On Thu, Jan 28, 2016 at 5:34 PM, Sonny Rajagopalan <
sonny.rajagopa...@gmail.com> wrote:
> Hi,
>
> I am using Asterisk 13.6.0 and was wondering if I can programmatically add
> users (to pjsip.conf) and dialplan (to extensions.conf) to the Asterisk
> server using API of some sort.
>
>
You can use
On Thu, Jan 28, 2016 at 6:58 PM, James Cloos wrote:
> > "AS" == A J Stiles writes:
>
> AS> If you are paying for a business-grade Internet connection, you
> AS> should get a static IP address -- or a block of them -- as
> AS> standard. Maybe you need to change your ISP?
>
> In some places (
> "AS" == A J Stiles writes:
AS> If you are paying for a business-grade Internet connection, you
AS> should get a static IP address -- or a block of them -- as
AS> standard. Maybe you need to change your ISP?
In some places (including here) static ip is not affordable.
-JimC
--
James Cloo
Hi,
I am using Asterisk 13.6.0 and was wondering if I can programmatically add
users (to pjsip.conf) and dialplan (to extensions.conf) to the Asterisk
server using API of some sort.
Please do let me know.
Thanks,
Sonny.
--
_
--
On 01/28/2016 03:39 PM, sean darcy wrote:
i've got calls coming into an 11.21.0 box. The internal phones are
analogue off a TDM400 board, and SIP extensions.
Using an analogue internal phone, the remote party always hears an echo
on it's side. We do not hear an echo. Doesn't matter who is the ca
i've got calls coming into an 11.21.0 box. The internal phones are
analogue off a TDM400 board, and SIP extensions.
Using an analogue internal phone, the remote party always hears an echo
on it's side. We do not hear an echo. Doesn't matter who is the calling
party.
But if we use a SIP exten
Hello,
I'm giving a try to Resource List Subscriptions feature also called BLF
List in several phone vendors documentation (see [1]).
I could successfully configured this feature woth Yealink phones but I've
got some issues with Aastra phones (6757i with 3.3.1 firmware).
Before diving deeper int
Hello,
Thanks for your reply.
Is this mentioned in any RFC ? I checked RFC3325 for PAI and RFC3261,
but nothing mentioned there.
Best regards
On Thu, Jan 28, 2016 at 2:50 PM, Laurent Schweizer <
laurent.schwei...@peoplefone.com> wrote:
> Hello,
>
>
>
> Usually in the P-Asserted you have the n
Hello,
Usually in the P-Asserted you have the network number and in the From the
preferred number.
In this case the Preferred (from) number is displayed.
BR
Laurent
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Aziz TestAccount
E
Hi All,
When receiving an invite containing two different caller ID, one in FROM
header and the other in "P-Asserted Identity" Header, Which one will be
used by the callee ? I couldn't find any RFC specifying this detail.
Thank you.
--
__
when you say load - how many concurrent calls? Is there transcoding
happening? sip / PRIs ? what load?
On Thu, Jan 28, 2016 at 9:57 AM, Marek Červenka wrote:
> Dne 27.1.2016 v 17:50 A J Stiles napsal(a):
>
>> On Wednesday 27 Jan 2016, Marek Červenka wrote:
>>
>>> Dne 27.1.2016 v 13:14 A J Stiles
Dne 27.1.2016 v 17:50 A J Stiles napsal(a):
On Wednesday 27 Jan 2016, Marek Červenka wrote:
Dne 27.1.2016 v 13:14 A J Stiles napsal(a):
On Wednesday 27 Jan 2016, Marek Červenka wrote:
hi,
i have strange problem with asterisk 13 mixmonitor, recording to wav
(centos6)
when the system is under l
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