if more than one matches? Then what?
Use one of them!
And if the one I choose to use doesn't work because of some kind of
policy routing or
A machine with more than one default gateway is a VERY special case
(used for load-balancing or possibly failover). Most systems will not
allow it. I mean... logically, it's odd. Default means when not applied
to any other special rule, I choose this one.Not this two. Not this
three. This one.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Brian J. Murrell
Sent: Friday, September 26, 2008 10:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] sip forking needed for ekiga 3.0
I've read
2008/9/26 randulo [EMAIL PROTECTED]
Get Olle to call in for once in his life!
Mark did say IM and video, IM first. It's all gonna happen. (just not
right away)
http://lists.digium.com/mailman/listinfo/asterisk-users
Video ? that could be really nice but limited to pc/macasteriskwhatever.
carl Lougher wrote:
Howdy,
Running asterisk 1.4.13
Sometime when running a sip reload the clients are unable to make and receive
calls..
Any pointers?
That can happen when Asterisk is contacting DNS servers to resolve host
names and there are delays in responses (which is done with a sip
On Fri, 2008-09-26 at 10:16 -0400, SIP wrote:
The RFCs are there for a reason. All SIP forking is UAS territory. Not
UAC territory.
From http://bugzilla.gnome.org/show_bug.cgi?id=553810 Damien Sandras
asks:
I repeat, Ekiga is doing something perfectly legal.
The real
Brian J. Murrell wrote:
On Fri, 2008-09-26 at 10:16 -0400, SIP wrote:
The RFCs are there for a reason. All SIP forking is UAS territory. Not
UAC territory.
From http://bugzilla.gnome.org/show_bug.cgi?id=553810 Damien Sandras
asks:
I repeat, Ekiga is doing something
On Fri, 2008-09-26 at 10:41 -0400, SIP wrote:
Oh yes. It's perfectly legal.
It's also a) NOT SIP forking, b) Lazy, and c) Poorly designed.
Sending multiple requests and hoping and praying that the recipient will
ignore two of them (it will NOT in many cases -- specifically set out by
the
On Fri, Sep 26, 2008 at 11:59:35AM +0300, Tzafrir Cohen wrote:
On Thu, Sep 25, 2008 at 05:25:52PM -0700, Fred Posner wrote:
On Sep 25, 2008, at 11:06 AM, Steve Anness wrote:
So what a minute. They will charge us to use Skype with our Asterisk
servers? Yes, I think I shall move
On Wednesday 24 September 2008 19:28:17 Michael J. Liberatore wrote:
Hi, i am running a small personal asterisk server for my business, and
instead of getting a dedicated machine to run linux which would waste
power and money i decided to run it on my windows xp sp2 machine. The
machine is
Steve Murphy wrote:
On Thu, 2008-09-25 at 14:21 +, Tariq .. wrote:
Greetings,
i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox..
i tried to creat an SIP link between both servers and i discovered that one of my servers is not allowing the other to send calls
Brian J. Murrell wrote:
And so will this channel driver also allow Skype to use my resources
(CPU, bandwidth -- i.e. Internet for which many have usage caps, etc.)
the way the Skype client does?
The Skype engine in Skype For Asterisk does not currently have 'relay'
support, so it does not
I have a box running asterisk 1.4.17 that had been working.
it has 2 uniden phones connected on it.
This was working and now the phones dont ring when calling each other.
below is the sip debug. I cant see why the other phone does not ring?
I also tried changing the canreinvite for no to yes but
- Tzafrir Cohen [EMAIL PROTECTED] wrote:
There are some awkward methods for sending some text messages oversome
channels (SMS in european POTS, SIMPLE and simpler texxt messages in
SIP, XMMP for Jingle, and well, probably nothing in IAX. Bristuff ads
even a few more bits there).
But do
Hi, when I make a call I need that the caller can** hang up by dialing
***(H option in Dial command), the call but it don´t work.
Command
EXEC DIAL Zap/g1/433391|20|H
In CLI...
-- AGI Script Executing Application: (DIAL) Options: (Zap/g1/433391|20|H)
-- Requested transfer capability: 0x00
Asterisk version 1.2.27
We are running into issues where people are not deleting their
voicemails and it is filling up the storage for voicemail. We would
like to run a script that dumps all voicemail that are older than X
days.
