I have an HTC P4350 so far..With Windows Mobile 6.1
And if you install Sip Config Tool, phone is able to get calls from SIP
account, and optionally it can dial thru SIP account if GSM is not
available(or all the time). This tool is only available if 3G or wireless
is connected.
On Tue, 24 Mar
Our work around is to lower the registration expiration on the phones.
Under account settings in the web interface on the phones, we reduced
the Register Expiration from 60 minutes to 15.
This means the phones re-register every 15 minutes...and when they
register the BLF updates. Now when
Out of shear curiosity, do you mind running asterisk -rx show hints | grep
your_exten 16 minutes after you unplugged your phone, as in my example test
below?
Also, in the * CLI or log, do you see a message such as:
Extension Changed 4063[ext-local] new state XXX for Notify User 4062
right
Michael Robertson a écrit :
Anyone connected up to it yet?
http://www.skypeforsip.com/
This service is vaporware. It's just surveyware at this point with no actual
service. An alternative is OpenSky which is a launched service which does
SIP to Skype and Skype to SIP so you can
Stefan Guenther a écrit :
Hello,
is anyone on the list using a normal cell/mobile phone which is able to
act as a SIP client over WLAN?
Or has anyone heard of a SIP client for cell/mobile phones running
windows mobile 6.x?
The phone should use SIP, when the asterisk server is reachable
Deal All Asterisk Expert
If this possible to Create Voice Recording System Beside Main Asterisk PBX?,
so Call be handle by 1 Server and Recording by other server.
1. How to accomplish.
Thanks.
___
-- Bandwidth and Colocation Provided by
Well,
anyone knows a good Skype vs SIP channel or program or something else to
integrate it into an Asterisk machine, to call normal skype users and not
and receive normal skype calls?
I red that Digium and Skype are working to integrate a chan_skype. Anyone
can tell me about?
Bye
Marco
hello,
i'm also interaisted about Sip / Skype Intgration
any News ?
thanks
Marco Sambo a écrit :
Well,
anyone knows a good Skype vs SIP channel or program or something else
to integrate it into an Asterisk machine, to call normal skype users
and not and receive normal skype calls?
I red that
Hi All,
Does anyone know if it is possible to define a different ring tone for
internal/external calls within Asterisk? If so how? We are using
Grandstream 200 and Aastra 55i's
If any one has any links/docs they could forward that would be great.
Cheers
I seem to recall qualify being part of the problem. I'm not qualifying any
extensions. In addition, the BLF I'm doing is for parked lines not for SIP
extensions, so not sure if that's part of the problem either.
I do not recall seeing the notify messages you're asking about, but I believe
The problem is that I cannot put the outboundproxy statement to the
applicable sip extension context, due to the fact that I want to force every
ENUM call to go via the proxy; and ENUM calls don't use any context to leave
asterisk.
Even so, putting outboundproxy statement is in the global section
Dears;
If I am going to user AsteriskNow, how can I use the GUI tool only with my
current running Asterisk which has 1.4 version? I think it will be needless to
install AsteriskNow while already I am using Normal Asterisk.
But I need only the GUI tool from the AsteriskNow, how to take it?
Hi All;
I need to use the recording for the calls, did anyone try this on Asterisk? How
it works?
By the way: Asterisk support recording or it is another module that I have to
download it and install it? Stable?
Regards
Bilal
___
--
Hi List;
I am going to use a post and prepaid billing with my Asterisk vesion 1.4, can
anyone advise me which of the billing software will be more stable and it is
opensource to be used:
ASTCC or a2billing?
Any advise that might help to take the proper decision?
Regards
Bilal
On Wed, Mar 25, 2009 at 8:39 AM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
I need to use the recording for the calls, did anyone try this on Asterisk?
How it works?
By the way: Asterisk support recording or it is another module that I have to
download it and install it? Stable?
Perhaps the Monitor CMD is what you are looking for.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Monitor
Good Luck!!
Jim
bilal ghayyad wrote:
Hi All;
I need to use the recording for the calls, did anyone try this on Asterisk?
How it works?
