Hi!
notifyringing = no ; Control whether subscriptions already
INUSE get sent RINGING when another call is sent (default: yes)
Does this mean that when I mark this as yes, a phone that already has
taken a call will be send a second and third call ?!
No, not directly: This
Hi all,
to be able to transfer calls I have set call-limit to 2 for all of my peers.
Now how can I determine if a peer is in busy state using the first line if I
don't want to route a second call to it?
Thanks in advance,
Oliver
--
What I found is when I use sip.conf instead of realtime and
Hi all,
to be able to transfer calls I have set call-limit to 2 for all of my peers.
Now how can I determine if a peer is in busy state using the first line if I
don't want to route a second call to it?
Thanks in advance,
Oliver
--
For this one, you need to ensure that res_odbc.so loads before
res_config_odbc.so. Actually, the load order in 1.8 is such that, unless
you're using static realtime, you should not be using the 'preload' directive
at all, and everything will just naturally load in the right order.
--
If I got
Hi all,
I am testing with Asterisk 1.8.0 beta3 using realtime with a mssql server using
freetds and unixodbc, which works with 1.6.1.20.
With the same config in 1.8 I get an error when trying to start asterisk which
says:
[Aug 23 15:06:12] WARNING[7180]: loader.c:387 load_dynamic_module:
[Aug 23 15:06:12] WARNING[7180]: loader.c:387 load_dynamic_module: Error
loading module 'res_config_odbc.so':
/usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol:
ast_odbclear_cache
For this one, you need to ensure that res_odbc.so loads before
res_config_odbc.so.
For this one, you need to ensure that res_odbc.so loads before
res_config_odbc.so. Actually, the load order in 1.8 is such that, unless
you're using static realtime, you should not be using the 'preload' directive
at all, and everything will just naturally load in the right order.
No ideas? Sorry but I'm new to Linux and I am wondering why I can't stop or
tart the deamon
the way it works with 1.6.1.20. I installed Asterisk 1.8 with all defaults
et. Maybe something
is missing in any conf file?
Make sure it starts without the daemon. Try asterisk -cvvv. Does
I am using Debian Lenny, not RedHat.
-Original Message-
From: Faisal Hanif fai...@vopium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Mon, Aug 16, 2010 11:33 am
Subject: Re: [asterisk-users] Asterisk 1.8 beta3 - Unable to
Hi all,
using Asterisk 1.8 beta3 installed from scratch I am not able to
top/start/restart Asterisk deamon with
/etc/init.d/asterisk stop|start|restart
It just happens nothing, no warnings, errors etc.
Next step: start tracing.
sh -x /etc/init.d/asterisk start
--
h -x
Hi all,
using Asterisk 1.8 beta3 installed from scratch I am not able to
stop/start/restart Asterisk deamon with
/etc/init.d/asterisk stop|start|restart
It just happens nothing, no warnings, errors etc.
I am running Debian Lenny.
Any ideas what is wrong?
Thanks,
Oliver
--
Yes I do.
And yes, I checked, it is executable.
--
Sorry if this is a stupid question but you are doing
sudo /etc/init.d/asterisk stop|start|restart
Aren't you?
have you made sure /etc/init.d/asterisk is executable?
Ish
On 11/08/10 11:14, unsero...@aol.com wrote:
Hi all,
using Asterisk
Hi all,
using Asterisk 1.8 beta3 installed from scratch I am not able to
top/start/restart Asterisk deamon with
/etc/init.d/asterisk stop|start|restart
It just happens nothing, no warnings, errors etc.
Next step: start tracing.
sh -x /etc/init.d/asterisk start
--
h -x
Hi all,
i finally got it working to use Asterisk 1.6.1.20 as my PSTN gateway for
OCS2007 R2 following the HowTo
http://blogs.technet.com/b/gclark/archive/2008/10/09/asterisk-1-6-with-office-communications-server-2007.aspx.
I can call the OCS from Asterisk and vice versa.
The only thing that
Hi all,
i finally got it working to use Asterisk 1.6.1.20 as my PSTN gateway for
OCS2007 R2 following the HowTo
http://blogs.technet.com/b/gclark/archive/2008/10/09/asterisk-1-6-with-office-communications-server-2007.aspx.
I can call the OCS from Asterisk and vice versa.
The only thing
I did it straight Asterisk to OCS using the OCS Mediation Server. We do have
Dialogic Diva Server Cards which are able to be used as Media Gateway too using
an additional software called SipControl (not 100% sure about the name) but as
this software needs to be licensed separately I prefer a
You cannot use realtime static and the other realtime tables at the
same time. You will need to use realtime and then use something like
the EXEC command in sip.conf to execute a script that then pulls the
register statement from your database. Or use the realtime static table
for
Please note that I don't claim myself a guru, just happened to be working with
Asterisk for some good number of years, so probably know some stuff better than
others.
