Re: [asterisk-users] Need help for Aculab Prosody X PCI card installation and configuration with Asterisk

[Mitul Limbani]

AFAIK its a propreitary card from Aculab and wont work on Asterisk unless
you buy software or support or both from them.

My advice is to dump it n get a digium card in same or lesser cost which
you need to pay aculab.

Mitul Limbani
On Nov 12, 2012 1:23 PM, "RAJNI VANZA"  wrote:

> Hi All,
>
> I need to install and configuration of Aculab prosody X PCI card with
> Asterisk-1.8.9.1 on Centos-5.7 system.
>
> I will try for that but not success. so, please suggest me way to achieve
> it.
>
> Thanks in Advance.
>
> --
> Best Regards,
> Rajni Vanza
>
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[asterisk-users] 回覆︰ Asterisk SIP authenticate using Radius / LDAP

[kingman chui]

HI,
  I have connect to Radius with asterisk 1.8.11 before.
For CDR, I use the cdr_radius and the cdr can write to radius server.
 
For auth with radius server, I use php-radius to write php script and use agi 
in dialplan to auth the account .
 
It is work ..
 
 
Regard/chui king man

寄件人︰ "qasimak...@gmail.com" 
>收件人︰ Samira Hosseini ; Asterisk Users Mailing List 
>- Non-Commercial Discussion  
>傳送日期︰ 2012年11月12日 (週一) 3:50 PM
>主題︰ Re: [asterisk-users] Asterisk SIP authenticate using Radius / LDAP
>
>
>You can use Radius Agi developed by PortaOne from following link.
>
>http://www.voip-info.org/wiki/view/PortaOne+Radius+auth
>
>Regards,
>Qasim
>
>
>
>
>On Mon, Nov 12, 2012 at 11:24 AM, Samira Hosseini  
>wrote:
>
>
>>
>>Hi all,
>>based on the following link, I am going to authenticate SIP asterisk users 
>>via Radius client that is installed on my Asterisk then the radius client 
>>connect to asterisk using the radius and ldap: 
>>https://who.rocq.inria.fr/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html#AEN237
>>
>>
>>
>>So I want to know for implementing the mentioned authentication method I need 
>>to use the patched asterisk as follow :
>>https://issues.asterisk.org/jira/browse/ASTERISK-5278?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel
>>
>>
>>
>>Thanks.
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>>
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[asterisk-users] Make asterisk voicemail use say.conf for saying CID

[OCEANET - Cédric BASSAGET]

Hello,

As a feature request, I think it would be really interesting to make 
asterisk voicemail system able to user patterns defined in say.conf.


Actually, only single digits are played to the caller. In many countries 
(example : France), phone numbers are sayed like "zero one twenty-three 
fourty-five sixty-seven..." instead of "zero one two three four five six 
seven...".


I've posted on asterisk jira as a feature request, and asked on IRC. 
People told me to post on asterisk-users ML.


refs :
- ASTERISK-20657
- ASTERISK-20661

Thanks for your replies.
Cédric


--
OCEANET
---
[AGENCE DU MANS]
7, rue des Frênes
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72190 SARGE LES LE MANS
[t] +33 (0)2.43.50.26.50
[f] +33 (0)2.43.72.21.14

[AGENCE D'ANGERS]
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[t] +33 (0)2.41.19.28.65
[f] +33 (0)2.52.19.22.00

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http://www.oceanet-telecom.com



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[asterisk-users] 回覆︰ 回覆︰ Asterisk SIP authenticate using Radius / LDAP

[kingman chui]

I try before . I use asterisk 1.8.11 , I cannot compile the patch under 
asterisk 1.8.11 .
So, it isn ot work in asterisk 1.8.11 an Use php-radius to write ph p script  
to auth radius server 
Please advice other method to auth with radius server under asterisk 1.8.11 if 
you know ..
 
