[asterisk-users] h323 channel compile error

[Shehzad Pankhawala]

I have following settings done on my Fedora8:
Downloaded
openh323-v1_19_0_1-src-tar.gz
pwlib-v1_11_1-src.tar.gz
Extracted them in /root/openh323 and /root/pwlib

Exported the following variables:
PWLIBDIR=/root/pwlib
export PWLIBDIR
OPENH323DIR=/root/openh323
export OPENH323DIR
LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib
export LD_LIBRARY_PATH


Then I compiled pwlib and it was fine.

But in compilation of openh323 as below i got the error, which failed to 
find solution in any forum:
./configure
make

--
make P_SHAREDLIB=1 opt
make[1]: Entering directory `/root/openh323'
make -C src opt
make[2]: Entering directory `/root/openh323/src'
g++ -D_REENTRANT -Wall -fPIC -DPIC -DPTRACING -I/root/openh323/include 
-I/root/pwlib/include -Os -felide-constructors -Wreorder -c h323ep.cxx 
-o /root/openh323/lib/obj_linux_x86_r/h323ep.o
/root/openh323/include/h4601.h: In member function 
‘H460_FeatureContent::operator H460_FeatureTable*()’:
/root/openh323/include/h4601.h:292: warning: type-punning to incomplete 
type might break strict-aliasing rules
h323ep.cxx: In constructor ‘H323EndPoint::H323EndPoint()’:
h323ep.cxx:1001: error: ‘PSoundChannel’ has not been declared
h323ep.cxx:1001: error: ‘PSoundChannel’ has not been declared
h323ep.cxx:1002: error: ‘PSoundChannel’ has not been declared
h323ep.cxx:1002: error: ‘PSoundChannel’ has not been declared
h323ep.cxx: In member function ‘virtual BOOL 
H323EndPoint::OpenAudioChannel(H323Connection&, BOOL, unsigned int, 
H323AudioCodec&)’:
h323ep.cxx:2841: error: ‘PSoundChannel’ was not declared in this scope
h323ep.cxx:2841: error: ‘soundChannel’ was not declared in this scope
h323ep.cxx:2843: error: ‘PSoundChannel’ is not a class or namespace
h323ep.cxx:2845: error: expected type-specifier before ‘PSoundChannel’
h323ep.cxx:2845: error: expected `;' before ‘PSoundChannel’
h323ep.cxx:2854: error: ‘PSoundChannel’ is not a class or namespace
h323ep.cxx:2855: error: ‘PSoundChannel’ is not a class or namespace
h323ep.cxx:2869: error: type ‘’ argument given to ‘delete’, 
expected pointer
h323ep.cxx: In member function ‘virtual BOOL 
H323EndPoint::SetSoundChannelPlayDevice(const PString&)’:
h323ep.cxx:3047: error: ‘PSoundChannel’ has not been declared
h323ep.cxx:3047: error: ‘PSoundChannel’ has not been declared
h323ep.cxx: In member function ‘virtual BOOL 
H323EndPoint::SetSoundChannelRecordDevice(const PString&)’:
h323ep.cxx:3057: error: ‘PSoundChannel’ has not been declared
h323ep.cxx:3057: error: ‘PSoundChannel’ has not been declared
h323ep.cxx: In member function ‘virtual BOOL 
H323EndPoint::SetSoundChannelPlayDriver(const PString&)’:
h323ep.cxx:3074: error: ‘PSoundChannel’ has not been declared
h323ep.cxx:3074: error: ‘PSoundChannel’ has not been declared
h323ep.cxx: In member function ‘virtual BOOL 
H323EndPoint::SetSoundChannelRecordDriver(const PString&)’:
h323ep.cxx:3091: error: ‘PSoundChannel’ has not been declared
h323ep.cxx:3091: error: ‘PSoundChannel’ has not been declared
make[2]: *** [/root/openh323/lib/obj_linux_x86_r/h323ep.o] Error 1
make[2]: Leaving directory `/root/openh323/src'
make[1]: *** [opt] Error 2
make[1]: Leaving directory `/root/openh323'
make: *** [optshared] Error 2
__
I also tried make opt
but the error remain same.
Any idea.
Please reply,
Thanks
Shehzad.


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[asterisk-users] BRI AND DATA connection

[Anton]

Hello!

Does anyone tried BRI with asterisk for DATA transfer? My 
customer
wants BRI connection, but he wants it for the data, and I 
have to
bring connection to his office, so I see the connection as 
follows:

E1<->(CORE_ASTERISK)<->(IAX2)<->(EDGE_ASTERISK)<->BRI - so 
will data
work in such scenario? If no what scenario would be 
suggested?

Regards!

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Re: [asterisk-users] does astcanary really work?

[Tilghman Lesher]

On Wednesday 06 August 2008 04:09:13 Pavel Jezek wrote:
> A week ago, I tried give realtime priority to asterisk proces using -p
> switch,
> asterisk was running inside astcanary,
> but yestarday asterisk probably starts eating all cpu and lock any
> access to computer, only ping was possible,
> so, anybody have experience, that ascanary process does really work to
> lower process priority in case of overloading?

Not sure, but it should work if the sole issue is that the realtime process is
sitting in a busy loop.  Note that if the machine is inaccessible for other
reasons, like a kernel oops, then astcanary can do nothing about that.

Note that a Linux machine being pingable says nothing about the status
of the kernel.  A Linux machine will still reply to pings, even with the
kernel halted -- it's one of the rather strange behaviors peculiar to that
kernel.

-- 
Tilghman

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Re: [asterisk-users] AGI and Call Center to do CRM integration

[Matt Florell]

We have done this several times for customers with VICIDIAL. I have
also seen companies use AGI scripts to enable this kind of application
as well. So, yes it is possible.