Can we simply check the date time stamp on the message
Hi, all. We've got a PoS legacy PBX at my company that doesn't have call
accounting. I figured, Hey, why not stick a dual-span T1 Asterisk-based
system in the middle? Then, I just passively pass in-bound calls to the
PBX, and outbound calls to the PSTN. I can then have Asterisk do all the
call
Jerry Geis wrote:
I have a box running asterisk 1.4.17 that had been working.
it has 2 uniden phones connected on it.
This was working and now the phones dont ring when calling each other.
below is the sip debug. I cant see why the other phone does not ring?
I also tried changing the
Brian J. Murrell wrote:
On Fri, 2008-09-26 at 10:16 -0400, SIP wrote:
The RFCs are there for a reason. All SIP forking is UAS territory. Not
UAC territory.
From http://bugzilla.gnome.org/show_bug.cgi?id=553810 Damien Sandras
asks:
I repeat, Ekiga is doing something perfectly
On Fri, 26 Sep 2008, Asterisk User List wrote:
Asterisk version 1.2.27
We are running into issues where people are not deleting their
voicemails and it is filling up the storage for voicemail. We would
like to run a script that dumps all voicemail that are older than X
days.
Can we
snip
Based on the SIP debug included here, it appears that Asterisk is not
receiving
a response to the INVITE it is sending to 522 (192.168.1.99). Since the phone
is
not ringing, it makes me suspect that for some reason the linksys is
preventing
the INVITE from reaching the phone.
Asterisk User List schrieb:
Asterisk version 1.2.27
We are running into issues where people are not deleting their
voicemails and it is filling up the storage for voicemail. We would
like to run a script that dumps all voicemail that are older than X days.
Can we simply check
Hi,
I have a complex job totally unrelated to asterisk. I only post here
because there are so many bright people on the list. Sorry, but
someone may need a buck so write me if you are interested. Otherwise,
ignore.
We have as input a newsletter type document, originally in MS Word
(but obviously
Your idea (and adam's to run xen) is a very good idea. I have
considered it but I'd rather not do a complete reinstall on this xp
machine, but if I can deal with that then it would prob work well.
I am going to play with the settings, etc to try to get this working
first though. Or like I
Dean Collins wrote:
I'd also like to know what happens when someone 'chats' to the account
connected to the Asterisk server.
Keeping in mind that the product has not yet entered beta testing... at
this time, all chat messages are ignored by the Skype For Asterisk
product. We have discussed
randulo schrieb:
I have a complex job totally unrelated to asterisk. I only post here
because there are so many bright people on the list. Sorry, but
someone may need a buck so write me if you are interested. Otherwise,
ignore.
We have as input a newsletter type document, originally in MS
2008/9/26 Kevin P. Fleming [EMAIL PROTECTED]
Brian J. Murrell wrote:
And so will this channel driver also allow Skype to use my resources
(CPU, bandwidth -- i.e. Internet for which many have usage caps, etc.)
the way the Skype client does?
The Skype engine in Skype For Asterisk does not
On Thu, Sep 25, 2008 at 7:34 AM, Rizwan Hisham [EMAIL PROTECTED] wrote:
The fax is originated from a fax machine connected to an ata which supports
t38.
That would be great if Asterisk had true T.38 support. It can pass the
T.38 packets it receives to another SIP endpoint (it will do this even
I know of someone who was involved in a software project like this -
scanning paper documents and importing files into a massive searchable
database for a large legal company.
Many of the documents were more than 1000 pages long.
The amount of money spend on the project was stunning.
PaulH
Hello there, I wan to know what is the files that have the control of
the quality the sound, When I call a extension, and reproduced a file
gsm, or I tolk why another extension, have noise... I thinks that is
because have bad quality in the .conf.
Thanks.
Abel
Hi!
I think all - at least all PSTN - calls have the same quality in means of
bitrate, number of channels and samplerate.
It's 8kHz, 16bit and mono.
About noise, I didn't have problems with that. Seems it's not really about
quality. Probably it would be helpful, if you tell us, which
Grygoriy Dobrovolskyy wrote:
Will it be packed into the base asterisk package, or to asterisk-addons?
or into some third party ?