By the way: Asterisk support
hello bilal,
try a2Billing
is a small asterisk add-on that do the necesary billing steps
otherwise:
trixbox / Elastix / PBXInAFlash have a integrated billing system (if i'm
sur)
thanks
bilal ghayyad a écrit :
Hi List;
I am going to use a post and prepaid billing with my Asterisk vesion 1.4,
After a variety of connectivity problems, my itsp (Unlimitel.ca) blamed the
problem on the IAX protocol. They told me that as of Asterisk 1.4 the IAX
protocol went downhill and many carriers (like VoicePulse) are discontinuing
support for IAX.
Is this correct? We are all heading for SIP?
I have had great luck with OrecX, port mirroring, just make sure your
switch can handle the PPS rating (bandwidth is rarely a problem.
Thanks,
Steve Totaro
On Wed, Mar 25, 2009 at 5:38 AM, joko pitoyo joko.pit...@gmail.com wrote:
Deal All Asterisk Expert
If this possible to Create Voice
hello,
(if this is correct):
IAX is no maintained now
but IAX2 is maintained by the Asterisk Developers Team
(i'm not sur) please si other responces
thanks
OCG Technical Support a écrit :
After a variety of connectivity problems, my itsp (Unlimitel.ca)
blamed the problem on the IAX protocol.
OCG Technical Support wrote:
After a variety of connectivity problems, my itsp (Unlimitel.ca) blamed
the problem on the IAX protocol. They told me that as of Asterisk 1.4
the IAX protocol went downhill and many carriers (like VoicePulse) are
discontinuing support for IAX.
Is this
Side note and very telling. IAX.cc (Vitelity now) advised against
using IAX a long time ago.
That would be the same as CiscoGear.com advising to not use Cisco but 3Com.
Thanks,
Steve
On Wed, Mar 25, 2009 at 9:04 AM, OCG Technical Support supp...@ocg.ca wrote:
After a variety of connectivity
Hi,
In sip.conf, it's possible to add a line such as
setvar=MYFIELD=foo
and access this value from diaplan with SIPPEER function.
1. Which function is available to access values in users.conf such as
vmsecret ?
2. Is it possible to extend users.conf with custom keys/values ?
Regards.
On Wed, Mar 25, 2009 at 9:14 AM, Dr. Michael J. Chudobiak
m...@avtechpulse.com wrote:
OCG Technical Support wrote:
After a variety of connectivity problems, my itsp (Unlimitel.ca) blamed
the problem on the IAX protocol. They told me that as of Asterisk 1.4
the IAX protocol went downhill and
Dears,
- Anyone know how to play an early media as (background song) with
no billing and when the call is connected the song will stop and the billing
starts.
Regards
*
No employee or agent is authorized to conclude any binding
Here is a page that provides a solution to your question:
http://wiki.astripedia.org/index.php/Asterisk_HowTo%27s#Asterisk_Ringtone_id
entification
This is the actual snippet from the page:
Asterisk Ringtone identification
An Asterisk macro using ringtone identification. In companies, it makes
Thanks Meftah;
Any a2billing is open source, correct?
About the others u mentioned, they are post and prepaid billing and open open
source also?
What do u mean by integrated billing that a2billing does not have it?
Regards
Bilal
--- On Wed, 3/25/09, Meftah Tayeb tayeb.mef...@gmail.com
The choice of router/NAT is critical though. Unlimitel recommended the
SnapGear 560 to me, and it eliminated all the issues I was having with
IAX going through my Sonicwall devices.
Just another datapoint for you...
Just curious.
Since IAX only uses ONE port, do you have any idea what the
Thsi Monitor CMD is a part of AsteriskNow so I have to download AsteriskNow and
then take only the Monitor CMD or I can download the Monitor CMD alone? From
where?
