[asterisk-users] Resource List Subscriptions/BLF List and Aastra phones

2016-01-28 Thread Olivier 
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 into this, can you share your own experience here ?

More specifically, here are my current results:

- dedicated keys (softkey type) are automatically populated
- when pressing one dedicated key when matching extension is idle, nothing
is dialed (speed dialing not working)
- when pressing one dedicated key when matching extension is ringing,
nothing is dialed (directed pickup not working)

Comments ? Suggestions ?

Regards

[1] https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=30278158
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Re: [asterisk-users] 11.21.0 : echo woes : can't install canceller

2016-01-28 Thread sean darcy 

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 calling
party.

But if we use a SIP extension, no echo.

I've built /lib/modules/4.3.3-303.fc23.x86_64/dahdi/dahdi_echocan_oslec.ko

And requested oslec echo cancel:

grep oslec /etc/dahdi/system.conf
echocanceller=oslec,1,2,4

grep echo /etc/asterisk/chan_dahdi.conf
echocancel=yes
echocancelwhenbridged=no
echotraining=yes

but it's never loaded:

# lsmod | grep echo
[root@asterisk ~]#

dahdi_cfg -vvv
DAHDI Tools Version - 2.10.0

DAHDI Version: 2.11.0
Echo Canceller(s):
Configuration
==


Channel map:


0 channels to configure.

I can manually insert the oslec module using modprobe.

Thats seems to work.

CLI> dahdi show version
DAHDI Version: 2.11.0 Echo Canceller: OSLEC

But it's not persistent across reboots.

sean




And even if I do manually load the oslec kernel module, I don't think 
it's actually being used


cat /proc/dahdi/1
Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER)

1 WCTDM/4/0 FXOKS (In use)
2 WCTDM/4/1 FXOKS (In use)
3 WCTDM/4/2 Reserved
4 WCTDM/4/3 FXSKS (In use) RED

AFAIK, the echo canceller should show up here.

sean


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[asterisk-users] 11.21.0 : echo woes : can't install canceller

2016-01-28 Thread sean darcy 
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 extension, no echo.

I've built /lib/modules/4.3.3-303.fc23.x86_64/dahdi/dahdi_echocan_oslec.ko

And requested oslec echo cancel:

grep oslec /etc/dahdi/system.conf
echocanceller=oslec,1,2,4

grep echo /etc/asterisk/chan_dahdi.conf
echocancel=yes
echocancelwhenbridged=no
echotraining=yes

but it's never loaded:

# lsmod | grep echo
[root@asterisk ~]#

dahdi_cfg -vvv
DAHDI Tools Version - 2.10.0

DAHDI Version: 2.11.0
Echo Canceller(s):
Configuration
==


Channel map:


0 channels to configure.

I can manually insert the oslec module using modprobe.

Thats seems to work.

CLI> dahdi show version
DAHDI Version: 2.11.0 Echo Canceller: OSLEC

But it's not persistent across reboots.

sean


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[asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API

2016-01-28 Thread Sonny Rajagopalan 
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.
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Re: [asterisk-users] asterisk 13 mixmonitor - random missing syllables

2016-01-28 Thread Brian :: 
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 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 load, there are sometimes missing syllable
>
> there arent BIG spikes on cpus
> recordings are to ramdisk (/dev/shm)
>
> any hints?
>
 First, try recording to a real disk  (preferrably a separate drive, so
 nothing else will be seeking the heads about; and connected by SATA, not
 USB, for full speed).  Does that work any better?

>>> i tried before. IO is not the problem
>>>
>> Are you saying that it records fine when you use a real disk, but not
>> with a
>> ramdisk?
>>
>> And why are you using a ramdisk for your mixmonitor recordings?
>>
>>
> i have problem in both scenarios
> im using ramdisk because is faster and IO cannot be problem
>
> --
> ---
> Marek Cervenka
> ===
>
>
> --
> _
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>
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Re: [asterisk-users] asterisk 13 mixmonitor - random missing syllables

2016-01-28 Thread Marek Červenka 

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 load, there are sometimes missing syllable

there arent BIG spikes on cpus
recordings are to ramdisk (/dev/shm)

any hints?

First, try recording to a real disk  (preferrably a separate drive, so
nothing else will be seeking the heads about; and connected by SATA, not
USB, for full speed).  Does that work any better?

i tried before. IO is not the problem

Are you saying that it records fine when you use a real disk, but not with a
ramdisk?

And why are you using a ramdisk for your mixmonitor recordings?



i have problem in both scenarios
im using ramdisk because is faster and IO cannot be problem

--
---
Marek Cervenka
===


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Re: [asterisk-users] PJSIP Stun/ICE

2016-01-28 Thread James Cloos 
> "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 Cloos  OpenPGP: 0x997A9F17ED7DAEA6





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[asterisk-users] Caller ID Sent in PAI header.

2016-01-28 Thread Aziz TestAccount 
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.
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Re: [asterisk-users] Caller ID Sent in PAI header.

2016-01-28 Thread Laurent Schweizer 
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
Envoyé : jeudi 28 janvier 2016 15:46
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] Caller ID Sent in PAI header.

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.
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Re: [asterisk-users] PJSIP Stun/ICE

2016-01-28 Thread George Joseph 
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 (including here) static ip is not affordable.
>

​Please create a JIRA issue and let me know what the number is.  I've just
posted a patch for review that allows reloading transports from the command
line.​  I'd like to know what else you actually need.
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Re: [asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API

2016-01-28 Thread George Joseph 
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 the Asterisk Manager Interface to ​modify the config files and
reload.
https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=4817239
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Re: [asterisk-users] Caller ID Sent in PAI header.

2016-01-28 Thread Aziz TestAccount 
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 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
> *Envoyé :* jeudi 28 janvier 2016 15:46
> *À :* asterisk-users@lists.digium.com
> *Objet :* [asterisk-users] Caller ID Sent in PAI header.
>
>
>
> 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.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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>
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