MATT---

On 8/7/08, bilal ghayyad <[EMAIL PROTECTED]> wrote:
> CRM: Customer Record Module which is any kind of application.
>
>  For example, a bank has an application and the agent sit on his PC, when 
> call come, the application fetched with the customer information based on the 
> card number which is entered with the IVR, 
>
>  How the application of the bank was able to fetch the infomation? It was 
> passing to it from the call center.
>
>  Also another example: when call come to call center, and before call routing 
> to the proper skill group, then we need to check the data related to caller, 
> based on these data we determine which skill group need to be routed, how 
> this to be done? AGI can do?
>
>  Regards
>  Bilal
>
>
>  --- On Thu, 8/7/08, Steve Totaro <[EMAIL PROTECTED]> wrote:
>
>  > From: Steve Totaro <[EMAIL PROTECTED]>
>  > Subject: Re: [asterisk-users] AGI and Call Center to do CRM integration
>  > To: [EMAIL PROTECTED], "Asterisk Users Mailing List - Non-Commercial 
> Discussion" 
>  > Date: Thursday, August 7, 2008, 6:12 PM
>
> > On Thu, Aug 7, 2008 at 5:55 PM, bilal ghayyad
>  > <[EMAIL PROTECTED]> wrote:
>  > > Hi All;
>  > >
>  > > Did anyone used AGI to do te CRM integration in the
>  > Asterisk call center?
>  > >
>  > > If yes, I would like to know the overview to know from
>  > where to start?
>  > >
>  > > Regards
>  > > Bilal
>  > >
>  >
>  > What CRM?  FastAGI to hit a box that has logic to update
>  > the backend
>  > DB of the CRM.
>  >
>  > Thanks,
>  > Steve Totaro
>
>
>
>
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[asterisk-users] FWD $30 membership-fee

[Dave Platt]

> I just received an email notice from FWD about $30 membership fee.
> My question: Is the email genuine? Did anybody else receive it?
> 
> I'm just trying to be sure that it is real and not a scam.
> The (FWD) does not do anything to authenticate such emails (implementing 
> GPG/PGP signature etc.)
> 
> If the email is genuine, I hope they will improve their service; as of now 
> their IAX server is not working.

I received the same email.  It was sent to a tagged email address that
I had used only when signing up for FWD last month, and never gave to
anyone else.

It seems to have been sent out through a mass-mailing service (MagnetMail)...
they feel slightly on the "this might possibly be an outfit whose
service could be used for spamming" side of things but are not overtly
crooked or bogus as far as I can tell.

My guess is that the email is probably legitimate.  Although there's
nothing new on the FWD website to confirm that free accounts are
going away (as the email implies), the statements in the email seem
consistent with the info on the FWD website about a "re-launch" and
"spin-off".

Any service such as FWD is going to need a source of revenue in order
to keep their servers running.  I don't think FWD ever got around
to launching an FWD-based "dial out onto the PSTN" service with
a per-minute charge from which they could earn revenue... and the
VOIP market for such things seems to be extremely competitive and
perhaps cut-throat.  Since FWD seems to be spinning off and will need
independent revenue, they're likely to have to do *something* to keep
the lights on!

The interesting thing will be to see if people find it worthwhile
to pay $30/year, primarily for a SIP registration service that doesn't
provide either outdial-to-PSTN or indial-from-PSTN.  If not, FWD might
cease to exist.

If the email is legit, I do think that FWD really ought to update their
web site Real Soon Now, to reflect that the services now available
"for free" will no longer be free.



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Re: [asterisk-users] AGI and Call Center to do CRM integration

[bilal ghayyad]

CRM: Customer Record Module which is any kind of application.

For example, a bank has an application and the agent sit on his PC, when call 
come, the application fetched with the customer information based on the card 
number which is entered with the IVR, 

How the application of the bank was able to fetch the infomation? It was 
passing to it from the call center.

Also another example: when call come to call center, and before call routing to 
the proper skill group, then we need to check the data related to caller, based 
on these data we determine which skill group need to be routed, how this to be 
done? AGI can do?

Regards
Bilal


--- On Thu, 8/7/08, Steve Totaro <[EMAIL PROTECTED]> wrote:

> From: Steve Totaro <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] AGI and Call Center to do CRM integration
> To: [EMAIL PROTECTED], "Asterisk Users Mailing List - Non-Commercial 
> Discussion" 
> Date: Thursday, August 7, 2008, 6:12 PM
> On Thu, Aug 7, 2008 at 5:55 PM, bilal ghayyad
> <[EMAIL PROTECTED]> wrote:
> > Hi All;
> >
> > Did anyone used AGI to do te CRM integration in the
> Asterisk call center?
> >
> > If yes, I would like to know the overview to know from
> where to start?
> >
> > Regards
> > Bilal
> >
> 
> What CRM?  FastAGI to hit a box that has logic to update
> the backend
> DB of the CRM.
> 
> Thanks,
> Steve Totaro


  

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Re: [asterisk-users] G722 capable soft phone?

[marek cervenka]

> Does anyone know where I might purchase a G.722 capable SIP soft phone?
> Counterpath say that they offer one, but only in the OEM versions do
> they support G.722. I need only a couple of licenses.

www.qutecom.org

---
Marek Cervenka
===


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Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 19

[dimitri . osler]

Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a 
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al 
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], 
altrimenti vi risponderò al mio rientro.

Dimitri Osler

I will be on vacation until Tuesday 19th of August with limited access to voice 
and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 
0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on 
my return.

Dimitri Osler





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Re: [asterisk-users] FAX t.38 on Asterisk 1.6?

[Arturo Ochoa]

Thanks Memo,

I've already see that article before, the problem is that this solution is
useful when you want asterisk (via t38modem) to terminate the call...
Someone send you a Fax using t.28 and this software
(t38modem+asterisk+hylafax) will handle the incomming fax. In fact I have a
working installation of Iaxmodem+asterisk+hylafax working.

The problem seems to be on the implementation of some application to handle
the t38 gateway capability of the Asterisk Server.

