Hello,
What version of Asterisk You are use?
And what version of A2Billing You are use?
If You use version 1.4.X of Asterisk You can put call-limit string in
sip.conf for this trunk
If You use A2B ver 1.7 and Asterk 1.4 you can announce this trunk using
sip config in A2B, and the are
Just needed to install libresample1 and libresample1-dev and all worked great.
Thanks for suggestions.
Because of the (2) reason I am planning to execute this line:
*CLI core set chanvar SIP/$channel
JACK_HOOK(manipulate,i(SIP/$channel:input),o(SIP/$channel:output)) on
Through AMI. Executing
Hello everybody.
I have been running some load tests on IAX2, as we are finding out our future
hardware investments.
Here is the setup:
We have three virtual machines.
A: running SIPP and Asterisk 1.6.2.7
B: running Asterisk 1.6.2.7
C: running Asterisk 1.6.2.7.
All of the Asterisks have been
Try it with the from= in sip.conf
You can give a from IP there.
On Sun, May 30, 2010 at 4:06 PM, CDR vene...@gmail.com wrote:
I have an Asterisk with multiple IP's, on the same subnet. When a call
comes in, I need to send it back out via SIP, but need that only one IP is
used as originating
hi, guys,
when i create a manager account used for freepbx, the follow info
produce all the time?
do you know that's the reason?
== Manager 'bitzsk' logged off from 127.0.0.1
== Manager 'bitzsk' logged on from 127.0.0.1
== Manager 'bitzsk' logged off from 127.0.0.1
== Manager
This is the problem:
Call coming into a queue in ringall strategy, if a member (SIP) of the
queue is busy when entering the queue, and this member comes free
after a little time, the member never rings..
How to solve this?
I changed all parameters of the queue with no results...
Wath i need:
On Sat, 29 May 2010 11:31:05 +0530 (IST), Nivin Kumar
nivinkuma...@yahoo.in wrote:
I would like to connect my blackberry or any other cell phone to asterisk so
that
I can send calls through the sim card. I would also like to send SMS through
this as well.
Since wifi isn't as reliable and
If you are interested in really integrating GSM phones into an
Asterisk based system without any telco involved check the OpenBTS
project. I have done a research and trial project and this combination
of open hardware (USRP), the OpenBTS open source project and Asterisk
is pretty amazing.
It is normal if you have a freepbx home screen open on one or more of your
computers.
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-05-31 4:52 AM, Zhang Shukun bit...@gmail.com wrote:
hi, guys,
when i create a manager account used for freepbx, the follow info
Asterisk 1.4.31 IAX2/RSA BUG
If A tries to register on B and the RSA key from A does not match the
key on inkeys on B, B do not send a REGREJ, instead it sends a
REGAUTH with a new CHALLENGE, then A send a new REGREQ for this
CHALLENGE with the same wrong RSA key and it loops forever on this
On Mon, 31 May 2010 13:12:25 +0200, lesouvage i...@meetmecall.nl
wrote:
If you are interested in really integrating GSM phones into an
Asterisk based system without any telco involved check the OpenBTS
project. I have done a research and trial project and this combination
of open hardware
Hi fellow asterisk users,
I am running Asterisk 1.4.29 with an Digium TE121 card (wcte12xp kernel
module) an approx. 100 snom320. The whole installation is running
without issues since 5 months.
Without having changed anything on the asterisk server for at least 2
months, i noticed clicking
a portion of your quues.conf and you sip.conf pasted can be helpful? try using
autofull=yes in your queues.conf and see if it works
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308
Date: Mon, 31 May 2010
it's autofill=yes i'm sorry for the typo
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308
Date: Mon, 31 May 2010 11:33:09 +0200
From: mass...@archivio.it
To: asterisk-users@lists.digium.com
Subject:
Dear all,
How do I set the UUI informations for outgoing calls and read the UUI
information for incoming call in asterisk?
Thanks in advance..
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
See bindaddr here:
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
That should do exactly what you want.
Regards,
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of CDR
Sent: Sunday, May 30, 2010 10:06
To:
Thanks for the advice, but I have to keep the customer on hold till the line
becomes available. Is that possible by the method you mentioned? I am using
A2B 1.7 and Asterisk 1.4.
Thanks,
On Mon, May 31, 2010 at 2:27 AM, Vardan Harutyunyan hvarda...@gmail.comwrote:
Hello,
What version of
Sorry, that made no sense, just re-read your problem.
I believe Asterisk simply takes the default IP, which would in this case be
eth0/first IP (not the virtual IPs) as outgoing IP.