Regards
Bilal
--- On Wed, 3/25/09, Steve Totaro stot...@first-notification.com wrote:
From: Steve Totaro
YMMV, but you might try this
Exten = s,1,background(background_song)
Exten = s,n,Answer() ;start billing
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Wednesday, March 25, 2009 8:27 AM
To:
On Wed, Mar 25, 2009 at 9:04 AM, OCG Technical Support supp...@ocg.ca wrote:
After a variety of connectivity problems, my itsp (Unlimitel.ca) blamed the
problem on the IAX protocol. They told me that as of Asterisk 1.4 the IAX
protocol went downhill and many carriers (like VoicePulse) are
What I am meaning is .
I want to start a music on hold and dial the number (009713045212) In the
same time and when the call is connected the music will stop and I will
talk to the called number
Exten = 444,1,--
exten = 444,n,Dial(SIP/OutGoingGateway/009713045212|300|)
is it
Hi all
I have been hacked but no idea how!!! I noticed somebody in Eastern Europe
came from an American IP and tried to call loads of international numbers.
Thankfully I had no credit with my VOIP out provider so the calls went
nowhere. But if I had credit it would all have been used up.
I
Change line 2 to this:
exten = 444,n,Dial(SIP/OutGoingGateway/009713045212|300|m)
this will play moh for 300 seconds or until the other end answers. The only
issue you may have is that some carriers don't generate a proper response
when answering to the music would continue over your
We are not having a problem in the BLF area, and we do qualify our remote
phones. I don't know if qualify does any thing beyond pinging the
address, but perhaps it does carry a data payload too.
Cary Fitch
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Yes, If you are using IAX2 , you could check iax.conf and check for a default
config..
[default] is used when non auth’ed usually.
1-888-372-6501
sa...@contacttel.com
http://www.contacttel.com/ http://www.contacttel.com
From: asterisk-users-boun...@lists.digium.com
We had some carrier suggest we don't use IAX because SIP had a better fail
over capability than IAX.
I don't remember the details. We do use both however, with acceptable
results.
Cary Fitch
_
From: asterisk-users-boun...@lists.digium.com
I use simple port forwarding on an Linux firewall (iptables)...so that's not
the issue.
I was referring to IAX2 of course (IAX has be gone a long time I think)...
Unlimitel is running * 1.4.x (and so am I)...
I just can't understand IAX2 connections suddenly dropping (on one day)
being protocol
On Wednesday 25 March 2009 04:55:51 Marco Sambo wrote:
anyone knows a good Skype vs SIP channel or program or something else to
integrate it into an Asterisk machine, to call normal skype users and not
and receive normal skype calls?
I red that Digium and Skype are working to integrate a
On Wednesday 25 March 2009 09:59:27 OCG Technical Support wrote:
I use simple port forwarding on an Linux firewall (iptables)...so that's
not the issue.
I was referring to IAX2 of course (IAX has be gone a long time I think)...
Unlimitel is running * 1.4.x (and so am I)...
I just can't
On Wednesday 25 March 2009 08:19:31 Olivier wrote:
In sip.conf, it's possible to add a line such as
setvar=MYFIELD=foo
and access this value from diaplan with SIPPEER function.
1. Which function is available to access values in users.conf such as
vmsecret ?
2. Is it possible to extend
It was probably Voice pulse that suggested we not use IAX, and we are
getting an IAX error at this time on another connection where we do use it.
The error is:
[Mar 25 05:46:16] WARNING[5102]: chan_iax2.c:1056 __send_lagrq: I was
supposed to send a LAGRQ with callno 9779, but no such call exists
I screen all incoming calls for unknown, asterisk, anonymous,restricted
and perhaps others that don't immediately come to mind. Incoming calls
don't lead either to any termination that would cost me.
You should be smart enough to set up a way to use when you are out and
about, keeping in mind
Hello Cary,
Can I see your configuration files in any form, I mean only part of BLF
settings
2009/3/25 Cary Fitch ca...@usawide.net
We are not having a problem in the BLF area, and we do qualify our remote
phones. I don't know if qualify does any thing beyond pinging the
address, but
Really nothing elaborate to this part: We have the dual entries because some
extensions are/were treated as 3 digit extensions and some as full
NPANXX. Better safe than sorry.
These Hints are in the Context handling these phone numbers. (They have to
be.)