I've read this article http://bugs.digium.com/view.php?id=12931 but it's
closed.

Any Ideas?



Ing. Arturo Ochoa N
Electrosystems S RL 
Tel. (656)-6230794
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Guillermo
Salas M.
Enviado el: Thursday, August 07, 2008 2:00 PM
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] FAX t.38 on Asterisk 1.6?

El jue, 07-08-2008 a las 13:31 -0600, Arturo Ochoa escribió:
> Has anyone have experiencies on this kind of scenario... what
> version?.. patches?... or any information regarding this goal will be
> VERY helpful...


Hi Arturo,

Please ckeck the following URL (on spanish):

http://www.sinologic.net/2008-07/como-configurar-un-fax-virtual-t38-en-aster
isk/


Regards,

-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net
SIP  : [EMAIL PROTECTED]
FWD  : 558563
USA  : 1 360 968 1701

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting


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Re: [asterisk-users] Asterisk end-user GUI?

[Ming Yong]

Ken,
Druid is based on Asterisk and we love asterisk for the call control
functionality. We built value on top of asterisk by extending
functionalities to the unified communications space (e-fax, mobility,
sugarcrm integration, google apps integration, auto-provisioning of 5
brand phones)

Check out this review that mentions how we actually preserve the
configuration file
http://blog.tmcnet.com/blog/tom-keating/asterisk/voiceroute-druid-open-source-edition-launches-offering-new-open-source.asp
If you would like more information about our blackberry application,
send me a mail off-list.

Ming

On Fri, Aug 8, 2008 at 6:30 AM, Ken D'Ambrosio <[EMAIL PROTECTED]> wrote:
> Wow.  Okay, Druid has my attention; I'll definitely be kicking the tires.
> That being said, though, I do have a quick question (that I always have
> about GUIs):
>
> First, I assume that Druid is based on Asterisk; is this true?
> Second, is it possible to make system modifications w/o using the GUI?  I
> love GUIs, but sometimes there's just nothing as cool as a quick Perl
> script.
>
> Thanks!
>
> -Ken
>
>> Ken,
>> You might want to check out our free Druid Open source unified
>> communications project. It is not proprietary and has open source soap
>> API for third party applications.
>> http://www.voiceroute.org
>> We have mobile integration with blackberry & iphone that no vendors
>> open source or otherwise has. Our Druid SOAP API that powers this
>> integration is free.
>> Check out our youtube oscon presentation on Druid & SOAP API
>> http://www.youtube.com/user/voiceroute
>> Ming
>>
>>
>>
>> On 8/7/08, Ken D'Ambrosio <[EMAIL PROTECTED]> wrote:
>>> I badly want to roll out Asterisk at my job.  Unfortunately, my boss is
>>> dazzled by shiny objects.  We had a vendor in today who showed us their
>>> system which, honestly, didn't suck -- but boy, is it going to be
>>> expensive!  One major component of the eye candy was an end-user
>>> interface
>>> that allowed the user to initiate calls to a contact list, check for
>>> presence, create conferences, etc.  Is there anything like that, aimed
>>> at
>>> end-users (as opposed to admins) for Asterisk?  I'd even be willing to
>>> go
>>> with proprietary; I just don't want a wholly-proprietary, hobbled,
>>> licensed-to-Heck-and-back system, which is where it looks like my boss
>>> is
>>> leaning.
>>>
>>> Thanks!
>>>
>>> -Ken
>>>
>>>
>>> ___
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>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>> --
>> Sent from Gmail for mobile | mobile.google.com
>>
>> Ming Yong
>> CEO, www.voiceroute.org
>> Druid - Open Source Unified Communications
>> DID: +1-877-242-3704
>> Office: +1-866-915-2407 ext 301
>> SIP/email: [EMAIL PROTECTED]
>> --
>> Meet us at LinuxWorld 2008, 4-7 Aug 2008, Moscone Center, San
>> Francisco, Booth 1626
>> http://druidlinuxworld.eventbrite.com
>>
>> Meet us at WEB 2.0 EXPO, 16-19 Sept 2008, Javits Center, NYC, Booth #17
>> http://druidweb20.eventbrite.com
>>
>> See Voiceroute OSCON 2008 Druid project presentation on youtube
>> http://www.youtube.com/watch?v=2gfIAXm5vTc
>> http://www.youtube.com/watch?v=dkm6P4O0oac
>>
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>



-- 
Ming Yong
CEO, www.voiceroute.org
Druid - Open Source Unified Communications
DID: +1-877-242-3704
Office: +1-866-915-2407 ext 301
SIP/email: [EMAIL PROTECTED]
--
Meet us at LinuxWorld 2008, 4-7 Aug 2008, Moscone Center, San
Francisco, Booth 1626
http://druidlinuxworld.eventbrite.com

Meet us at WEB 2.0 EXPO, 16-19 Sept 2008, Javits Center, NYC, Booth #17
http://druidweb20.eventbrite.com

See Voiceroute OSCON 2008 Druid project presentation on youtube
http://www.youtube.com/watch?v=2gfIAXm5vTc
http://www.youtube.com/watch?v=dkm6P4O0oac

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Re: [asterisk-users] Asterisk end-user GUI?

[Ken D'Ambrosio]

[Sorry if this is a duplicate; originally sent from an address the list
doesn't "know."]

Wow.  Okay, Druid has my attention; I'll definitely be kicking the tires. 
That being said, though, I do have a quick question (that I always have
about GUIs):

First, I assume that Druid is based on Asterisk; is this true?
Second, is it possible to make system modifications w/o using the GUI?  I
love GUIs, but sometimes there's just nothing as cool as a quick Perl
script.

Thanks!

