Hello,
For realtime configuration, in /etc/asterisk/extconfig.conf file, what should
be the family name to configure general sip.conf parameters.
=> ,,
thanks,
Kamlesh
> From: i...@pack-net.co.uk
> To: asterisk-users@lists.digium.com
> Date: Wed, 2 May 2012 13:59:58 +0100
> Subject: Re
2012/5/3 JIMMY GATHAGE :
> I am using a SIP trunk to make outgoing calls. Outgoing calls are
> going through okay. I am using the AMI to Originate a call. The
> channel is not returning any event when the phone on the PSTN is
> ringing. How can i detect the phone ringing on the SIP channel?
It is
On 05/02/2012 04:41 PM, JIMMY GATHAGE wrote:
Hey guys,
I am using a SIP trunk to make outgoing calls. Outgoing calls are
going through okay. I am using the AMI to Originate a call. The
channel is not returning any event when the phone on the PSTN is
ringing. How can i detect the phone ringing on
Hey guys,
I am using a SIP trunk to make outgoing calls. Outgoing calls are
going through okay. I am using the AMI to Originate a call. The
channel is not returning any event when the phone on the PSTN is
ringing. How can i detect the phone ringing on the SIP channel?
Am desperate.
Thanks.
--
_
The Asterisk Development Team has announced the release of Asterisk 10.4.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 10.4.0 resolves several issues reported by the
community and would have not been possible wit
The Asterisk Development Team has announced the release of Asterisk 1.8.12.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 1.8.12.0 resolves several issues reported by the
community and would have not been possible
Or even Hangup(-${Z})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry Miller
Sent: Wednesday, May 02, 2012 2:11 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] parsing issue
On Wed,
On Wed, May 02, 2012 at 01:48:04PM -0400, CDR wrote:
> I get an error when I execute this code
> exten => rejected,n,Hangup($[-1*${Z}])
>
> May 2 13:42:09] WARNING[23128]: ast_expr2.fl:468 ast_yyerror:
> ast_yyerror(): syntax error: syntax error, unexpected $end, expecting
> '-' or '!' or '(' or
Have you tried the MATH() function?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of CDR
Sent: Wednesday, May 02, 2012 1:48 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] parsing issue
I get a
3 possibilities - 1 use an AGI; 2 use a system command 3 (hopefully the
simplest) just use a Set command. Which one you actually end up using may
depend on your Asterisk flavor.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.co
Eric,
You were right.
Thanks :)
Michael
On Wed, May 2, 2012 at 7:25 PM, Eric Wieling wrote:
> If you have quotes on one side of the = sign, then you need quotes on the
> other side. In your dialplan line you are comparing + with "+". A plus
> sign is not equal to quote plus sign quote
>
> e
If you have quotes on one side of the = sign, then you need quotes on the other
side. In your dialplan line you are comparing + with "+". A plus sign is not
equal to quote plus sign quote
exten => _X., n, Set(CALLERID(num)=${IF($["${CALLERID(num):0:1}" =
"+"]?${CALLERID(num)}:0${CALLERID(nu
I think you need to "escape" the + "\+"
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael
Sent: Wednesday, May 02, 2012 11:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] detecting intl
Hello asterisk users,
I need to convert the CLI received according to national/international
format:
55-555- to 055-555- (add 0 in the beginning)
+55-55-555- to +55-55-555- (remains unchanged)
I put the following line in my dial plan:
exten => _X., n, Set(CALLERID(num)=${IF($[ ${
Hello all,
I'm trying to solve a problem on a T1 span setup wherein calls are
apparently not hanging up properly.
The system in question is using a Xorcom Astribank with 1 full and 1
partial T1 span, and running Asterisk 1.4.36.
The symptom is that when a call hangs up on a DAHDI channel (accord
On Wed, 2012-05-02 at 12:04 +, Kamlesh Kumar wrote:
> Hi,
>
> I need to configure global parameters in sip.conf like rtptimeout,
> rtpholdtimeout, rtpkeepalive, domain, session-timers etc... in real
> time architecture. Please suggest the way to do it.
>
> thanks,
> Kamlesh
>
Hi
You can
On Wed, May 2, 2012 at 3:44 PM, SamyGo wrote:
>> I can't figure out if it's a known issue, or a new bug.
>
>
> Or a new feature !!
>
> Can you share the dialplan code where you are executing the mixmon
> application !
I use it using the manager:
DEBUG ami 2012-05-02 15:43:53 Sending AMI action:
>
> I can't figure out if it's a known issue, or a new bug.
Or a new feature !!
Can you share the dialplan code where you are executing the mixmon
application !
Regards,
Sammy.
On Wed, May 2, 2012 at 5:09 PM, ik wrote:
> Hello,
>
> I have weird issue with Asterisk 8 lately.
>
> When I call
>> I'm using Asterisk 8.11.1
As far as I'm aware, there is no Asterisk 8.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
--
___
2012/5/2 Kamlesh Kumar
> Hi,
>
> I need to configure global parameters in sip.conf like rtptimeout,
> rtpholdtimeout, rtpkeepalive, domain, session-timers etc... in real time
> architecture. Please suggest the way to do it.
>
> thanks,
> Kamlesh
>
>
For what I have discovered, it is not possible
Hello,
I have weird issue with Asterisk 8 lately.
When I call MixMonitor without mixing the channels, it changes the
sides of "in" and "out".
Sometimes the first leg of the call is "in" and sometimes it's "out".
I can't figure out if it's a known issue, or a new bug.
I'm using Asterisk 8.11.1
Hi,
I need to configure global parameters in sip.conf like rtptimeout,
rtpholdtimeout, rtpkeepalive, domain, session-timers etc... in real time
architecture. Please suggest the way to do it.
thanks,
Kamlesh --
I'm currently doing some testing with Asterisk ( 1.8.11.0) on RHEL6
using realtime for sippeers, sipusers and musiconhold
I have Avaya definity <-> PRI E1 <-> Asterisk 1 <-> IAX2 <-> Asterisk
2
I have peers (sip) snom 821s on both Asterisk 1 and 2 all calls working
between all systems.
Caller
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