Hello,
> I'm playing with the optional URL parameter of the Dial() command, which
> "will also be sent to the called party upon successful connection, if the
> channel technology supports the sending of URLs in this way."[1]
>
> Basically, the following asterisk dialplan directive :
>
> - -
Hello,
I'm playing with the optional URL parameter of the Dial() command, which "will
also be sent to the called party upon successful connection, if the channel
technology supports the sending of URLs in this way."[1]
Basically, the following asterisk dialplan directive :
- - 8< - - 8< - -
Hi,
I am using Asterisk 11.5.1. As far as I understood, the following
configuration allows a sip client only to receive calls (type=peer) but
not to place calls
(http://www.voip-info.org/wiki/view/Asterisk+sip+type). Why can I place
calls though with this config?
sip.conf
...
[thorsten]
On Tue, Sep 3, 2013 at 8:11 AM, Thorsten Göllner t...@ovm-group.com wrote:
Hi,
I am using Asterisk 11.5.1. As far as I understood, the following
configuration allows a sip client only to receive calls (type=peer) but not
to place calls
Thanks a lot. Seems to be a good hidden page, isn't it? ;-)
Am 03.09.2013 14:30, schrieb Steve Totaro:
On Tue, Sep 3, 2013 at 8:11 AM, Thorsten Göllner t...@ovm-group.com
mailto:t...@ovm-group.com wrote:
Hi,
I am using Asterisk 11.5.1. As far as I understood, the following
Hi,
I'm looking for SIP client that supports T.38 Fax other than zoiper.
Please advise at earliest.
--
Regards,
Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
Steve Totaro wrote:
Just use a SIP client on your phone. Many providers have multiple
failover paths for inbound calls.
This thread morphed from a nice home phone system into something
completely different.
Yup.
For my situation, DISA is pointless except for road warriors who
call
On Thu, Aug 25, 2011 at 2:27 AM, Per Jessen p...@computer.org wrote:
Steve Totaro wrote:
Just use a SIP client on your phone. Many providers have multiple
failover paths for inbound calls.
This thread morphed from a nice home phone system into something
completely different.
Yup.
On Thu, Aug 25, 2011 at 3:06 AM, Steve Totaro
stot...@totarotechnologies.com wrote:
On Thu, Aug 25, 2011 at 2:27 AM, Per Jessen p...@computer.org wrote:
Steve Totaro wrote:
Just use a SIP client on your phone. Many providers have multiple
failover paths for inbound calls.
This
Steve Totaro wrote:
Steve, even if I could get SIP clients for our phones, doesn't this
mean using a data connection rather than just voice? That would make
it a lot pricier than the current setup with DISA (which is largely
free).
/Per Jessen, Zürich
A Wifi connection? I guess that
On Thu, Oct 28, 2010 at 1:42 AM, Jonas Kellens jonas.kell...@telenet.bewrote:
On 10/28/2010 12:52 PM, Gordon Henderson wrote:
On Thu, 28 Oct 2010, Jonas Kellens wrote
On 10/28/2010 10:44 AM, Kevin Keane wrote:
I assume that you checked and the remote IP is a legitimate IP phone?
If
Hello,
any more input on this subject ?!
Kind regards,
Jonas.
Original Message
Subject:Re: [asterisk-users] SIP client floods port 5060 and gets
blocked
Date: Thu, 28 Oct 2010 13:42:12 +0200
From: Jonas Kellens jonas.kell...@telenet.be
To: Asterisk Users
Hello,
Is there any reason why an IP-phone would pounder on port 5060 ? My
firewall blocks the public IP because it thinks the remote IP is port
scanning on port 5060.
I think the phone is just registering but for some reason it does this
repeatedly in a very short time.
Oct 28 09:01:48
-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, October 28, 2010 12:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP client floods port 5060 and gets blocked
Hello,
Is there any
On 10/28/2010 10:44 AM, Kevin Keane wrote:
I assume that you checked and the remote IP is a legitimate IP phone?