-Ken

> Ken,
> You might want to check out our free Druid Open source unified
> communications project. It is not proprietary and has open source soap
API for third party applications.
> http://www.voiceroute.org
> We have mobile integration with blackberry & iphone that no vendors open
source or otherwise has. Our Druid SOAP API that powers this integration
is free.
> Check out our youtube oscon presentation on Druid & SOAP API
> http://www.youtube.com/user/voiceroute
> Ming
>
>
>
> On 8/7/08, Ken D'Ambrosio <[EMAIL PROTECTED]> wrote:
>> I badly want to roll out Asterisk at my job.  Unfortunately, my boss is
dazzled by shiny objects.  We had a vendor in today who showed us their
system which, honestly, didn't suck -- but boy, is it going to be
expensive!  One major component of the eye candy was an end-user
interface
>> that allowed the user to initiate calls to a contact list, check for
presence, create conferences, etc.  Is there anything like that, aimed
at
>> end-users (as opposed to admins) for Asterisk?  I'd even be willing to go
>> with proprietary; I just don't want a wholly-proprietary, hobbled,
licensed-to-Heck-and-back system, which is where it looks like my boss
is
>> leaning.
>>
>> Thanks!
>>
>> -Ken
>>
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> --
> Sent from Gmail for mobile | mobile.google.com
>
> Ming Yong
> CEO, www.voiceroute.org
> Druid - Open Source Unified Communications
> DID: +1-877-242-3704
> Office: +1-866-915-2407 ext 301
> SIP/email: [EMAIL PROTECTED]
> --
> Meet us at LinuxWorld 2008, 4-7 Aug 2008, Moscone Center, San
> Francisco, Booth 1626
> http://druidlinuxworld.eventbrite.com
>
> Meet us at WEB 2.0 EXPO, 16-19 Sept 2008, Javits Center, NYC, Booth #17
http://druidweb20.eventbrite.com
>
> See Voiceroute OSCON 2008 Druid project presentation on youtube
> http://www.youtube.com/watch?v=2gfIAXm5vTc
> http://www.youtube.com/watch?v=dkm6P4O0oac
>
> ___
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> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>





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Re: [asterisk-users] AGI and Call Center to do CRM integration

[Steve Totaro]

On Thu, Aug 7, 2008 at 5:55 PM, bilal ghayyad <[EMAIL PROTECTED]> wrote:
> Hi All;
>
> Did anyone used AGI to do te CRM integration in the Asterisk call center?
>
> If yes, I would like to know the overview to know from where to start?
>
> Regards
> Bilal
>

What CRM?  FastAGI to hit a box that has logic to update the backend
DB of the CRM.

Thanks,
Steve Totaro

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[asterisk-users] AGI and Call Center to do CRM integration

[bilal ghayyad]

Hi All;

Did anyone used AGI to do te CRM integration in the Asterisk call center?

If yes, I would like to know the overview to know from where to start?

Regards
Bilal


  

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Re: [asterisk-users] Voicemail on PRI

[c james]

Yann Derichard wrote:
> Hi,
> 
> I am trying to install a Voicemail on PRI after a redirection on an away 
> or a busy (a normal call which is redirected to voicemail in fact) but I 
> can't find the function in Asterisk which allow me using the phone 
> number of the callee (because I have only the number of asterisk and of 
> the caller).
> 
> Is someone could give me a clue ?
> 

I believe you are looking for RDNIS (www.voip-info.org/wiki-RDNIS).  If 
you find a vendor that supports this, please let me know also.

Clinton


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Re: [asterisk-users] Asterisk end-user GUI?

[Ming Yong]

Ken,
You might want to check out our free Druid Open source unified
communications project. It is not proprietary and has open source soap
API for third party applications.
http://www.voiceroute.org
We have mobile integration with blackberry & iphone that no vendors
open source or otherwise has. Our Druid SOAP API that powers this
integration is free.
Check out our youtube oscon presentation on Druid & SOAP API
http://www.youtube.com/user/voiceroute
Ming



On 8/7/08, Ken D'Ambrosio <[EMAIL PROTECTED]> wrote:
> I badly want to roll out Asterisk at my job.  Unfortunately, my boss is
> dazzled by shiny objects.  We had a vendor in today who showed us their
> system which, honestly, didn't suck -- but boy, is it going to be
> expensive!  One major component of the eye candy was an end-user interface
> that allowed the user to initiate calls to a contact list, check for
> presence, create conferences, etc.  Is there anything like that, aimed at
> end-users (as opposed to admins) for Asterisk?  I'd even be willing to go
> with proprietary; I just don't want a wholly-proprietary, hobbled,
> licensed-to-Heck-and-back system, which is where it looks like my boss is
> leaning.
>
> Thanks!
>
> -Ken
>
>
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-- 
Sent from Gmail for mobile | mobile.google.com

Ming Yong
CEO, www.voiceroute.org
Druid - Open Source Unified Communications
DID: +1-877-242-3704
Office: +1-866-915-2407 ext 301
SIP/email: [EMAIL PROTECTED]
--
Meet us at LinuxWorld 2008, 4-7 Aug 2008, Moscone Center, San
Francisco, Booth 1626
http://druidlinuxworld.eventbrite.com

Meet us at WEB 2.0 EXPO, 16-19 Sept 2008, Javits Center, NYC, Booth #17
http://druidweb20.eventbrite.com

See Voiceroute OSCON 2008 Druid project presentation on youtube
http://www.youtube.com/watch?v=2gfIAXm5vTc
http://www.youtube.com/watch?v=dkm6P4O0oac

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Re: [asterisk-users] Asterisk end-user GUI?

[bkruse]

I would checkout Switchvox :)

http://www.digium.com/en/products/switchvox/

-Brandon

Ken D'Ambrosio wrote:
> I badly want to roll out Asterisk at my job.  Unfortunately, my boss is
> dazzled by shiny objects.  We had a vendor in today who showed us their
> system which, honestly, didn't suck -- but boy, is it going to be
> expensive!  One major component of the eye candy was an end-user interface
> that allowed the user to initiate calls to a contact list, check for
> presence, create conferences, etc.  Is there anything like that, aimed at
> end-users (as opposed to admins) for Asterisk?  I'd even be willing to go
> with proprietary; I just don't want a wholly-proprietary, hobbled,
> licensed-to-Heck-and-back system, which is where it looks like my boss is
> leaning.
>
> Thanks!
>
> -Ken
>
>
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Re: [asterisk-users] Asterisk end-user GUI?