Thank
Regard/chui king man

寄件人︰ "s...@yahoo.com" 
>收件人︰ kingman chui  
>傳送日期︰ 2012年11月12日 (週一) 6:22 PM
>主題︰ Re: 回覆︰ [asterisk-users] Asterisk SIP authenticate using Radius / LDAP
>
>
>Hello, thanks for your help,
>but do you think I will able to connect asterisk(that is installed radius 
>client on it) to the radius server by the following link?
>https://issues.asterisk.org/jira/browse/ASTERISK-5278?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel
>
>
>
>From: kingman chui 
>>To: Asterisk Users Mailing List - Non-Commercial Discussion 
>>; Samira Hosseini 
>> 
>>Sent: Monday, 12 November 2012, 12:26:32
>>Subject: 回覆︰ [asterisk-users] Asterisk SIP authenticate using Radius / LDAP
>>
>>
>>HI,
>>  I have connect to Radius with asterisk 1.8.11 before.
>>For CDR, I use the cdr_radius and the cdr can write to radius server.
>> 
>>For auth with radius server, I use php-radius to write php script and use agi 
>>in dialplan to auth the account .
>> 
>>It is work ..
>> 
>> 
>>Regard/chui king man
>>
>>
>>寄件人︰ "qasimak...@gmail.com" 
>>>收件人︰ Samira Hosseini ; Asterisk Users Mailing 
>>>List - Non-Commercial Discussion  
>>>傳送日期︰ 2012年11月12日 (週一) 3:50 PM
>>>主題︰ Re: [asterisk-users] Asterisk SIP authenticate using Radius / LDAP
>>>
>>>
>>>You can use Radius Agi developed by PortaOne from following link.
>>>
>>>http://www.voip-info.org/wiki/view/PortaOne+Radius+auth
>>>
>>>Regards,
>>>Qasim
>>>
>>>
>>>
>>>
>>>On Mon, Nov 12, 2012 at 11:24 AM, Samira Hosseini 
>>> wrote:
>>>
>>>

Hi all,
based on the following link, I am going to authenticate SIP asterisk users 
via Radius client that is installed on my Asterisk then the radius client 
connect to asterisk using the radius and ldap: 
https://who.rocq.inria.fr/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html#AEN237



So I want to know for implementing the mentioned authentication method I 
need to use the patched asterisk as follow :
https://issues.asterisk.org/jira/browse/ASTERISK-5278?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel



Thanks.
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>>>
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>>>
>>
>>
>
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Re: [asterisk-users] Asterisk 1.8.16 Monitoring tools

[Lenz Emilitri]

Hello Motty,
it really depends on what you want to do and the level of detail you want.
There are a number of free and commercial applications that can help you in
doing this :)
l.


2012/11/9 motty.cruz 

> Hello,
> I want to monitor my Asterisk 1.8, inbound, outbound, status calls, queue
> call? Any suggestions?
>
> I found Monast, I'm having issues configurating.
>
> Thanks,
>
>
-- 
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Test-drive WombatDialer beta @ http://wombatdialer.com
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[asterisk-users] Astricon 2012 presentations

[Lenz Emilitri]

Hello all,
anybody knows if the PDFs for presentations held at Astricon 2012 are
available somewhere? I looked at the website but cannot find anything.
Thanks
l.


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Re: [asterisk-users] Astricon 2012 presentations

[Dan Jenkins]

Hi,

As far as I'm aware the videos are still being produced and there's no
definitive list anywhere for the slide decks.

However, my one is here:
http://www.slideshare.net/danjenkins/asterisk-html5-and-nodejs-a-world-of-endless-possibilities-14881614

Dan Jenkins

-- 
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email: dan.jenk...@holidayextras.com
twitter: dan_jenkins 
linkedin: jenkinsdaniel 
skype: d-jenkins
blog: www.dan-jenkins.co.uk
about.me: about.me/dan_jenkins



On 12 November 2012 11:05, Lenz Emilitri  wrote:

> Hello all,
> anybody knows if the PDFs for presentations held at Astricon 2012 are
> available somewhere? I looked at the website but cannot find anything.
> Thanks
> l.
>
>
> --
> Loway - home of QueueMetrics - http://queuemetrics.com
> Test-drive WombatDialer beta @ http://wombatdialer.com
>
>
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Re: [asterisk-users] how to lookup a call

[Lenz Emilitri]

I would not know if this is something that can be helpful to you, but in
WombatDialer we associate a channel variable to with an unique-id to each
call, so that we can reattach to a set of calls if the AMI connection goes
down and we can be absolutely sure that what we are looking at is the call
we think it is. It is not really expensive to do - just a GetVar per
channel to mek sure our assumptions are correct.