If not, it could be an attempt to break into your system.
If it is a legitimate IP phone, make sure that the SIP configuration
is correct -- if the SIP authentication fails, you
On Thu, 28 Oct 2010, Jonas Kellens wrote:
On 10/28/2010 10:44 AM, Kevin Keane wrote:
I assume that you checked and the remote IP is a legitimate IP phone? If
not, it could be an attempt to break into your system.
If it is a legitimate IP phone, make sure that the SIP configuration is
On 10/28/2010 12:52 PM, Gordon Henderson wrote:
On Thu, 28 Oct 2010, Jonas Kellens wrote
On 10/28/2010 10:44 AM, Kevin Keane wrote:
I assume that you checked and the remote IP is a legitimate IP phone? If
not, it could be an attempt to break into your system.
If it is a legitimate IP
hello,
is there any facility to get SIP client (ex. softphone,ipphone) MAC address
on asterisk.
based on that we authenticated client in anyway.
i tried with sip debug but i didn't got any MAC address related field in all
packets.
regards
Dhaval
Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP client MAC address.
hello,
is there any facility to get SIP client (ex. softphone,ipphone) MAC
address on asterisk.
based on that we authenticated client in anyway.
i tried with sip debug but i didn't got any MAC address related
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] SIP client MAC address.
hello,
is there any facility to get SIP client (ex. softphone,ipphone) MAC address
on asterisk.
based on that we authenticated client in anyway.
i tried with sip debug
think your screwed.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Wednesday, 28 October 2009 4:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP client MAC address
:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA
*Sent:* Wednesday, 28 October 2009 4:47 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] SIP client MAC address.
hello david
This is a very strange discussion.
MAC addresses can only be discovered for peers that are on the same
broadcast segment - which is the realm within which ARP lookups
participate.
Any peers not on the same logical Layer 2 network are reached through
a Layer 3 hop. MAC addresses behind that
hello,
I have SIP phone registered with my server
now if they send me any number for dialing then i want to give a response
code
actually this number is conference number and i need to chek via DB query
that this conference is valid or not
if conference is not valid then i want to send a
Have you tried using another phone and compare the results ?
- Original Message -
From: Chris Mason (Lists) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 16, 2007 2:34 PM
Subject: [asterisk-users] Sip
I have a remote user with Eyebeam on a laptop. Internet connectivity
seems good, there is no packet loss to that location from the PBX.
Everytime the user starts eyebeam, the application tries to register.
Asterisk accepts the registration but the reply never gets to the client
application, so
On 5 Oct 2006, at 21:05, Joe wrote:
Actually the xclients could run the SIP clients but they have firewall
restrictions.
Use IAX on the xclients (I'm semi-serious)
Tim Panton
www.mexuar.com
___
--Bandwidth and Colocation provided by
On Thu, Oct 05, 2006 at 03:05:40PM -0500, Joe wrote:
Actually the xclients could run the SIP clients but they have firewall
restrictions.
I want to SSH to the machines which aren't behind the firewall and
pull the SIP client interfaces back via X Windows.
ssh is a very poor choice for such
Actually the xclients could run the SIP clients but they have firewall
restrictions.
I want to SSH to the machines which aren't behind the firewall and
pull the SIP client interfaces back via X Windows.
On 10/4/06, Tim Panton [EMAIL PROTECTED] wrote:
On 4 Oct 2006, at 18:35, Joe wrote:
Hello,
I'm looking for a SIP client that work with Asterisk that will run on
Linux or Solaris and will work with X Windows. I know X won't all the
media to work but I'm really only interested in SIP signaling.
Thanks!
Joe
___
--Bandwidth and
On 4 Oct 2006, at 18:35, Joe wrote:
Hello,
I'm looking for a SIP client that work with Asterisk that will run on
Linux or Solaris and will work with X Windows. I know X won't all the
media to work but I'm really only interested in SIP signaling.
What are you running your X displays on ?
