I answer myself: since Asterisk 1.6 you can use the SIPPEER function to
retrieve the peer's setvar variables.
regards
klaus
Klaus Darilion schrieb:
Hi!
Using setvar in a peer configuration (sip.conf) I can set the channel
variables for the incoming channel. Is there a similar method which
Hi!
When I enter the Asterisk console and press the up key I get the
command history. But the quit is twice in the history. Why?
thanks
klaus
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25
Hi,
I tried to apply both patchs, but there isn't improvements.
Finally, I have implemented the idea of Stefan. I use another B2B
equipment which create a new session after the first passage in
Asterisk and before the second passage. It's not very clean but it
works ...
Thanks for help :)
Hi, thanks for the reply ,
- Original Message -
From: Philipp Kempgen [EMAIL PROTECTED]
Maybe something is broken in recent versions of chan_iax2.c?
http://lists.digium.com/pipermail/asterisk-users/2008-September/218560.html
Not the same issue though.
I doubt it. It has been
We have been using asterisk for a while now but have recently needed to
install a second server in a remote office and set up a iax trunk
between the 2 servers. The dial plan seems to work well when I tested it
on the same LAN. However this afternoon I connected the system at the
remote office and
Hi Bilal,
On Tue, Sep 23, 2008 at 11:11 PM, bilal ghayyad [EMAIL PROTECTED] wrote:
Dear Philippe;
Thanks a lot for ur kindly answer.
How can I use the Radius with CDR (Accounting)?
Here is the documentation :
http://svn.digium.com/view/asterisk/branches/1.4/doc/radius.txt?view=markup
You can use the ResetCDR() application with the w flag in it after you get
the unavailable, busy or etc message from the callee. It will store the cdr
of that call and after forwarding to mobile, that cdr will be dumped again.
On Wed, Sep 24, 2008 at 8:26 AM, Nhadie [EMAIL PROTECTED] wrote:
hi,
Hi all,
Sorry to interrupt. I need some help regarding fax passthru mode.
We are trying to configure fax passthru mode in asterisk using sip. For out
of network calls/fax we use trunk configuration. i am using asterisk
1.4.2.The user has to use fax machine connected to their ata and dial
the
You maybe using wrong username. If the user is defined in sip, you should be
able to register using the correct username and password. Also, see if
asterisk is listening on a defferent sip port instead of default 5060. If
its different use that port.
On Wed, Sep 24, 2008 at 3:32 AM, michel freiha
On Wed, Sep 24, 2008 at 4:45 AM, Martin Seebach [EMAIL PROTECTED] wrote:
Hi, thanks for the reply,
- Original Message -
From: Philipp Kempgen [EMAIL PROTECTED]
Maybe something is broken in recent versions of chan_iax2.c?
We have done it too. www.axvoice.com
On Tue, Sep 23, 2008 at 3:39 PM, Benjamin Jacob [EMAIL PROTECTED]wrote:
Hi Bilal,
Yes it is definitely possible. And I've done it myself for a couple of our
clients.
Does that answer your two questions?
cheers
- Ben.
--- On Tue, 9/23/08, bilal
you must share your configuration with us. otherwise we cant even make a
wild guess.
On Mon, Sep 22, 2008 at 7:48 PM, Cindy Tan [EMAIL PROTECTED] wrote:
may i noe wad can i do because my asterisk is working fine but the calls
cannot proceed between 2 asterisk servers.
hope anyone can help me
In article [EMAIL PROTECTED],
Nathan Dennis [EMAIL PROTECTED] wrote:
We have been using asterisk for a while now but have recently needed to
install a second server in a remote office and set up a iax trunk
between the 2 servers. The dial plan seems to work well when I tested it
on the same
Hi,
- Steve Totaro wrote:
Do a show codecs.
It looks right. ulaw is loaded, and that's the only thing I allow, on both SIP
and IAX.
Maybe IAX2 is not loaded.
Looks like it:
filserver*CLI module show like iax2
Module Description Use Count
chan_iax2.so Inter Asterisk eXchange (Ver 2)
Greetings list,
I've had problems on a few of our asterisk boxes where voicemail tends to cut
out after about 30 seconds, despite the maximum message length being set at
240s (4m).
I've tried reducing the silence detection threshold from 128 down to 32, which
has helped, but not resolved the
Hi,
On Tue, Sep 23, 2008 at 01:05:17PM -0600, Joseph wrote:
I need to upgrade my Asterisk, currently I'm using 1.2.27 from Gentoo portage
but I think this version has a problem with RFC2833 DTMF signaling and I
don't think there
will be any newer version available anytime soon on portage.
On Tue, Sep 23, 2008 at 1:48 PM, Thanos Koukoulis [EMAIL PROTECTED] wrote:
On Tue, Sep 23, 2008 at 1:19 PM, Gergo Csibra [EMAIL PROTECTED] wrote:
Tuesday, September 23, 2008, 11:57:00 AM, Thanos wrote:
Hello
I have just set up Asterisk Asterisk 1.4.21.2 on a CentOS 5.2 machine.