Would it be possible to buy some comminication licences use them while
disabling the 'relay' function ?
Skype For Asterisk will be distributed as a separate
Philipp Kempgen wrote:
Junghanns' BriStuff can do it via ESEL (extension state export
logic). Basically that's a connection between the AMIs.
In Asterisk 1.6 you could do it via DEVSTATE().
http://www.asterisk.org/blog/8
Asterisk 1.6.1 will have distributed device state as well, although
Dave Poirier wrote:
I have an Asterisk server running 1.4.20 and I have all my users in
users.conf. Inside users.conf I used...
#include ww-users.conf
Thats seems to work great with one exception...
The exception is that anytime anyone updates their voicemail password,
Asterisk rewrites
My outbound dialing rule was incredibly complex:
exten = _X.,1,Dial(${PASSTHROUGHTRUNK}/${EXTEN})
And everything seemed to be working ducky, until I went to call Germany
and got -- a local cell phone number. Needless to say, this puzzled me
greatly. A quick look at my log, though, showed
- Original Message -
From: Julien Claassen [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, September 26, 2008 8:03 PM
Subject: Re: [asterisk-users] Audio Files
Hi!
I think all - at least all PSTN - calls
One option might be to run in the opposite vmware direction. That is, run
Linux as the native OS and run Windows within a vmware instance. That
gives you the Windows compatibility for your applications, while at the same
time providing the critical hardware timing for your Asterisk instance.
Has anyone seen or know of a iphone/ipod sip client that may be in the
works?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
Hi,
I was wondering if there is anyway to split, say, 300 calls that come in
from the SIP provider across 10 asterisk servers with 30 agents each,
without having the telco do the splitting. Is there any way to do call
distribution, e.g. we send an incoming call to a similar queue on the next
You can set up a proxy to round-robin/load-balance the incoming calls
across three servers.
If you need to do this with a view to queue utilisation, an outside
process can be set up to mediate this via the Manager API and provide
this information to the proxy process in real time.
A proxy can
- Forrest Beck [EMAIL PROTECTED] escribió:
Has anyone seen or know of a iphone/ipod sip client that may be in the
works?
http://www.voip-info.org/wiki/view/Apple+iPhone+%252FiPod+Touch+and+SIP+:+SIPHON
Regards,
--
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio
I have used RF.com with my iPhone. Works well.
Sent from my iPhone
Eric Moniz
On Sep 26, 2008, at 10:11 PM, Forrest Beck
[EMAIL PROTECTED] wrote:
Has anyone seen or know of a iphone/ipod sip client that may be in
the works?
___
-- Bandwidth
But will this allow the proxy to handle a load of 300 simultaneous calls? I
mean will the calls be sent off to other asterisk servers and the proxy be
left load-free to route new calls?
--
Dr. Haider Raza
BM 5203
3508 North West 114 Av.
Doral, Florida 33178
Mobile+(809)-659-0623
On Fri,
Proxies do not handle media, so, one can definitely handle 300
simultaneous calls.
Haider Raza wrote:
But will this allow the proxy to handle a load of 300 simultaneous
calls? I mean will the calls be sent off to other asterisk servers and
the proxy be left load-free to route new calls?
I guess what I want to ask is...how do I setup a proxy? In a nutshell...how
are calls transfered or handed off to other asterisk servers leaving the
originating server free from all call handling once the transfer is done.
What dialplan command would do that? Do I setup a trunk and then Dial the
Asterisk is not a SIP proxy. You would have to use another piece of
software, such as Kamailio/OpenSIPS (formerly OpenSER).
Haider Raza wrote:
I guess what I want to ask is...how do I setup a proxy? In a
nutshell...how are calls transfered or handed off to other asterisk
servers leaving
I will now look into reinvites and openser. Thank you so much for your time
and all the excellent advice.
--
Dr. Haider Raza
BM 5203
3508 North West 114 Av.
Doral, Florida 33178
Mobile+(809)-659-0623
On Fri, Sep 26, 2008 at 11:59 PM, Alex Balashov
[EMAIL PROTECTED]wrote:
Asterisk is not
79 matches
Mail list logo