As for the number of lines, 1800 lines will come down to 1000 lines using AEL
but not the opposite.
When I'll be back home,
I've got 2 asterisk servers on the same box: ubuntu 10.04 lucid. I have not
een able to send useful callerid info between them (callerid becomes
serverB).
serverA register statement: (serverB has the exact opposite statement)
egister = serverA:serverapassw...@ip_of_serverb_nic/serverB
Hi all,
I have
exten = _X.,1,Macro(dundi-priv,${EXTEN})
exten = _X.,2,DIAL(CAPI/contr1/${EXTEN})
Now my problem is, that after hanging up a call, the call is instantly
re-established using the h-extension which is almost a loop.
I am sure this is a stupid question, but what am I
AEL is very simple and the instructions on voip-info.org are enough to learn
it. In fact I can't understand how can one write complex dial plans not using
AEL, you simply can't do it using standard format used in extensions.conf.
As for the tutorials, there is no specific website for them as
Hi all,
I am just trying to implement DUNDi-Routing like described here
http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP+with+no+passwords
and have a most probably stupid question:
My config is exactly like described except that instead of
exten =
Hi all,
I have
exten =_X.,1,Macro(dundi-priv,${EXTEN})
exten = _X.,2,DIAL(CAPI/contr1/${EXTEN})
Now my problem is, thatafter hanging up a call, the call is instantly
re-established using theh-extension which is almost a loop.
I am sure this is astupid question, but what am I doing
Hi all,
I have two questions regarding DUNDi and Asterisk Realtime. I have
successfully set up DUNDi on my two Asterisk boxes, which means
dundi show peers on each box shows the other box as known and dialplan
show
dundiextens shows the extensions on each box configured in sip.conf.
Hi all,
I have two questions regarding DUNDi and Asterisk Realtime. I have successfully
set up DUNDi on my two Asterisk boxes, which means
dundi show peers on each box shows the other box as known and dialplan show
dundiextens shows the extensions on each box configured in sip.conf.
1. But
Hi all,
I am wondering if the Clustering concept described in Leif Madsens presentation
http://comunidad.asterisk-es.org/documentos/astricon2008/why-cluster-an-introduction-to-asterisk-clustering-and-database-integration-astricon-2008.pdf
is still up to date or if there are newer or improved
Hi all,
I am wondering if the Clustering concept described in Leif Madsens
presentation
http://comunidad.asterisk-es.org/documentos/astricon2008/why-cluster-an-introduction-to-asterisk-clustering-and-database-integration-astricon-2008.pdf
is still up to date or if there are newer or
Do you know if it is possible to interconnect 1.6 with Microsoft Office
Communications Server 2007 and use the Office
Communicator as a softclient for telephone calls and the Communicator for
Instant Messaging? I believe you can set up a mediation
server within MOC but i don't know if
No, not until Microsoft builds a compatible soft phone. Microsoft
built software that only speaks SIP over TCP. Most SIP stacks work
over RTP.
I suspect you meant UDP, not RTP. They use TCP or UDP for SIP signaling and
RTP for the actual voice traffic.
--
This is what i thought.
I am also getting segmentation fault when doing a reload from CLI.
Asterisk crashes and i see
segfault at 46 ip b752827d sp b2bc38f8 error 4 in libc-2.7.so[b74ca000+155000]
in dmesg.
I use Debian Lenny 32bit.
--
_
--
I am also getting segmentation fault when doing a reload from CLI.
I believe this is your issue : https://issues.asterisk.org/view.php?id=17704
If not, create a new issue on the tracker with an unoptimized backtrace.
--
Yes, the patch fixed it. Thanks.
--
Hi,
i just tried to use the CONNECTEDLINE() feature but it does not work, at least
with my softphones (zoiper, 3CX, Xlite)
in sip.conf under general I have:
trustrpid = yes
sendrpid = rpid,pai
rpid_update = yes
in extensions.conf I have:
exten = 2000,1,Set(CONNECTEDLINE(number,i)=98)
exten =
Hi,
i just tried to use the CONNECTEDLINE() feature but it does not work, at
least
with my softphones (zoiper, 3CX, Xlite)
in sip.conf under general I have:
trustrpid = yes
sendrpid = rpid,pai
rpid_update = yes
in extensions.conf I have:
exten =
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon
Henderson
Sent: Thursday, July 22, 2010 3:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Soft
Hi,
I am trying to write the regserver value into my database using ARA but the
field keeps empty.
Afaik all that needs to be done to make it work is having a db field called
regserver, the var systemname set in asterisk.conf and
rtsavesysname=yes in sip.conf.
But the regserver is not
Hi,
I am trying to write the regserver value into my database using ARA but the
field keeps empty.
Afaik all that needs to be done to make it work is having a db field called
regserver, the var systemname set in asterisk.conf and
rtsavesysname=yes in sip.conf.