You
The problems with X-Lite 3 are:
- just accepts one SIP registration
- doesnt send video to other X-Lite or eyeBeam versions
- sometimes loses the SIP informations when you reboot the PC
.
Regards
Joao Pereira
Blake Krone wrote:
What's wrong with X-Lite 3.0? I haven't had any issues with it
I've never experienced any of those problems. I can send video to other eyeBeam versions, 3.0 is the only version that supports video on the lite side. I've never lost any SIP information and only one registration isn't a big deal to me. If you need more than one buy the full eyebeam version.
On
Hi,
i have xlite too and it works without any problems.
ps: what about ekiga? (www.ekiga.org)
rich
--- Joao Pereira [EMAIL PROTECTED] wrote:
Hello to all
can someone recommend me a nice SIP client with
video for windows??
I tried X-Lite 3.0 but it's a lousy piece of
software.
Hello to all
can someone recommend me a nice SIP client with video for windows??
I tried X-Lite 3.0 but it's a lousy piece of software.
Does someone knows about a better software?
Regards
Joao Pereira
___
--Bandwidth and Colocation provided by
What's wrong with X-Lite 3.0? I haven't had any issues with it and find it to be one of the best SIP video software choices, and it's free.On 7/27/06, Joao Pereira
[EMAIL PROTECTED] wrote:Hello to all
can someone recommend me a nice SIP client with video for windows??I tried X-Lite 3.0 but it's a
Hi all,
I've a some users on my network, reporting this:
Sjphone is registered , and some times just looses registry in
Asterisk, I don't know if it is expiration ( instead of loosing
registry).
Then to get registered again they need to restart their own PC.
Why could this beeing happening?
-- Original Message --
From: bodra [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Sun, 28 Aug 2005 02:35:01 -0700
Hi all
i am developing a client for the asterisk that controls ur phone from an Xp c#
application
what functions in Asterisk that
Hello!
If I want to build a Sip client application in Java .
What kind of Java Api would I use to connect to the server and to implement
the sip signaling?
Thanks Ale
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
[EMAIL PROTECTED] wrote:
Hello!
If I want to build a Sip client application in Java .
What kind of Java Api would I use to connect to the server and to implement
the sip signaling?
Thanks Ale
___
Asterisk-Users mailing list
Hi ,
I have setup SER and Asterisk on the samebox (behind a NAT) and try to
connect to SER from outside using Xlite.
But I got 405 error.
Here is my setup diagram.
NAT
SIP Client - Internet (PPPoE ---Router
Hi:
We have got SIP clients connecting to our Asterisk fine with a DSL
connection behind router (NAT), but when we bring the Sipura 2000 ATA
to a Rogers Cable connection behind a Netgear router (NAT), the SIP
clients aren't able to reach the Asterisk at all.
We enabled the SIP debug in Asterisk,
9:15 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP client cannot connect to Asterisk
Hi:
We have got SIP clients connecting to our Asterisk fine with a DSL
connection behind router (NAT), but when we bring the Sipura 2000 ATA
to a Rogers Cable connection behind a Netgear
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
put this in sip.conf
;externip = x.x.x.x; Address that we're going to put in outbound
SIP messages ( official ip address)
;localnet=10.0.0.0/255.255.255.0; if we're behind a NAT
add
nat=yes to every sip account which is behind of NAT
BR Hanson
Unless Symbian has branched off of cell phones, I doubt it. SIP on a
cell phone right now doesn't make sense.
well
running GSM or some fancy codec over GPRS or UMTS may well make sense :)
roy
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi Dean -
Noah, what client were you using on your treo for this 600ms voip call?
Oh, I wasn't using a SIP client (is there one for palm?). Sorry if
that was misleading - this is just web browsing and email. Once the
connection gets going, it is able to do the 2.2 KB/s that standard GPRS
On 09/12/2004 05:54 Steven Critchfield said the following:
Unless Symbian has branched off of cell phones, I doubt it. SIP on a
cell phone right now doesn't make sense.
the nokia 9500 (communicator) as sold in asia uses symbian and has built-in
802.11b. i can see a software SIP phone here being
Hi,
I'm looking for a SIP client for Symbian
OS...