[Dean Collins]

Druid has a user portal that might cover what you are looking for.

Yes I think it's something that is under utilized in the current
offerings.
At one stage I was thinking of rolling out a 3rd party user portal that
would use the ami to sit over and above any asterisk platform but I
could never find anyone interested in developing it for me.



Cheers,

Dean



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken
D'Ambrosio
Sent: Thursday, 7 August 2008 4:05 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk end-user GUI?

I badly want to roll out Asterisk at my job.  Unfortunately, my boss is
dazzled by shiny objects.  We had a vendor in today who showed us their
system which, honestly, didn't suck -- but boy, is it going to be
expensive!  One major component of the eye candy was an end-user
interface
that allowed the user to initiate calls to a contact list, check for
presence, create conferences, etc.  Is there anything like that, aimed
at
end-users (as opposed to admins) for Asterisk?  I'd even be willing to
go
with proprietary; I just don't want a wholly-proprietary, hobbled,
licensed-to-Heck-and-back system, which is where it looks like my boss
is
leaning.

Thanks!

-Ken


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[asterisk-users] Asterisk end-user GUI?

[Ken D'Ambrosio]

I badly want to roll out Asterisk at my job.  Unfortunately, my boss is
dazzled by shiny objects.  We had a vendor in today who showed us their
system which, honestly, didn't suck -- but boy, is it going to be
expensive!  One major component of the eye candy was an end-user interface
that allowed the user to initiate calls to a contact list, check for
presence, create conferences, etc.  Is there anything like that, aimed at
end-users (as opposed to admins) for Asterisk?  I'd even be willing to go
with proprietary; I just don't want a wholly-proprietary, hobbled,
licensed-to-Heck-and-back system, which is where it looks like my boss is
leaning.

Thanks!

-Ken


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Re: [asterisk-users] FAX t.38 on Asterisk 1.6?

[Guillermo Salas M.]

El jue, 07-08-2008 a las 13:31 -0600, Arturo Ochoa escribió:
> Has anyone have experiencies on this kind of scenario... what
> version?.. patches?... or any information regarding this goal will be
> VERY helpful...


Hi Arturo,

Please ckeck the following URL (on spanish):

http://www.sinologic.net/2008-07/como-configurar-un-fax-virtual-t38-en-asterisk/


Regards,

-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net
SIP  : [EMAIL PROTECTED]
FWD  : 558563
USA  : 1 360 968 1701

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting


smime.p7s
Description: S/MIME cryptographic signature
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[asterisk-users] FAX t.38 on Asterisk 1.6?

[Arturo Ochoa]

Dear list,

 

I got this scenario.

 

FAX Machine -> FXS (tdm800) ->Asterisk -> SIP <-> OPENSER <-> SIP ->
Asterisk -> FXO(tdm400) -> PSTN -> FAX Machine

 

I' been reading a lot of Faxes and t.38 protocol... and I found that
Asterisk 1.6 has the possibility to do FAX t.38 Gateway funtion...and also
that CallWeaver (Asterisk Fork) also has this feature.

 

Has anyone have experiencies on this kind of scenario... what version?..
patches?... or any information regarding this goal will be VERY helpful...

 

 

 

Ing. Arturo Ochoa N

Electrosystems S RL 

Tel. (656)-6230794

 

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Re: [asterisk-users] FWD $30 membership-fee

[Steve Totaro]

Without TICC Capital investing in Pulvermedia/VON, maybe he/they need
another revenue source to pay the bills.

Thanks,
Steve T

On Thu, Aug 7, 2008 at 3:05 PM, Alex Robar <[EMAIL PROTECTED]> wrote:
> FWD has had paid membership options for years. The paid memberships help to
> improve the network and increase it's reach. As far as I've heard (and as
> far as the site mentions), paid membership is not a requirement. That would
> sort of go against the "talk... for free... for good" slogan.
>
> AR
>
> --
> Alex Robar
> [EMAIL PROTECTED]
>
>
> On Thu, Aug 7, 2008 at 2:48 PM, SIP <[EMAIL PROTECTED]> wrote:
>>
>>  From what I can ascertain, this is a way to essentially fund Jeff
>> Pulver's political agenda. I remember writing something a couple of
>> years back (
>>
>> http://neil.ideasip.com/2006/03/08/von-coalition-and-the-ideals-of-the-little-guy/
>> ) about how the VON Coalition, which is meant to be a political action
>> committee to help foster new communications, has a somewhat high barrier
>> to entry (minimum $10,000 per year).
>>
>> As far as I can tell, this FWD membership is a less expensive way for
>> people to put their money behind a similar agenda (well... okay, Jeff's
>> agenda, whatever that may be).
>>
>> The only real issue I see with it is that, a political action committee
>> is a committee. The FWD membership seems a little less transparent. It
>> could very well be a way to fund Jeff Pulver's personal vision. While
>> he's done some great things in the community, I still feel awkward with
>> the idea of funding the whole "One man. One voice. One decision. No
>> oversight" idea.
>>
>> I'm eager to see how it pans out, though.
>>
>>
>> N.
>
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Re: [asterisk-users] FWD $30 membership-fee

[Alex Robar]

FWD has had paid membership options for years. The paid memberships help to
improve the network and increase it's reach. As far as I've heard (and as
far as the site mentions), paid membership is not a requirement. That would
sort of go against the "talk... for free... for good" slogan.

