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
Hi Matthew,
Thanks for the response.
> From: "Matthew J. Roth"
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Date: Wed, 15 May 2013 12:28:11 -0500 (CDT)
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Subject: Re: [asterisk-users] Initial REGISTER Re
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.
--
On Wed, May 15, 2013 at 12:10 PM, Ken D'Ambrosio 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 original phon
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 wrote:
> Current configuration follows:
>
> [general]
> context=default
> allowguest=no
> alwaysauthreject=yes
> allowoverlap=yes
> allowtr
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
allow=ula
> 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?
please show us peer configuration.
On Wed, May 15, 2013 at 9:46 PM, Daniel - Asterisk wrote:
> Users (softphones) are behind a NAT, Asterisk has its own public ip address
>
>
> On Wed, May 15, 2013 at 1:30 PM, Asghar Mohammad wrote:
>
>> asterisk is behind nat?
>>
>>
>> On Wed, May 15, 2013 at 8
There was 2-way audio and suddenly, the calls when down.
On Wed, May 15, 2013 at 1:30 PM, Gertjan Baarda wrote:
> 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
Users (softphones) are behind a NAT, Asterisk has its own public ip address
On Wed, May 15, 2013 at 1:30 PM, Asghar Mohammad wrote:
> asterisk is behind nat?
>
>
> On Wed, May 15, 2013 at 8:18 PM, Daniel - Asterisk
> wrote:
>
>> Hello everyone,
>>
>> I've suffering cut offs after 6 or 7 seconds
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
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
asterisk is behind nat?
On Wed, May 15, 2013 at 8:18 PM, 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
> are being dropped:
>
> [2013-05-15 12:5
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
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 th
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
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
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
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 wrote:
> I think you could use twice the Park action to park the channels ->
> https://wiki.asterisk.org/wiki/display/AST/ManagerA
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 wrote:
> I never actually used parking, but should
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
> The dial n bridge might work, but there ain't indefinite wait in that
> scen
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 incl
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,
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 liv
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
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