I tried to do what I with regular SIP to Transfer a call via 302
Redirect. In asterisk 12 we need to add the Tech, or not, but in any
case, there is no transfer done. The call is closed.
Here is a trace. How do I do this?
[Jul 9 21:39:29] DEBUG[47716][C-0002]: pbx.c:4869
pbx_extension_helpe
The description of busy() in the asterisk documentation wiki states:
"This application will indicate the busy condition to the calling channel."
Wouldn't 'indicate the busy condition' on a PRI channel imply setting cause 17?
-Justin
From: asterisk-users-boun...@l
Generally if you want to send a cause 17 to the caller you would use Hangup(17)
and let the caller's switch generate the busy tone.
If the dialplan has already answered the call, then you might want to use Busy
or Playtones.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
Okay, I think I need a sanity check here - If I call a person that's on the
phone, I should get a busy signal.
Now more specifically, a call comes into the pbx via PRI. The destination
dialplan runs busy(20). Now, the PRI causecode should get set to 17 (user
busy) so that the originating end
Justin Killen wrote:
Is there a channel variable / status indicator / function that indicates
if the current channel has been answer()’ed?
${CHANNEL(state)} will return the state the channel is currently in. If
the channel is answered the state will be "Up".
Cheers,
--
Joshua Colp
Digium, I
Is there a channel variable / status indicator / function that indicates if the
current channel has been answer()'ed?
-Justin
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us f
I figured it out - the busy tone was being generated by the local end. It
seems that upon receiving a hangup, freepbx tries the next trunk. In this case
there isn't any other trunk, so I get "all circuits are busy now".
For anyone interested, the site B playback()/busy() condition has been
su
I found a very strange proble whit two asterisk servers in the same network.
Scenario
Asterisk A with extensions 5XX
Asterisk B with extensions 2XX
There is NO link between the two asterisks.
Call from 501 to 503, 503 ringing
Call from 201 to 203, 203 ringing
The 202 extension comand a pickup
I tried changing the dialplan to use Hangup(17) instead of Playback/Busy. Now
instead of ringing for 20 seconds and then getting the "all circuits are busy
now", I get "all circuits are busy now" immediately.
New snippet from site A:
...
[2014-07-09 10:53:43] VERBOSE[23020][C-db0d] app_dial
Quoting Ishfaq Malik (i...@pack-net.co.uk):
> On 9 July 2014 16:19, Olivier wrote:
>
> > Hi,
> >
> > I'm seeing a trend in which SIP devices such as Yealink SIP phones
> > (with v72 firmware), are dropping support of SNMP in favor of "HTTP
> > eventing"
> > How do deal with those devices ?
> If y
If you use Playtones you should put an Answer and a Wait(1) before the Playtones
I recommend using the Hangup app instead. Busy would be Hangup(17).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Wednesday, July
I have two servers, each connected to the PTSN via PRI. When I call from site
A (951-999-) to site B (555-1212) and the phone at site B is on the phone,
I hear the normal ring tone for about 20 seconds, then the message "all
circuits are busy now. please try your call again latter" followe
On 9 July 2014 16:19, Olivier wrote:
> Hi,
>
> I'm seeing a trend in which SIP devices such as Yealink SIP phones (with
> v72 firmware), are dropping support of SNMP in favor of "HTTP eventing" if
> may call this as such :
> when configuring the SIP device, you can define a couple of HTTP URL whi
asterisk-users-boun...@lists.digium.com wrote on 07/09/2014 10:19:11 AM:
> From: Olivier
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> ,
> Date: 07/09/2014 10:19 AM
> Subject: [asterisk-users] How to monitor non-SNMP SIP devices ?
> Sent by: asterisk-users-boun...@lists.digium
Hi,
I'm seeing a trend in which SIP devices such as Yealink SIP phones (with
v72 firmware), are dropping support of SNMP in favor of "HTTP eventing" if
may call this as such :
when configuring the SIP device, you can define a couple of HTTP URL which
triggered when some event occur (end of boot, o
On Wed, Jul 9, 2014 at 4:56 AM, Sameer Rathod wrote:
> Hi,
>
> with canreinvite=no and directmedia=no I and getting the message in the logs
> for all calls
>
> "switching from simple_bridge technology to native_rtp"
>
>
> -- Executing [102@mkg:1] Dial("SIP/101-0017", "SIP/102") in new stack
>
On Wed, Jul 9, 2014 at 2:47 AM, Sameer Rathod wrote:
> Hi,
>
> Please clear me on this topic I am confused
>
> My log show "switching to native rtp".
> Did this line means that the audio is not coming to the asterisk server any
> more and asterisk only send the re- invite packet to both the client
On Wed, Jul 02, 2014 at 10:05:44PM +0200, Thomas wrote:
> Hello,
>
> in Squeeze Asterisk 1.8.23.1 is installed,
Self-installed
> in Wheezy older version
> 1.8.13.1~dfsg1-3+deb7u3.
>From a package.
>
> With version 1.8.13.1 I have some problems so I would like to install version
> 1.8.23.1
Hi Ishfaq,
I am getting the the flow as attached Could you please read and check if
the rtp is passing directly as I am new and dont know much about this all
On Wed, Jul 9, 2014 at 3:24 PM, Ishfaq Malik wrote:
> use tcpdump on the server to see if the RTP traffic is passing through it.
>
>
>
Hi,
with canreinvite=no and directmedia=no I and getting the message in the
logs for all calls
"switching from simple_bridge technology to native_rtp"
-- Executing [102@mkg:1] Dial("SIP/101-0017", "SIP/102") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/102
-- SIP/102-000
use tcpdump on the server to see if the RTP traffic is passing through it.
On 9 July 2014 10:48, Sameer Rathod wrote:
> Hi Mitul,
>
> I checked that the re-invite packet are sent what I want to check is
> whether the audio packets is going through the server or not ?
>
>
> On Wed, Jul 9, 2014 a
Hi Mitul,
I checked that the re-invite packet are sent what I want to check is
whether the audio packets is going through the server or not ?
On Wed, Jul 9, 2014 at 1:42 PM, Mitul Limbani wrote:
> Put sip debug on to know if reinvite packets are sent.
> On 09-Jul-2014 1:17 PM, "Sameer Rathod"
Le 08/07/2014 16:07, Eric Wieling a écrit :
If you are executing "database put Agora modele/IVR/AstreinteNagios/1
${ASTR_State}" while in the Asterisk CLI, that won't work. You cannot access
DIALPLAN variables from the CLI.
I didn't know that, thanks. Will try another way.
Regards
-O
Put sip debug on to know if reinvite packets are sent.
On 09-Jul-2014 1:17 PM, "Sameer Rathod" wrote:
> Hi,
>
> Please clear me on this topic I am confused
>
> My log show "switching to native rtp".
> Did this line means that the audio is not coming to the asterisk server
> any more and asterisk
Hi,
Please clear me on this topic I am confused
My log show "switching to native rtp".
Did this line means that the audio is not coming to the asterisk server any
more and asterisk only send the re- invite packet to both the clients ?
Am I right or wrong ?
On Tue, Jul 8, 2014 at 11:50 PM, Mitu
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