Someone known one? (free or not)
Tks
Andre Remor
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.289 / Virus Database: 265.4.7 - Release Date: 07/12/04
___
On Wed, 2004-12-08 at 19:44 -0200, Andre Remor wrote:
Hi,
I'm looking for a SIP client for Symbian OS...
Someone known one? (free or not)
Unless Symbian has branched off of cell phones, I doubt it. SIP on a
cell phone right now doesn't make sense.
--
Steven Critchfield [EMAIL PROTECTED]
Sprint's PTT uses SIP.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, December 08, 2004 4:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Client for Symbian
On Wed
08, 2004 4:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Client for Symbian
On Wed, 2004-12-08 at 19:44 -0200, Andre Remor wrote:
Hi,
I'm looking for a SIP client for Symbian OS...
Someone known one? (free or not)
Unless Symbian has
On Wed, 8 Dec 2004, Steven Critchfield wrote:
On Wed, 2004-12-08 at 19:44 -0200, Andre Remor wrote:
I'm looking for a SIP client for Symbian OS...
Someone known one? (free or not)
Unless Symbian has branched off of cell phones, I doubt it. SIP on a
cell phone right now doesn't make
On Wed, Dec 08, 2004 at 03:54:10PM -0600, Steven Critchfield wrote:
On Wed, 2004-12-08 at 19:44 -0200, Andre Remor wrote:
Hi,
I'm looking for a SIP client for Symbian OS...
Someone known one? (free or not)
Unless Symbian has branched off of cell phones, I doubt it. SIP on a
cell phone
On Wed, 2004-12-08 at 17:16 -0500, dean collins wrote:
Steven, I think it makes total sense. I'm currently in the process of
trying to source a better solution than my treo 600 in order to stay in
contact (looking at either ppc or xp with cf/gsm adaptor)
Anything that will enable me to
I'm looking for a SIP client for Symbian OS...
Someone known one? (free or not)
Unless Symbian has branched off of cell phones, I doubt it. SIP on a
cell phone right now doesn't make sense.
Steven, I think it makes total sense. I'm currently in the process of
trying to source a better solution
Take a look a JSR 180 for MIDP Cellphones.
google: j2me sip jsr 180
finds lots of links,
including a Nokia reference implementation.
But from what I can see it only addresses the SIP protocol itself,
not the media streams or how to play them.
Tim.
___
Noah, what client were you using on your treo for this 600ms voip call?
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah
Miller
Sent: Wednesday, December 08, 2004 5:50 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] SIP Client
, 2004 5:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] SIP Client for Symbian
On Wed, 2004-12-08 at 17:16 -0500, dean collins wrote:
Steven, I think it makes total sense. I'm currently in the process of
trying to source a better solution than my treo
I think it's about 600ms under ideal conditions. I normally see
around 1000ms or higher on Tmo GPRS. EDGE will not aid in latency.
We'll have to wait for UMTS with possible HSDPA to see around 200ms,
or so I hear.
On Wed, 8 Dec 2004 17:49:58 -0500, Noah Miller [EMAIL PROTECTED] wrote:
I
I have an xlite client registering with asterisk. The client is on a
separate subnet routed via a site to site VPN tunnel to the subnet
hosting the * server, with no active firewall between the subnets.
With one IP address (192.168.3.56) set on the client the * box ignores
the SIP REGISTER
On Mon, 12 Jul 2004 11:42:57 -0700, Dameon D. Welch-Abernathy
[EMAIL PROTECTED] wrote:
Through my Asterisk server, I am trying to use IAXTel to dial 800-type
numbers (yes, I know I can do the same thing with FWD and others via
SIP, but I wanted to play with IAX a little). It appears I'm running
Through my Asterisk server, I am trying to use IAXTel to dial 800-type
numbers (yes, I know I can do the same thing with FWD and others via
SIP, but I wanted to play with IAX a little). It appears I'm running
into some sort of a codec mismatch or something because it's not working
right. The SIP
Good day.