AR

-- 
Alex Robar
[EMAIL PROTECTED]


On Thu, Aug 7, 2008 at 2:48 PM, SIP <[EMAIL PROTECTED]> wrote:

>
>  From what I can ascertain, this is a way to essentially fund Jeff
> Pulver's political agenda. I remember writing something a couple of
> years back (
>
> http://neil.ideasip.com/2006/03/08/von-coalition-and-the-ideals-of-the-little-guy/
> ) about how the VON Coalition, which is meant to be a political action
> committee to help foster new communications, has a somewhat high barrier
> to entry (minimum $10,000 per year).
>
> As far as I can tell, this FWD membership is a less expensive way for
> people to put their money behind a similar agenda (well... okay, Jeff's
> agenda, whatever that may be).
>
> The only real issue I see with it is that, a political action committee
> is a committee. The FWD membership seems a little less transparent. It
> could very well be a way to fund Jeff Pulver's personal vision. While
> he's done some great things in the community, I still feel awkward with
> the idea of funding the whole "One man. One voice. One decision. No
> oversight" idea.
>
> I'm eager to see how it pans out, though.
>
>
> N.
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Re: [asterisk-users] FWD $30 membership-fee

[SIP]

Gonzalo Servat wrote:
> On Thu, Aug 7, 2008 at 2:04 PM, Joseph <[EMAIL PROTECTED] 
> > wrote:
>
> I just received an email notice from FWD about $30 membership fee.
> My question: Is the email genuine? Did anybody else receive it?
>
> I'm just trying to be sure that it is real and not a scam.
> The (FWD) does not do anything to authenticate such emails
> (implementing GPG/PGP signature etc.)
>
> If the email is genuine, I hope they will improve their service;
> as of now their IAX server is not working.
>
>
> I just went to www.freeworlddialup.com 
>  and the top banner says something 
> about Paid Membership which links to:
>
> http://www.acteva.com/booking.cfm?bevaid=138192
>
> I'm not sure whether they will discontinue offering the "free" 
> accounts. I sure hope not, they would have to consider a name change 
> too. I don't really use FWD but I'm sure a lot of users would be affected.
>
> - Gonzalo

 From what I can ascertain, this is a way to essentially fund Jeff 
Pulver's political agenda. I remember writing something a couple of 
years back ( 
http://neil.ideasip.com/2006/03/08/von-coalition-and-the-ideals-of-the-little-guy/
 
) about how the VON Coalition, which is meant to be a political action 
committee to help foster new communications, has a somewhat high barrier 
to entry (minimum $10,000 per year).

As far as I can tell, this FWD membership is a less expensive way for 
people to put their money behind a similar agenda (well... okay, Jeff's 
agenda, whatever that may be).

The only real issue I see with it is that, a political action committee 
is a committee. The FWD membership seems a little less transparent. It 
could very well be a way to fund Jeff Pulver's personal vision. While 
he's done some great things in the community, I still feel awkward with 
the idea of funding the whole "One man. One voice. One decision. No 
oversight" idea.

I'm eager to see how it pans out, though.


N.

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Re: [asterisk-users] Improving the speed of chan_sip

[Mark Michelson]

Pavel Jezek wrote:
> 
> Steve Murphy wrote:
>> Hello--
>>
>> Why do I target chan_sip for so much effort?  Because, 
>> it seems to me, chan_sip is probably the most used channel 
>> driver in the asterisk community!! (and, of course,
>> the zap/dahdi driver, is also pretty popular)
>>
>> I haven't had time to follow up on chan_sip, and I probably 
>> won't for several months. 
>>
>> But, if I had time, here is what I'd do:
>>
>> There are two ways to speed up chan_sip, and they are separate issues,
>> tied together on how many cpu cycles they use up:
>>
>> 1. Call setup/teardown (invites/hangups) -- limits the calls/sec
>> asterisk can handle.
>>   
> 
> one of the big issues in sip callsetup performance, that appears to me 
> in current trunk, is about 500ms delay in propagation SIP/OK message 
> between bridged parties
> eg.: one party answers call, send SIP/OK with SDP to asterisk, asterisk 
> then forwards it to other party, but with unacceptable delay about 500ms!
> this is so much, that users complaining about lost first word of speech 
> communication,
> I posted info about this to bugreport, that seems to be related to this, 
> look at my message:
> http://bugs.digium.com/view.php?id=12708#91173
> I also attach graph picture from wireshark, that clearly ilustrated, 
> where is problem (OK-SDP-delay.png )
> PJ
> 

If you check ast_answer in channel.c of trunk, you can see that it calls 
__ast_answer(chan, 500). The 500 there is a 500 ms delay that occurs before 
calling the channel's answer callback. In the case of SIP, this would indeed 
mean that there is a 500 ms delay between receiving the 200 OK from the callee 
and sending a 200 OK to the caller.

Mark Michelson

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Re: [asterisk-users] Strange beep during calls

[Guido Hecken]

Hi Felippe,
 
in the past we had some trouble with a specific SNOM Firmware, which did not
handle dtmf tones correctly. As a workarround, we tried to set
"relaxdtmf=yes" in sip.conf.
As a result we had these "beep-tones" generated randomly.
Not shure, if this is your problem too...
 

Friendly Regards,


Guido

 

gwsNetTech
Guido Hecken

Quirrenbacher Str. 36
53639 Königswinter
Germany


fon +49(2244) 870663
fax +49(2244) 870664
mobil  +49(179) 1267353
web   http://www.gwsnettech.de
mail  [EMAIL PROTECTED]

 


  _  

Von: Felippe Silvestre [mailto:[EMAIL PROTECTED] 
Gesendet: Mittwoch, 6. August 2008 19:46
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] Strange beep during calls


Hi all,
 
Our users are complaining about beeps that happen in the middle of some
calls. They are similar to the sound heard you are in a call and press any
button in your phone. Please find bellow some examples of these beeps(the
recordings are in Portuguese, but the beeps are easy to identify):
 
 
http://www.katizak.locaweb.com.br/asterisk/beep.mp3
http://www.katizak.locaweb.com.br/asterisk/beep2.mp3
 
http://www.katizak.locaweb.com.br/asterisk/beep3.mp3
 
http://www.katizak.locaweb.com.br/asterisk/beep4.mp3
 
 
We are sure that our users are not pressing any button in the softphones
during the conversations.
Do you guys are able to identify where these beeps are coming from? Maybe an
* functionality that we need to turn off... We are using Asterisk 1.4.21.2.
 