;Provides LED Lights on phones
Dear list,
I've set up an RPM repository with several asterisk-related RPMs that I
think contain some improvements upon what are already out there. The
first goal is to be able to build an Asterisk + FreePBX system on CentOS
5 with the EPEL repo enabled; in our environment, where all our
am i right in understanding that this feature is called color ring back
tone?
On Wed, Mar 25, 2009 at 8:16 PM, Danny Nicholas da...@debsinc.com wrote:
Change line 2 to this:
exten = 444,n,Dial(SIP/OutGoingGateway/009713045212|300|m)
this will play moh for 300 seconds or until the other
Daniel wrote:
For us, opensky can be OK for individual users, not for allowing
Asterisk users to call Skype users. Why? Simply that when you buy the 20
USD connection to Skype and don't want your calls to be cutted after 5
mn, you have to use the Gizmo Skype aliases system which is in your
Other than the price (nearly £150 difference), is there any particular
reason not to pick an OpenVox A400-based solution for my UK Asterisk needs?
I have only a single line (so 1x FXO) and use SIP phones exclusively on-site
(no FXS needed). I need to replace my current AX-100P card, as my new
Marco wrote:
anyone knows a good Skype vs SIP channel or program or something else to
integrate it into an Asterisk machine, to call normal skype users and not
and receive normal skype calls?
OpenSky can be setup for free to allow any Asterisk system to call Skype
users. Setup instructions for
It's possible. My understanding of it is just as another feature of Dial.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kinjal Dixit
Sent: Wednesday, March 25, 2009 12:15 PM
To: Asterisk Users Mailing List -
El mié, 25-03-2009 a las 10:28 -0700, Michael Robertson escribió:
OpenSky can be setup for free to allow any Asterisk system to call
Skype users. Setup instructions for Asterisk are at:
http://www.gizmo5.com/opensky Free calls are available up to 5
minutes. If you need longer calls there's a
On Wednesday 25 March 2009 12:23:31 Asterisk wrote:
Other than the price (nearly £150 difference), is there any particular
reason not to pick an OpenVox A400-based solution for my UK Asterisk needs?
You're better off going with a TDM410-based solution, rather than
TDM400-based, which the A400
How about letting skypers call into one's SIP system?
Imagine a web audio stream that wants to take calls, but whose originator
has a well-developed SIP infrastructure...
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Guillermo Salas M. a écrit :
El mié, 25-03-2009 a las 10:28 -0700, Michael Robertson escribió:
OpenSky can be setup for free to allow any Asterisk system to call
Skype users. Setup instructions for Asterisk are at:
http://www.gizmo5.com/opensky Free calls are available up to 5
minutes. If
El mié, 25-03-2009 a las 19:09 +0100, Administrator TOOTAI escribió:
Can be used to receive calls from skype?
Yes
Great,and how? Have you any link to read?
Regards,
--
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esquina
Edificio Barre #2 Primer Piso
Telefono : +593 5
I have a client that wants to put a phone on their web page for customers
to call them via their Asterisk server.
) A keypad is needed to enter credit card details.
) Speed dial buttons like Tech Support, Sales, etc. are a
requirement. Actually, passing the SIP address in the HTTP link would
On Wednesday 25 March 2009 10:45:59 Cary Fitch wrote:
It was probably Voice pulse that suggested we not use IAX, and we are
getting an IAX error at this time on another connection where we do use it.
The error is:
[Mar 25 05:46:16] WARNING[5102]: chan_iax2.c:1056 __send_lagrq: I was
supposed
Is your client completely committed to using a web-based softphone and
requiring them to make sure they have speakers turned on and a
microphone plugged in? I think it's a fair guess that if they have a CC
they have a phone which makes some sort of click to call technology
more attractive to
Since you're hosting this page and it is secured, why don't you just create
2 Gizmo accounts for each department and have the browser do a call from
gizmo-a to gizmo-b? Then make your speed dial buttons do like this:
1. a-b
2. c-d
3. e-f
Etc.