2012/11/7 Jerry Geis 

> I am using 1.4.43 currently.
>
> I am using the AMI to originate a call over a SIP Trunk to my cell
> XXX506. works fine.
> when the call is active I do a "core show channels concise" and I get:
>
> SIP/testsystem-0ad0!**smvoice-dialout!callprogress!**
> 4!Up!AGI!smvoice!0!!3!24!(**None)
>
> My AGI is called smvoice.
> No place does my number show up.
> How do I "lookup" my call so I can "hangup" the call at a later time.
>
> In my case there my be more than one call active at a time, and I want to
> hangup the correct call. I know I need the data "testsystem-0ad0" to
> cancel my call
> but how do I "associate" that with my number so I can find the right call
> to hangup.
>
> Thanks,
>
> Jerry
>
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Re: [asterisk-users] Random crash of the machine ? due to Asterisk 11

[Matthew Jordan]

On 11/08/2012 12:25 AM, asterisk asterisk wrote:
> No, I put it in Xen VPS with Centos 5.8. Only things I added are skype
> support using siptosis and java.
> 
> Asterisk 11 is complied with no issue, siptosis and skype call no
> issues. But hangs unexpectedly.
> 
> Any clue is welcome?
> 

Are you experiencing a crash, or is Asterisk becoming unresponsive?
Those would be two different error conditions.

Either way, please file an issue on the issue tracker:
https://issues.asterisk.org/jira

If you haven't already, please read the issue reporting guidelines on
the wiki.  This will help reduce the time your issue is in Triage:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

If you are experiencing a crash, you may also want to read up on how to
get a backtrace:

https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

Thanks!

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org



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Re: [asterisk-users] Asterisk crashing when trying to start a Jabber session with ejabberd

[Matthew Jordan]

On 11/11/2012 06:35 AM, Phil Reynolds wrote:
> On Sat, 10 Nov 2012 18:08:58 +
> Phil Reynolds  wrote:
> 


Asterisk should not crash.  If you are using a supported version of
Asterisk (1.8+), please file an issue on the issue tracker:

https://issues.asterisk.org/jira

Please read the issue reporting guidelines on the wiki here:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

Since your issue involves a crash, you will be asked to obtain a
backtrace.  This page details how to go about getting a valid backtrace
for your issue:

https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

Thanks!


-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org



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Re: [asterisk-users] DAHDI 1.4 on Kernel 3.0

[Alyed]

Thanks a lot for the link and the tip. Have been trying it these days and
think it wil work on my system.

Thanks again Shaun.

2012/11/8 Shaun Ruffell 

> On Tue, Nov 06, 2012 at 06:49:09PM -0600, Alyed wrote:
> > Hello listers,
> >
> > I'm trying to run DAHDI 1.4 on a 3.0 Debian Kernel in an embedded system,
> > but have faced lots of problems mainly because it has lots of functions
> > looking for the PCI.
> >
> > Have seen so many problems, I'm in fact thinking it cannot be possibly
> done
> > (at least not in a couple of weeks, by one only man). Has anyone out
> there
> > had any experience on something like this? or can someone shed some light
> > on how to overcome this issues?
> >
> > Any ideas are very welcome
>
> There isn't a 1.4 version of DAHDI. However version v2.6.0 will not
> build any PCI drivers if the Kernel does not have the PCI bus
> configured.
>
> [1] http://svnview.digium.com/svn/dahdi?view=revision&revision=10397
>
> --
> Shaun Ruffell
> Digium, Inc. | Linux Kernel Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: www.digium.com & www.asterisk.org
>
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[asterisk-users] Can I make asterisk do inband and rfc2833 at the same time?