Thanks.
 
Felippe Silvestre

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Re: [asterisk-users] FWD $30 membership-fee

[Gonzalo Servat]

On Thu, Aug 7, 2008 at 2:04 PM, Joseph <[EMAIL PROTECTED]> wrote:

> I just received an email notice from FWD about $30 membership fee.
> My question: Is the email genuine? Did anybody else receive it?
>
> I'm just trying to be sure that it is real and not a scam.
> The (FWD) does not do anything to authenticate such emails (implementing
> GPG/PGP signature etc.)
>
> If the email is genuine, I hope they will improve their service; as of now
> their IAX server is not working.
>

I just went to www.freeworlddialup.com and the top banner says something
about Paid Membership which links to:

http://www.acteva.com/booking.cfm?bevaid=138192

I'm not sure whether they will discontinue offering the "free" accounts. I
sure hope not, they would have to consider a name change too. I don't really
use FWD but I'm sure a lot of users would be affected.

- Gonzalo
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Re: [asterisk-users] problem controlling dialplan order

[Adrian Marsh]

Oh for

 

Stared at that for ages not seeing it

 

Thanks Felippe...

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Felippe
Silvestre
Sent: 07 August 2008 17:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] problem controlling dialplan order

 

Try this:

[local]

exten => _00165011091[45][0-9],1,NoOp(I AM HERE)

exten => _00165011091[45][0-9],n,Macro(setcli)

exten => _00165011091[45][0-9],n,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

exten => _00165011091[45][0-9],n,Hangup

 

The "[" before "0-9]" is needed.

 

 

 

Felippe Silvestre

 





From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adrian
Marsh
Sent: Thursday, August 07, 2008 07:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] problem controlling dialplan order

Hi All,

 

On a 1.4.15 system, I've a context as below, where I need to
catch some specific US ranges and dial direct via SIP rather than a PSTN
trunk.  But the logic always goes via the International Trunk and I cant
see why...

 

[local]

exten => _00165011091[45]0-9],1,NoOp(I AM HERE)

exten => _00165011091[45]0-9],n,Macro(setcli)

exten =>
_00165011091[45]0-9],n,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

exten => _00165011091[45]0-9],n,Hangup

 

. (same context)

 

Catch local (UK) numbers

exten => _0[1-9]X.,1,NoOp(Dialling UK number)

exten => _0[1-9]X.,n,Macro(setcli)

exten => _0[1-9]X.,n(jumpdial),Dial(SIP/+44${EXTEN:[EMAIL PROTECTED])

exten => _0[1-9]X.,jumpdial+101,Dial(${TRUNK}/${EXTEN},,Wr)

exten => _0[1-9]X.,n+101,Busy

 

;Catch any (00xx) numbers

exten => _00X.,1,NoOp(Dialling International number)

exten => _00X.,n,Macro(setcli)

exten => _00X.,n(jumpdial),Dial(SIP/+${EXTEN:[EMAIL PROTECTED])

exten => _00X.,jumpdial+101,Dial(${TRUNK}/${EXTEN},,Wr)

exten => _00X.,n+101,Busy

 

 

I've tried putting the Catch codes above into a sub-context, and
then put an include into the [local], but it still dials via the Catch
international...

The odd thing is that in either, the show dialplan seems to
suggest the correct order :

 

 

 

  '_00165011091[45]0-9]' => 1. NoOp(I AM HERE)
[pbx_config]

2. Macro(setcli)
[pbx_config]

3. Dial(SIP/${EXTEN:[EMAIL PROTECTED])
[pbx_config]

4. Hangup()
[pbx_config]

 (some others)

  '_00X.' =>1. NoOp(Dialling International number)
[pbx_config]

2. Macro(setcli)
[pbx_config]

 [jumpdial] 3. Dial(SIP/+${EXTEN:[EMAIL PROTECTED])
[pbx_config]

104. Dial(${TRUNK}/${EXTEN}||Wr)
[pbx_config]

206. Busy()
[pbx_config]

 

 

The page at voip-info isn't too clear in the differences between
1.2 and 1.4
(http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf+sort
ing) so I'm not sure where I've gone wrong.

 

 

Adrian Marsh

 

 

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Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 18

[dimitri . osler]

Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a 
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al 
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], 
altrimenti vi risponderò al mio rientro.

Dimitri Osler

I will be on vacation until Tuesday 19th of August with limited access to voice 
and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 
0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on 
my return.

Dimitri Osler





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[asterisk-users] FWD $30 membership-fee

[Joseph]

I just received an email notice from FWD about $30 membership fee.
My question: Is the email genuine? Did anybody else receive it?

I'm just trying to be sure that it is real and not a scam.
The (FWD) does not do anything to authenticate such emails (implementing 
GPG/PGP signature etc.)

If the email is genuine, I hope they will improve their service; as of now 
their IAX server is not working.

