hi,
in my small setup (just for home usage) i have 5 phones configured. but
only 2 of them are permanent connected to asterisk.
nevertheless i want to address beside those two phones other peers if
available. nowadays i address them always, resulting in error messages:
Unable to create channel of
Hi;
How I can make my configuration to allow the sip phones only from specific IP
addresses range (for example from 192.168.10.1 - 192.168.10.50) to be allowed
to connect for asterisk?
In other words, in addition to be authenticated based on the username and
password, it is required that the
Hi
You can achieve this with either permit/deny or contactpermit/contactdeny
Single IP should be defined like :
deny=0.0.0.0/0.0.0.0
permit=192.168.2.1/255.255.255.255
And networks in similar way with appropriate subnet mask
deny=0.0.0.0/0.0.0.0
permit=192.168.2.0/255.255.255.0
You can also
bilal ghayyad wrote:
Hi;
Hola,
How I can make my configuration to allow the sip phones only from specific IP
addresses range (for example from 192.168.10.1 - 192.168.10.50) to be allowed
to connect for asterisk?
In other words, in addition to be authenticated based on the username and
Andre Gronwald wrote:
hi,
Hola,
in my small setup (just for home usage) i have 5 phones configured. but
only 2 of them are permanent connected to asterisk.
nevertheless i want to address beside those two phones other peers if
available. nowadays i address them always, resulting in error
Jerry Geis wrote:
I think I have a race condition.
I am running something like this in my dialplan
call agi to bring my list of devices into my MeetMe
Playback beep
start MeetMe()
So in fact the meetme is not started before I bring the list
of devices into the meetme.
How can I do this
Carlos Chavez wrote:
The card itself does not have hardware echo cancellation so we use MG2.
I am not fixated on the card because this should not affect a SIP to SIP
internal call unless the card is really defective and provides bad
timing to Asterisk.
Actually when bridging channels Asterisk
Face wrote:
Hello,
Hola,
After Upgrade to Asterisk 11.1.0-rc1 I keep getting
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [603@DLPN_AlDimnaDialPlan:601]
Dial(SIP/601-0002, SIP/603)
Hello
In SIP.find you can to use
Deny=0.0.0.0/0.0.0.0
Permit=192.168.1.25/255.255.255
Regards
On Nov 19, 2012 7:12 AM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi;
How I can make my configuration to allow the sip phones only from specific
IP addresses range (for example from 192.168.10.1 -
Can you clarify what you mean by MeetMe to be active? What MeetMe
options are you using and what is your configuration like? With the
proper combination of options it shouldn't matter who gets into the
conference bridge first. This is what Page essentially does, with the
difference being that
Hello,
If would be possible to use a function concept in side of Asterisk DialPlan
For example:
I have following logic in my dial plan remove country code a add an 0 before
the rest of the numbers
exten = _X.,1, NoOp( call ID ${CALLERID(num)} exten: ${EXTEN}))
; remove my country code
You could do it as a function if you are C literate. The simpler way would
be to do it as an AGI where you passed the ${EXTEN} value to the AGI and had
the AGI pass the modified number back as a dialplan variable.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I had a similar problem (I work on 3 lans; when my firewall is down, the two
non-native lans are unaccessible) I wrote an AGI to execute sip show
peers and process only the ones that return OK and pass my peer numbers to
the AGI like this -
[dialall]
Exten = s,1,AGI(sipcheck.agi,100,200,300)
We are getting this message on an Asterisk 1.4.44 box.
[2012-11-19 08:49:27] WARNING[11785] app_voicemail.c: List of extensions is too
long (1323). Truncating.
I know Asterisk removed many of limitations in string lengths in in 1.6+. Does
anyone know if this also applies to app_voicemail?
I can tell you this warning does not exist in 10.9.0 or 11.0.0.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, November 19, 2012 12:01 PM
To: asterisk-users@lists.digium.com
Subject:
On Sun, Nov 18, 2012 at 11:32 AM, Michael voip.quest...@gmail.com wrote:
Gentlemen,
So, from your answers I understand that I have 2 options:
1. AMI Redirect command
2. Asterisk command ChannelRedirect
I'm inclined to prefer the 2nd option, as we've never used AMI, but I
don't know if it
Thanks. We will never upgrade to Asterisk 10 and we won't be upgrading to
Asterisk 11 for 12 - 18 months. In my experience with Asterisk 1.4, 1.6, and
1.8 is that it takes that long for Asterisk to be stable enough for our use.
I'm STILL in therapy because of the if you receive a VM
Good Day dear members,
We are trying to test asterisk in our office to extend the reach of our
present proprietary pabx system if successful.
I am using an oracle virualbox 4.2.4 as the virtual server platform with
ubuntu 12.04.1 server as the operating system.
I get errors while trying to
The warning is also non-existent in 1.8.17. I don't/won't mess with 1.6.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, November 19, 2012 12:32 PM
To: Asterisk Users Mailing List -
On Mon, Nov 19, 2012 at 08:47:23PM +0100, Adolphus Enaboifo wrote:
.. I get errors while trying to compile Libpri 1.4.13. (check
attachment} Can you guys please help me prescribe a fix.
[snip]
gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD
-MT pridump.o -MF
Am 19.11.2012 19:00, schrieb asterisk-users-requ...@lists.digium.com:
Subject: Re: [asterisk-users] addressing peers dynamically To:
Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com Message-ID:
50aa2586.80...@digium.com Content-Type: text/plain;
I need some advice on how to implement something in my dialplan.
Here's the scenario. A call comes in on my [incoming] context and I answer
it. The call turns out to be for my wife and she needs to answer it on a
different
handset somewhere else in the house.
I've tried call parking but the
Have you looked into SLA? I have had good results with it. Will let asterisk
act like a key system.
Sent from Samsung tablet
Chris Gentle gent...@gmail.com wrote:
I need some advice on how to implement something in my dialplan.
Here's the scenario. A call comes in on my [incoming] context
I need some advice on how to implement something in my dialplan.
Here's the scenario. A call comes in on my [incoming] context and I
answer it. The call turns out to be for my wife and she needs to
answer it on a different
handset somewhere else in the house.
I've tried call parking but
You can park the call, set the timeout low, and have it return to a ring
group.
On Nov 19, 2012 6:15 PM, Chris Gentle gent...@gmail.com wrote:
I need some advice on how to implement something in my dialplan.
Here's the scenario. A call comes in on my [incoming] context and I
answer it. The
On Mon, Nov 19, 2012 at 3:51 PM, Joshua Colp jc...@digium.com wrote:
Face wrote:
Hello,
Hola,
After Upgrade to Asterisk 11.1.0-rc1 I keep getting
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
On Mon, Nov 19, 2012 at 6:23 PM, Jared Baxley jared.bax...@gmail.comwrote:
You can park the call, set the timeout low, and have it return to a ring
group.
Thanks to everyone for the suggestions. I decided to try this approach
first and I think I have it working. However, I found a slight
Hello All,
Anyone have idea regarding below error.
After applying all patch, still faced the same issue.
--
Regards,
Chandrakant Solanki
On Fri, Nov 9, 2012 at 11:39 AM, Chandrakant Solanki
solanki.chandrak...@gmail.com wrote:
Hello All,
I am using asterisk 1.8.13.0 and which is
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