I am
Hi,
We saw this between Asterisk and an Audiocodes gateway. Whilst the
voicemail is being recorded asterisk is not sending *ANY* rtp. Silence
detection will always detect silence if it listens to this side of the
conversation. Adjusting the threshold wont work, you need to find the
You can use the gentoo voip overlay. Asterisk 1.4.21.2 is included in
the overlay.
# emerge layman
# layman -a voip
You may need to modify some of the .ebuild files, or your
/etc/portage/packages.unmask depending on your asterisk build.
Regards,
Chris
Hello,
I'm implementing a VoIP client and using Asterisk 1.4. The RTP transfer
should be handled in a direct connection from client to client. But the
Asterisk server does not reveal the IP address of the peer in the
contact header field of a SIP request nor in the connection header field
of
Hello,
I have nearly the same issue. Does anyone have a suggestion as to how
to find and fix this problem?
Mark
On Jul 16, 2008, at 10:59 AM, Mike (Asterisk) wrote:
[zaptel]
span=1,0,0,esf,b8zs
At least one of your spans should be getting it's timing from your
service provider. It
hi,
when a user register on my asterisk i can see it adding Noop for that
extension, but after awhile i won't see it anymore:
what are the reasons for it being removed on the dynamic context?
one thing i found when i unregister it's removed.
dialplan show myregcontext
[ Context
On Wed, 24 Sep 2008, Klaus Darilion wrote:
Hi!
When I enter the Asterisk console and press the up key I get the
command history. But the quit is twice in the history. Why?
Who knows - I suspect because it's always been like that and no-ones
bothered to report it :)
Gordon
On Wed, 24 Sep 2008, Chris Bagnall wrote:
Greetings list,
I've had problems on a few of our asterisk boxes where voicemail tends
to cut out after about 30 seconds, despite the maximum message length
being set at 240s (4m).
I've tried reducing the silence detection threshold from 128 down
canreinvite=yes
Atenciosamente,
Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Convergent Technologies Core
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brazil
+55 54 2104-7000
- Arno Scholz [EMAIL
I wonder which timeout will apply here: the one in master context or one from
the slave context?
[master]
exten=100,1,Dial(Local/[EMAIL PROTECTED], 20)
[slave]
exten=100,1,Dial(SIP/100, 30)
Thanks
Vadim
___
-- Bandwidth and Colocation Provided by
On Wed, Sep 24, 2008 at 5:43 AM, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
Sorry to interrupt. I need some help regarding fax passthru mode.
We are trying to configure fax passthru mode in asterisk using sip. For out
of network calls/fax we use trunk configuration. i am using asterisk
Dear Asterisk Users,
We have uploaded a bunch of videos on Astricon 08 Day 1
Interview with Mark Spencer on big announcement Astricon
http://www.youtube.com/watch?v=nzEeIEuQvf4
Interview with Allison Smith (See allison in a white dress with Asterisk
shoes with asterisk on it :)
Thank you for your reply Sir
i tried inserting ResetCDR(w) almost everywhere but i still end up with
this on the cdr:
FromTo:
500 100-CHANUNAVAIL
This is my current setting hat produces that CDR:
exten = 100,1,Macro(dial-ext-cf|SIP/${EXTEN}|vm-100|moh-100)
exten =
On 09/24/08 14:12, Chris Bagnall wrote:
You can use the gentoo voip overlay. Asterisk 1.4.21.2 is included in
the overlay.
# emerge layman
# layman -a voip
You may need to modify some of the .ebuild files, or your
/etc/portage/packages.unmask depending on your asterisk build.
Regards,
Let me know if I should post this on the asterisk-dev list instead.
I am building a Linux-Vserver (http://www.linux-vserver.org) host system
that will have several guests running Asterisk. Since the guests can't
load kernel modules or do other dangerous stuff, but can access them I
built
-- Executing [EMAIL PROTECTED]:1] Set(SIP/21-081ceea8,
CALLERID(all)= 88821268) in new stack
-- Executing [EMAIL PROTECTED]:2] Dial(SIP/21-081ceea8,
IAX2/88821268/40618405|30|r) in new stack
[Sep 11 12:05:58] WARNING[7098]: app_dial.c:1202 dial_exec_full:
Unable to
Thanks for pointing that out Tony, Should have included that in my first
post.