But the regserver is not
-Original Message-
From: Jonas Kellens jonas.kell...@telenet.be
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Mon, Jul 12, 2010 3:09 pm
Subject: [asterisk-users] Remote-Party-ID party=called
Hello list,
using Asterisk 1.4.30.
I
It doesn't work for me too..
exten = 1400,1,SIPAddHeader(Remote-Party-ID: Test
sip:2...@192.168.1.150:5060\;party=called)
exten = 1400,n,Dial(SIP/${EXTEN},15)
leads to
-- Executing [1...@default:1] SIPAddHeader(SIP/1401-0159,
Remote-Party-ID: Test
No, the receiving side shows name and number as it should.
But as calling person I only see the number of the called person instead of
name and number.
So we seem to struggle with the same issue.
-Original Message-
From: Jonas Kellens jonas.kell...@telenet.be
To: Asterisk Users
On Mon, 2010-07-12 at 10:48 -0400, unsero...@aol.com wrote:
No, the receiving side shows name and number as it should.
But as calling person I only see the number of the called person
instead of name and number.
So we seem to struggle with the same issue.
This is something that is not
Hi all,
i have a question regarding the dialplan when using a DNS round robin for
simple load balancing.
When i have 3 identically configured Asterisk servers and one DNS round robin
populated to the clients
Server1
Server2 - round robin voip.example.com
Server3
where all 3
But how can i determine on which physical server user B is registered?
Or is there an other, better way to achieve this? Maybe in replicating the
registrations between all 3 servers?
DUNDi is an options, same with DNS SRV records.
--
Could you please give me some more info?
Or is
On Sun, Jul 11, 2010 at 8:09 PM, unsero...@aol.com wrote:
But how can i determine on which physical server user B is registered?
Or is there an other, better way to achieve this? Maybe in replicating the
registrations between all 3 servers?
DUNDi is an options, same
Hi all,
i have a beginners question. How are SIP calls and IAX2 calls processed by
Asterisk over the network?
What i mean is, is there a permanent connection required between the Asterisk
Server and the clients or is the Asterisk Server only involved for lets call it
the routing?
From my
Sounds great, thanks for your answer.
Do i need to set this on the trunk, the friend or on both?
-Original Message-
From: bruce bruce bruceb...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Fri, Jul 9, 2010 8:13 pm
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In you are just using Asterisk in the dialplan you can set
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In you are just using Asterisk in the dialplan you can
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In you are just using Asterisk in the
The client needs to support the Remote-Party-ID SIP header. If you
want to verify the header is being added run tcpdump and analyze it
with Wireshark. I know that Polycom phones have support for this. I
just put a modified version of the Asterisk 1.6.1 patch into
production for 25 Polycom
I just did not want to spam the list with useless content but just reply to you
as you attacked me.
This was the reason i only replied to you and not to the whole list.
But as i realised now it seems to be usual to spam the whole mailing list with
useless information like your mail (or this
Hi,
one question again from an asterisk newbie.
Where can i get the header files Asterisk 1.6.1.20 for Debian Lenny?
I need them to install chan_capi for my Diva E1 Server Card.
Thanks in advance.
--
_
-- Bandwidth and
-Original Message-
From: Paul Belanger paul.belan...@polybeacon.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Fri, Jul 2, 2010 7:35 pm
Subject: Re: [asterisk-users] Asterisk header files for Debian Lenny 1.6.1.20
On
This are the header files for 1.4, not for 1.6.
Then how did you install asterisk 1.6?
--
from here http://downloads.digium.com/pub/asterisk/asterisk-1.6.1-current.tar.gz
but i can't find header-files or dev-files there.
--
but i can't find header-files or dev-files there.
include folder
--
It's that easy? Ok, so stupid question from me.
But thanks for your help.
=
--
_
-- Bandwidth and Colocation Provided by
Hi,
i have the same problem. Trying to use the dialplan function CONNECTEDLINE()
this way
Set(CONNECTEDLINE(name)=${SIPPEER(${EXTEN},callerid_name)})
Set(CONNECTEDLINE(num)=${EXTEN})
ends with
[Jul 1 16:30:50] ERROR[3954]: pbx.c:2902 ast_func_write: Function
Sorry, what does this mean? Only in trunk?
-Original Message-
From: Steve Howes steve-li...@geekinter.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 5:02 pm
Subject: Re: [asterisk-users] Remote Party ID issue
Sounds great.
Could you please give me a hint how to install the patch?
Sorry for my stupid question but I'm a newbie to Asterisk.
Thanks.
-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
Thanks a lot.
Applying the patch gives me a
Hunk #5 failed at 9881
-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 5:37 pm
Subject: Re: [asterisk-users]
-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 6:19 pm
Subject: Re: [asterisk-users] Update the LCD with the callee's name after
dialing
On Thu, Jul 1,
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