Perhaps I'm not thinking this through enough, but
Hi John,
On Thu, Mar 26, 2009 at 12:35:45AM +0800, John Morris wrote:
I've set up an RPM repository with several asterisk-related RPMs
that I think contain some improvements upon what are already out
there. [...]
I'm quite interested to get feedback on these RPMs, both on the need
for such
Steve Edwards wrote:
I have a client that wants to put a phone on their web page for
customers to call them via their Asterisk server.
) A keypad is needed to enter credit card details.
) Speed dial buttons like Tech Support, Sales, etc. are a
requirement. Actually, passing the SIP
On Wed, 25 Mar 2009, Steve Edwards wrote:
I have a client that wants to put a phone on their web page for customers
to call them via their Asterisk server.
) A keypad is needed to enter credit card details.
) Speed dial buttons like Tech Support, Sales, etc. are a
requirement. Actually,
I'm thinking a PERL solution, because that's the primary thing I do. You
would take the input from the webpage, pass it to an AGI that opened a new
web window to make the call and pass the connection back to your original
window.
Another thought on that thread; could you make the window a
On Wed, 25 Mar 2009, Asterisk wrote:
Other than the price (nearly £150 difference), is there any particular
reason not to pick an OpenVox A400-based solution for my UK Asterisk needs?
None whatsoever.
I think the new digium cards are better at interrupt sharing, but if
that's not an issue
From: Guillermo Salas M. gsa...@manta.telconet.net
http://www.gizmo5.com/opensky Free calls are available up to 5
minutes. If you need longer calls there's a commercial service you can
purchase.
Can be used to receive calls from skype?
Yes it can. For example anyone who calls me now on Skype
Both Montior and MixMonitor are part of the standard Asterisk
distribution. There is no need to download anything else.
bilal ghayyad wrote:
Thsi Monitor CMD is a part of AsteriskNow so I have to download AsteriskNow and
then take only the Monitor CMD or I can download the Monitor CMD alone?
On Wed, 25 Mar 2009, Steve Edwards wrote:
I have a client that wants to put a phone on their web page for customers
to call them via their Asterisk server.
) A keypad is needed to enter credit card details.
) Speed dial buttons like Tech Support, Sales, etc. are a
requirement. Actually,
Thanks Gordon for your suggestions and advices. I changed the passwords same
day, and was monitoring my system very closely. I also use a non standard
port for SSH, and also plan to move my SIP port to a non standard one too in
future. At this time things are ok, but I know that this problem is
Hi,
I want to send hangup command to the call which was logged in earlier via cli.
Lets say to '5aec0e7207b24c8e1bdb511a460f7...@callcentric.com
Basically I want to hang up the call when ever I want but from the script.
Thanks,
Singh
___
--
Hi,
I want to send hangup command to the call which was logged in earlier via cli.
Lets say to '5aec0e7207b24c8e1bdb511a460f7...@callcentric.com
Basically I want to hang up the call when ever I want but from the script.
Thanks,
Singh
___
--
On Wed, Mar 25, 2009 at 2:56 PM, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com wrote:
On Wednesday 25 March 2009 10:45:59 Cary Fitch wrote:
It was probably Voice pulse that suggested we not use IAX, and we are
getting an IAX error at this time on another connection where we do use it.
The
Hi Singh,
Have you tried soft hangup?
Andy
On Wed, Mar 25, 2009 at 4:38 PM, Singh Saimbhi singh.saim...@palm.com wrote:
Hi,
I want to send hangup command to the call which was logged in earlier via
cli. Lets say to '5aec0e7207b24c8e1bdb511a460f7...@callcentric.com
Basically I want to
Hello Andy,
I am using Net::Telnet to setup a session and using Manger API to call out:
$session-print(h2s(
Action = Originate,
Extension = s,
Context= $options{context},
Channel= $options{channel}/$number,
Steve Totaro wrote:
IAX2 has been a lemon since it's inception. Sure, some people have
success. It seems to work OK for IAXModem.
I chose to use IAX2 in developing IAXmodem because IAX2 is relatively
simple compared to SIP and because at the time I didn't know of any
easy-to-use SIP
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