[Matthew Schumacher]

I know I wouldn't normally want this due to double tones, but my
upstream provider has an issue where they negotiate rfc2833 but then
send dtmf inband.  I don't expect to get both at the same time, so is
there a way to make asterisk turn on both inband or rfc2833?  Auto
doesn't work because it sees the rfc2833 in SDP then ignores inband for
the remainder of the call.

Thanks.

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Re: [asterisk-users] Can I make asterisk do inband and rfc2833 at the same time?

[Danny Nicholas]

Have you tried auto instead?  This should do the switch when needed?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew
Schumacher
Sent: Monday, November 12, 2012 1:04 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Can I make asterisk do inband and rfc2833 at the
same time?

I know I wouldn't normally want this due to double tones, but my upstream
provider has an issue where they negotiate rfc2833 but then send dtmf
inband.  I don't expect to get both at the same time, so is there a way to
make asterisk turn on both inband or rfc2833?  Auto doesn't work because it
sees the rfc2833 in SDP then ignores inband for the remainder of the call.

Thanks.

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[asterisk-users] Hangup problems

[Agustina Berretta]

Hello!!
I have asterisk 1.6.2.10 and whenever there are more than 60 calls queued
the following problem ocurrs.
The agents hangup the calls but the do not receive new calls for some
seconds or even minutes.

If I seek throughout the full log I encounter that the Bye message coming
from asterisk to the agents failes to arrive.
If I make calls between extensions what I see is the following:

Extension A calls extension B.
Extension A hangups.
Bye message is send from extension A to asterisk.
Asterisk sends an Ok message to extension A.
After some seconds or even minutes, extension B receives Bye message from
asterisk.

Any help will be appreciated!!

Also I see messages like:

 Line 71344: [Nov  5 10:58:55] VERBOSE[19043] chan_sip.c: Scheduling
destruction of SIP dialog
'065e1e5e612fcbf947d4f9044dd8c...@xxx.xxx.xxx.xxx'in 6464 ms (Method:
INVITE)
 Line 71347: [Nov  5 10:58:55] VERBOSE[19043] chan_sip.c: set_destination:
Parsing  for
address/port to send to
 Line 71348: [Nov  5 10:58:55] VERBOSE[19043] chan_sip.c: set_destination:
set destination to XXX.XXX.XXX.XXX, port 
 Line 71349: [Nov  5 10:58:55] VERBOSE[19043] chan_sip.c: Reliably
Transmitting (NAT) to XXX.XXX.XXX.XXX:XXX:

Any body can tell m what these Scheduling destruction of SIP dialog
messages mean?

Thanks!!!
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[asterisk-users] asterisk and ericsson apg212-60 caller id issue

[Rafael Visser]

Hi gurus:
I have an asterisk workging fine with an ericsson apg212-60, the thing is
that when asterisk dials to the ericsson the callerid is not shown on
ericsson's network. The oposite works!!
Do you have any idea to solve this issue?
Thanks
rv
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Re: [asterisk-users] (no subject)

[Joseph Schwartz]

check this out http://msnbc.msn.com-report6.us/finance/--
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Re: [asterisk-users] Asterisk crashing when trying to start a Jabber session with ejabberd

[Phil Reynolds]

On Mon, 12 Nov 2012 09:49:37 -0600
Matthew Jordan  wrote:

> On 11/11/2012 06:35 AM, Phil Reynolds wrote:
> > On Sat, 10 Nov 2012 18:08:58 +
> > Phil Reynolds  wrote:
> > 
> 
> 
> Asterisk should not crash.  If you are using a supported version of
> Asterisk (1.8+), please file an issue on the issue tracker:
> 
> https://issues.asterisk.org/jira

It turns out to be a known issue:

https://issues.asterisk.org/jira/browse/ASTERISK-19532

... and can be fixed by applying the patch at:

https://issues.asterisk.org/jira/secure/attachment/43441/xmpp_no_crash_with_ejabberd.patch

I will file the details with Debian too...

-- 
Phil Reynolds
mail: phil-aster...@tinsleyviaduct.com
Web: http://phil.tinsleyviaduct.com/


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