Is this a problem? It is for me, I would like to define the IP used per
peer, but that's the way it is, at
On Sun, 30 May 2010 02:45:51 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
This is a bug of the netjet module. It should not try to handle those
devices. While they use the netjet chipset, they are not the ISDN BRI
devices drivven by it.
[Snip details]
If nobody beats me to it, I'll
This isn't an Asterisk issue, it's a routing issue. Take a look at iproute2
and routing policies.
Another way to view it is that Asterisk hands the communications over to Linux,
where the network route takes over. (The * bind statement just tells * what IP
to listen on)
If you have 3
Hi Michelle,
I have to say I am investigating this, and I realize this makes sense. But
I am having trouble sending packets back from where they come from. I have
setup routing policies based on networks correctly (if it comes from network
1, send it from NIC 1) but what I want is a more basic
Hello everyone!
I'm just trying to set up my new asterisk (version 1.6.2.8-rc1). I'd be
very grateful, if someone could help me here. I'd be very glad, if one of you
could test googletalk with me. Last time I tried (in 1.6.0.x times) it
wouldn't work in the end.
But here are my gtalk and
No, if You use call-limit the call will be dropped.
How you put your customer on hold?
If you use queue and the customer hear the music onhold, he will be
billed for this connection
I have try use queue and a2b, and I have do all connection using local
channel, so I have become all is works, and
On 5/31/2010 at 9:55 AM, Mike l...@virtutel.ca wrote:
See bindaddr here:
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
If I understand what you want to do, I believe you can do this with IP tables,
telling it to change the source IP of the outgoing packet to whatever you
Hi,
Asterisk 1.4.31 here. I have a queue with members defined, and those member
have member names member 1, member 2, etc. They are in a realtime DB.
When I modify those member names (column membername) the changes aren't
reflected in the queue status (show queues from cli. They aren't
Hello everyone!
So I tried to test gtalk with a friend. We could both see each other. He
uses the gtalk application for Windows.
So I tried to call him and he got a ringtone. But when he picked up, he got
a missed.
When he called me, he got a dial tone and then after one ring he got a
Hello there.
Have you tried reload in CLI?
Greeting
Aksel
Fra: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av Mike
Sendt: 31. mai 2010 21:00
Til: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Emne: [asterisk-users] Reloading
I did, show queues doesn't show the membernames, but the interface (which
is normal if the membername is NULL in the table, but it isn't).
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aksel Celasun
Sent: Monday, May 31,
Hello list,
google returns a discussion on the dev-list when I search for how to
mail a voicemail to multiple mail addresses.
Is there yet a seperator that actually works to define multiple mail
addresses ?
Kind regards,
Jonas.
--
Actually IIRC comma or semi-column worked. Try both, one of them will do.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Monday, May 31, 2010 16:08
To: Asterisk Users Mailing List - Non-Commercial
Hi Vardan,
I am using use_dnid=yes and then setting the Account Code in Asterisk
dialplan before sending the call to A2Billing _x. context which
automatically dials. So, before the call goes to A2Billing, I can check to
see if there is a channel up or not. I am not sure how the local channel you
On Mon, 2010-05-31 at 13:12 +0200, lesouvage wrote:
If you are interested in really integrating GSM phones into an
Asterisk based system without any telco involved check the OpenBTS
project. I have done a research and trial project and this combination
of open hardware (USRP), the
Hi again!
Just a short addition: I've looked a bit closer at the buddies list as well
and I see, that I ahve no resources. Though my friend could see me and I tried
sending a text message from freetalk (jabber-client) and it worked as well. I
just thought, that might be of use.
Can
Hi Andrew,
I tried that (ref.:
http://www.tipsternet.com/articles/advance%20routing.htm) but althought my
packets do seem to be marked (when I used -J LOG with iptables I do see the
appropriate packets being logged) but when Asterisk sends back the response,
the mark seems either to be gone or
A ok, I think I have understand what you want.
The first, are you want that a2b calculate the buying price?
If it for you not so important, the you can use failover trunk in a2b.
Try this.
If no, then you can you dialplan to explain what he must do on hangup cause.
I use AEL. For example,
Using in you dialplan..
GROUP
GROUP_LIST
GROUP_MATCH_COUNT
to limit outgoing calls per trunk
CLI show function GROUP_LIST
Returns a space separated list of all the groups set on a channel.
Helius
On Monday 31 May 2010 22:38:52 Vardan Harutyunyan wrote:
A ok, I think I have understand
Dear All,
Please anyone kindly help me to read and set the UUI in Asterisk.
Thanks
-- Forwarded message --
From: velusamy Krishnan velu.techni...@gmail.com
Date: Mon, May 31, 2010 at 7:02 PM
Subject: Read and set the UUI in asterisk
To: asterisk-users@lists.digium.com
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