Below is the version and the IAX config for each end
Server 1
Version : 1.4.18
IAX2.conf peer details
[brisbane]
type=friend
host=XXX.XXX.XXX.XXX
trunk=yes
context=internal
context=parkinglot
qualify=1
On 09/24/08 13:50, Artem Makhutov wrote:
Hi,
On Tue, Sep 23, 2008 at 01:05:17PM -0600, Joseph wrote:
I need to upgrade my Asterisk, currently I'm using 1.2.27 from Gentoo
portage but I think this version has a problem with RFC2833 DTMF signaling
and I don't think there
will be any newer
1.6 = Windows Vista :-P
On Tue, Sep 23, 2008 at 3:05 PM, Joseph [EMAIL PROTECTED] wrote:
I need to upgrade my Asterisk, currently I'm using 1.2.27 from Gentoo
portage but I think this version has a problem with RFC2833 DTMF signaling
and I don't think there
will be any newer version
I would think
1.6 = Windows Vista
:)
PaulH
Steve Totaro wrote:
1.6 = Windows Vista :-P
On Tue, Sep 23, 2008 at 3:05 PM, Joseph [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I need to upgrade my Asterisk, currently I'm using 1.2.27 from
Gentoo portage but I think this
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 barely used but it does have some crucial programs i need to
run in windows
Do you have ztdummy loaded in the VM?
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael J.
Liberatore
Sent: Wednesday, September 24, 2008 8:28 PM
To: Asterisk Users Mailing List -
I just installed *-1.4 and when I enter mail extension it keep asking me for
Mailbox #
I have in sip.conf under my extension mailbox=11 type=friend
*-1.2 was jumping straight to messages.
What did change?
--
#Joseph
___
-- Bandwidth and Colocation
Hi,
Does anyone know what happens if you exceed your G729 license
capacity? Lets say you have 10 of 10 licenses being used by a PBX,
then an 11th call comes in set up to use G729.
Does asterisk has the ability to stop offering that codec in the SDP
once the capacity is reached.
Robert
Mike,
Buy an asterisk appliance like http://www.taa.com/products-vdex-40.html
problem solved.
If you are worried about good call quality it's either a dedicated pc or
a dedicated appliance, one or the other.
Cheers,
Dean
From: [EMAIL
Hey Robert,
In my experience you get dead silence and the call goes through. We
run 1.4, it might be different for different setups.
On Wed, Sep 24, 2008 at 9:21 PM, Robert McNaught [EMAIL PROTECTED] wrote:
Hi,
Does anyone know what happens if you exceed your G729 license
capacity? Lets say
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: Wednesday, September 24, 2008 6:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] NVFaxDetect (1.0.6), NVBackgroundDetect was: Asterisk
1.4 or 1.6
On
I've setup a new asterisk box using asterisk 1.4.21.2 and used the
asterisk-gui 2.0 to configure trunks, etc. Everything is working fine except
that I am unable to register one of my trunks... stanaphone never responds
to a REGISTER request, and so they keep timing out. Other trunks are
You urge and help Bret with his terribly intelligent G729 license sharing,
clearing house plan. I think he should register a domain name and have a
PayPal Donation link. I would certainly donate for the development and even
share a few licenses.
Not sure of the legal ramifications but the idea
On Wed, Sep 24, 2008 at 05:02:53PM -0600, Joseph wrote:
I got this part: asterisk-1.4.21.2 but I need as add NVFaxDetect
NVBackgroundDetect.
There is an instruction on wiki to compile it from source but I need some
instructions on how to compile it on Gentoo.
When I try to compile current
On Wed, Sep 24, 2008 at 03:23:52PM -0700, Roderick A. Anderson wrote:
Let me know if I should post this on the asterisk-dev list instead.
I am building a Linux-Vserver (http://www.linux-vserver.org) host system
that will have several guests running Asterisk. Since the guests can't
load
Steve Totaro wrote:
You urge and help Bret with his terribly intelligent G729 license
sharing, clearing house plan. I think he should register a domain name
and have a PayPal Donation link. I would certainly donate for the
development and even share a few licenses.
Not sure of the legal
On 09/25/08 06:37, Tzafrir Cohen wrote:
On Wed, Sep 24, 2008 at 05:02:53PM -0600, Joseph wrote:
I got this part: asterisk-1.4.21.2 but I need as add NVFaxDetect
NVBackgroundDetect.
There is an instruction on wiki to compile it from source but I need some
instructions on how to compile it on
On 09/24/08 21:19, Jonn R Taylor wrote:
http://sourceforge.net/projects/agx-ast-addons/
This is the best way to install them.
Jonn
No, NVFaxDetect is not part of Extra AddOns.
The correct instruction are here:
According to Your description this is a phone problem.
Asterisk behaves as its expected.
post your dundi.conf to dig more in to this.
regards
rama
On Wed, Sep 24, 2008 at 9:52 PM, ronald ramos [EMAIL PROTECTED]wrote:
hi,
when a user register on my asterisk i can see it adding Noop for that
anyone know if there is an SNMP probe which can monitor the usage of
G729 licenses - I have had a browse through the MIB file and a google
and did not see one? - otherwise you would have no way of knowing if
it happening (other than people screaming at you!)
Robert
On Wed, Sep 24, 2008 at 8:08
On Wed, Sep 24, 2008 at 10:25:45PM -0600, Joseph wrote:
My problme is that few lines in a source code needs to be modified
before compiling it. Changing the source code is a simple thing but
now the ebuild needs to be modified as well to point to the source code;
too many problems.
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