I'm looking for a sip client 4 fedora???
Frdora?
Thanks
Altus
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Dear list,
does any one know how to do a SIP client auth via central database instead
of specifying in the sip.conf ?
if we could do with central database, should we use RADIUS or other better
way to do it.
Thanks,
George
___
Asterisk-Users mailing
Hi,
Yes Telesym, xten and one more I can't remember the name of it, they are all
for PPC-only. :(
/HHA
From: Ray Burkholder [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] SIP-Client for Handheld PC
Date: Mon, 12 Jan 2004 14:33:39 -0500
What
Anyone know a sip-client that will work on a Handheld PC running WINCE for
HPC.
I can find some for PocketPC, but the wont work on my HPC
??
/HHA
_
Scope out the new MSN Plus Internet Software optimizes dial-up to the max!
PROTECTED] On Behalf Of
Hans-Henrik Andresen
Sent: January 12, 2004 05:01
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIP-Client for Handheld PC
Anyone know a sip-client that will work on a Handheld PC
running WINCE for
HPC.
I can find some for PocketPC, but the wont work on my
PROTECTED]
http://www.oneunified.net
704 644 6999 x2002
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hans-Henrik Andresen
Sent: January 12, 2004 05:01
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIP-Client for Handheld PC
Anyone know a sip
-Users] SIP-Client for Handheld PC
Anyone know a sip-client that will work on a Handheld PC
running WINCE for
HPC.
I can find some for PocketPC, but the wont work on my HPC
Y'know reflecting upon this, and giving consideration to my once pricey
Toshiba e740 PPC with wifi, I can't think
xlite saying login timed out. contact network admin.
how to get rid of this. * is not behind NAT.
also, the grandstream SIP phone also seems to fail to register. IAX phones
are all ok.
DIAX works fine
___
Asterisk-Users mailing list
[EMAIL
Thanks very much !!
I thinks it could be very useful for me
Regards
Rattana
- Original Message -
From: Peer Oliver schmidt [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 29, 2003 7:14 PM
Subject: Re: [Asterisk-Users] SIP client
Christopher Stephens schrieb
hi everybody,
Is there SIP client which work with Asterisk and
can be embedded in a HTML page ?
Thanks
Rattana
On Wed, 29 Oct 2003 09:58:28 +0100, Rattana BIV wrote:
hi everybody,
Is there SIP client which work with Asterisk and can be embedded in a HTML page ?
Thanks
Rattana
hehe, why use a SIP client, why not a client which does IAX and bury
that in a web page ?
(yes, I haven't answered the
: Rattana BIV [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Date: Wed, 29 Oct 2003 09:58:28 +0100
Subject: [Asterisk-Users] SIP client
hi everybody,
Is there SIP client which work with Asterisk and can be embedded in a
HTML page ?
Thanks
Rattana
___
Asterisk
Christopher Stephens schrieb:
Is there SIP client which work with Asterisk and can be embedded in a
HTML page ?
It may not be *exactly* what you're looking for, but try:
http://fwd.pulver.com/callme.php?userid=411
[..]
Unfortunately this seem to work with Internet Explorer, only.
rgds
pos
Can anybody explain me what does canreinvite=yes really does?
Not sure how technical an answer you want becasue it look slike you know whats going on but as I unterstand it canreinvite=no tells the UA that reinvite is not supported and so causes all the RTP traffic to be routed via the * server..
WipeOut . wrote:
Any ideas on the client A to C (same LAN, same NAT box, unique
outside IP, same * server)?
Only thing that springs to mind is to install another * box
internally and then use IAX to connect the internal * box to the
external one.. then the internal phone will call each other
I have been trying to get SIP UA work with NAT but i have no been
successful has any one got NATed ATA working(i.e an ATA witha private IP
working with NAT).
Asterisk registers the 192.168.0.3 Ip but no call go through at all,
infact there is no log of any call made on asterisk console.
can
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