- solved -
it turned out that libpri was not compiled correctly...
and... Asgars comment about group systax is correct.
thx
regards,
yves
Am 13.05.2013 13:21, schrieb Yves A.:
that was the syntax before 1.8 or 11.x I think...
what about pseudo?
yves
Am 13.05.2013 13:16, schrieb Asghar
Hi Mitul,
I agree that the dialplan way is easier, but it's a client requirement to
avoid using it. I was wondering if there was a way to send a call directly
to a parking slot right from the originate, because that is cheaper than
running conferences, and then joining the second call right to the
Hi Warren,
the problem is that all I have is two channels, so the specs might be join
SIP/123 and SIP/345 not join SIP/123 to 456@from-internal. They might be
Local channels, but this should be able handle the general case. The reason
why I have channels and not ext@ctxt is that I read them live
The dial n bridge might work, but there ain't indefinite wait in that
scenario.
Direct calls to parking you might try Local(70X@from-internal) but I m not
sure if this method works reliably.
The method I mentioned is used by vicidial and it works flawlessly, yes it
comes with some computing load,
My SIP provider is not happy that credentials (in the Authorization header
field) are provided in the initial REGISTER request.
The SIP provider ONLY wants the credentials AFTER rejecting the message with
a 401.
I know it's dumb, because the RFC says that the the initial REGISTER message
MAY
I never actually used parking, but should it work if I call the Park
application as the second leg of the Originate (w/o going through the
dialplan)? I dont seem to be able to make it work.
l.
2013/5/15 Mitul Limbani mi...@enterux.in
The dial n bridge might work, but there ain't indefinite
I think you could use twice the Park action to park the channels -
https://wiki.asterisk.org/wiki/display/AST/ManagerAction_Park
In the end you will have to bridge the parked channels.
HTH,
Ioan
On Wed, May 15, 2013 at 1:03 PM, Lenz Emilitri lenz.lo...@gmail.com wrote:
I never actually used
BTW - what was exactly the problem when trying to bridge the two channels
that you have sent to the wait application?
On Wed, May 15, 2013 at 4:29 PM, Ioan Indreias indre...@gmail.com wrote:
I think you could use twice the Park action to park the channels -
You could use AsyncAGI to achieve this.
Originate the first call (passing in some unique identifier as a variable),
then using AMI you will see the channel data. When you see an Event: AysncAGI
for that channel (with that id, you have control of the call). Send a Dial
Action telling it to
Hi Leonardo
At first should be useful to post your message at
asterisk-r2-requ...@lists.digium.com group.
By the other way let me advice, to make an explained detail of your problem as;
Asterisk version
Openr2 version
Configurations files
Dialplan dahdi pattern detail
Detail of the call process
While doing a security audit on a system I maintain, I stumbled upon an unvalidated use of a variable to compose an Originate request to the local Asterisk instance via AMI. Taking as an example an earlier exploit for FreePBX, I realized that this,
combined with Application: System as an injected
Brian LaVallee wrote:
My SIP provider is not happy that credentials (in the Authorization header
field) are provided in the initial REGISTER request.
The SIP provider ONLY wants the credentials AFTER rejecting the message with
a 401.
I know it's dumb, because the RFC says that the the
Hello everyone,
I've suffering cut offs after 6 or 7 seconds a call is answered, incoming
calls are working fine, but outgoing ones show the gollowing messages when
are being dropped:
[2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3641 retrans_pkt:
Retransmission timeout reached on transmission
asterisk is behind nat?
On Wed, May 15, 2013 at 8:18 PM, Daniel - Asterisk earohua...@gmail.comwrote:
Hello everyone,
I've suffering cut offs after 6 or 7 seconds a call is answered, incoming
calls are working fine, but outgoing ones show the gollowing messages when
are being dropped:
When the call is snswered, is there 2-way audio? Seems a natting issue.
On Wednesday, May 15, 2013, Daniel - Asterisk wrote:
Hello everyone,
I've suffering cut offs after 6 or 7 seconds a call is answered, incoming
calls are working fine, but outgoing ones show the gollowing messages when
Hey, all. I've got an office set up with Asterisk, and forwarding's got
a bit of a glitch:
When they forward, they listen for the remote phone to ring, then hang
up. If the remote phone doesn't connect, it goes to the original
phone's VM. Is this Polycom's fault, or Asterisk's? I've been
Users (softphones) are behind a NAT, Asterisk has its own public ip address
On Wed, May 15, 2013 at 1:30 PM, Asghar Mohammad asghar...@gmail.comwrote:
asterisk is behind nat?
On Wed, May 15, 2013 at 8:18 PM, Daniel - Asterisk
earohua...@gmail.comwrote:
Hello everyone,
I've suffering
There was 2-way audio and suddenly, the calls when down.
On Wed, May 15, 2013 at 1:30 PM, Gertjan Baarda gertjan.baa...@gmail.comwrote:
When the call is snswered, is there 2-way audio? Seems a natting issue.
On Wednesday, May 15, 2013, Daniel - Asterisk wrote:
Hello everyone,
I've
please show us peer configuration.
On Wed, May 15, 2013 at 9:46 PM, Daniel - Asterisk earohua...@gmail.comwrote:
Users (softphones) are behind a NAT, Asterisk has its own public ip address
On Wed, May 15, 2013 at 1:30 PM, Asghar Mohammad asghar...@gmail.comwrote:
asterisk is behind nat?
Hey, all. I've got an office set up with Asterisk, and forwarding's
got
a bit of a glitch:
When they forward, they listen for the remote phone to ring, then
hang
up. If the remote phone doesn't connect, it goes to the original
phone's VM. Is this Polycom's fault, or Asterisk's? I've
Current configuration follows:
[general]
context=default
allowguest=no
alwaysauthreject=yes
allowoverlap=yes
allowtransfer=yes
tcpenable=no
tlsenable=no
srvlookup=yes
vmexten=vm
rtcachefriends=yes
nat=no
directmedia=nonat
directrtpsetup=no
videosupport=yes
maxcallbitrate=384
disallow=all
sip set debug peer 90102 and check in log why call drop or upload log
somewhere. configuration seems ok.
On Wed, May 15, 2013 at 10:09 PM, Daniel - Asterisk earohua...@gmail.comwrote:
Current configuration follows:
[general]
context=default
allowguest=no
alwaysauthreject=yes
On Wed, May 15, 2013 at 12:10 PM, Ken D'Ambrosio k...@jots.org wrote:
Hey, all. I've got an office set up with Asterisk, and forwarding's got a
bit of a glitch:
When they forward, they listen for the remote phone to ring, then hang up.
If the remote phone doesn't connect, it goes to the
Does SetCallerPres(Prohib) remove the ANI data from a SIP call or does
it simply set a flag telling other devices not to display the data?
In other words, could another system override that and see the caller ID
anyway? The answer may affect how I handle 911 calls, so I'm very curious.
--
Hi Matthew,
Thanks for the response.
From: Matthew J. Roth mr...@imminc.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Wed, 15 May 2013 12:28:11 -0500 (CDT)
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi Longst,
Sorry if I am leading my question in to meaningless.
Let me explain my requirement . I don't think this something new to forum.
Supposed I have setup an Asterisk box as a IVR.I want to get the traffic via
and telecom / mobile operator.
Meaning , for instance mobile user dial a
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