-- 
#Joseph
GPG KeyID: ED0E1FB7

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Re: [asterisk-users] Improving the speed of chan_sip

[Steve Murphy]

On Thu, 2008-08-07 at 18:35 +0200, Pavel Jezek wrote:
> 
> Steve Murphy wrote:
> > Hello--
> >
> > Why do I target chan_sip for so much effort?  Because, 
> > it seems to me, chan_sip is probably the most used channel 
> > driver in the asterisk community!! (and, of course,
> > the zap/dahdi driver, is also pretty popular)
> >
> > I haven't had time to follow up on chan_sip, and I probably 
> > won't for several months. 
> >
> > But, if I had time, here is what I'd do:
> >
> > There are two ways to speed up chan_sip, and they are separate issues,
> > tied together on how many cpu cycles they use up:
> >
> > 1. Call setup/teardown (invites/hangups) -- limits the calls/sec
> > asterisk can handle.
> >   
> 
> one of the big issues in sip callsetup performance, that appears to me 
> in current trunk, is about 500ms delay in propagation SIP/OK message 
> between bridged parties
> eg.: one party answers call, send SIP/OK with SDP to asterisk, asterisk 
> then forwards it to other party, but with unacceptable delay about 500ms!
> this is so much, that users complaining about lost first word of speech 
> communication,
> I posted info about this to bugreport, that seems to be related to this, 
> look at my message:
> http://bugs.digium.com/view.php?id=12708#91173
> I also attach graph picture from wireshark, that clearly ilustrated, 
> where is problem (OK-SDP-delay.png )
> PJ

Pavel--

Recently, we found the slowdown that made trunk a LOT slower
on cps ratings than 1.4... And it did indeed point to the INVITE
handling sequence. It involved stripping out a call gethostbyname
in certain circumstances (the one frequently involved when you
use sipp). Putting it back in made trunk and 1.4 run at about the same
speed (the new, better method is a little slower at the moment).

The slowdown involved a call to find_channel_locked code, which ended
up in a fight for a channel lock, in the which it loops and absorbs
a good amount of cpu cycles. This micro-storm of activity was perfectly
timed with the creation of threads in the driver to put a huge dent
in the cps rating. It could easily be, that this sort of activity
is what was causing your bug.

(Now, getting trunk back up to speed with 1.4 is nice, but still,
chan_sip could be faster. I hope, much faster.)

You might try this again, with the current trunk, and see if you still 
suffer the same  delays. If your problem persists, then the bug 
still stands and we'll look at it, I'm sure.

Just updated bug 12708...

murf

-- 
Steve Murphy
Software Developer
Digium


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[asterisk-users] BRI AND DATA connection

[Anton VG]

Hello!

Does anyone tried BRI with asterisk for DATA transfer? My customer
wants BRI connection, but he wants it for the data, and I have to
bring connection to his office, so I see the connection as follows:

E1<->(CORE_ASTERISK)<->(IAX2)<->(EDGE_ASTERISK)<->BRI - so will data
work in such scenario? If no what scenario would be suggested?

Regards!

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Re: [asterisk-users] problem controlling dialplan order

[Felippe Silvestre]

Try this:
[local]

exten => _00165011091[45][0-9],1,NoOp(I AM HERE)

exten => _00165011091[45][0-9],n,Macro(setcli)

exten => _00165011091[45][0-9],n,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

exten => _00165011091[45][0-9],n,Hangup

 

The "[" before "0-9]" is needed.

 

 

 
Felippe Silvestre




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adrian
Marsh
Sent: Thursday, August 07, 2008 07:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] problem controlling dialplan order



Hi All,

 

On a 1.4.15 system, I've a context as below, where I need to
catch some specific US ranges and dial direct via SIP rather than a PSTN
trunk.  But the logic always goes via the International Trunk and I cant
see why...

 

[local]

exten => _00165011091[45]0-9],1,NoOp(I AM HERE)

exten => _00165011091[45]0-9],n,Macro(setcli)

exten =>
_00165011091[45]0-9],n,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

exten => _00165011091[45]0-9],n,Hangup

 

. (same context)

 

Catch local (UK) numbers

exten => _0[1-9]X.,1,NoOp(Dialling UK number)

exten => _0[1-9]X.,n,Macro(setcli)

exten => _0[1-9]X.,n(jumpdial),Dial(SIP/+44${EXTEN:[EMAIL PROTECTED])

exten => _0[1-9]X.,jumpdial+101,Dial(${TRUNK}/${EXTEN},,Wr)

exten => _0[1-9]X.,n+101,Busy

 

;Catch any (00xx) numbers

exten => _00X.,1,NoOp(Dialling International number)

exten => _00X.,n,Macro(setcli)

exten => _00X.,n(jumpdial),Dial(SIP/+${EXTEN:[EMAIL PROTECTED])

exten => _00X.,jumpdial+101,Dial(${TRUNK}/${EXTEN},,Wr)

exten => _00X.,n+101,Busy

 

 

I've tried putting the Catch codes above into a sub-context, and
then put an include into the [local], but it still dials via the Catch
international...

The odd thing is that in either, the show dialplan seems to
suggest the correct order :

 

 

 

  '_00165011091[45]0-9]' => 1. NoOp(I AM HERE)
[pbx_config]

2. Macro(setcli)
[pbx_config]

3. Dial(SIP/${EXTEN:[EMAIL PROTECTED])
[pbx_config]

4. Hangup()
[pbx_config]

 (some others)

  '_00X.' =>1. NoOp(Dialling International number)
[pbx_config]

2. Macro(setcli)
[pbx_config]

 [jumpdial] 3. Dial(SIP/+${EXTEN:[EMAIL PROTECTED])
[pbx_config]

104. Dial(${TRUNK}/${EXTEN}||Wr)
[pbx_config]

206. Busy()
[pbx_config]

 

 

The page at voip-info isn't too clear in the differences between
1.2 and 1.4
(http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf+sort
ing) so I'm not sure where I've gone wrong.

 

 

Adrian Marsh

 

 

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Re: [asterisk-users] Improving the speed of chan_sip

[Pavel Jezek]

Steve Murphy wrote:
> Hello--
>
> Why do I target chan_sip for so much effort?  Because, 
> it seems to me, chan_sip is probably the most used channel 
> driver in the asterisk community!! (and, of course,
> the zap/dahdi driver, is also pretty popular)
>
> I haven't had time to follow up on chan_sip, and I probably 
> won't for several months. 
>
> But, if I had time, here is what I'd do:
>
> There are two ways to speed up chan_sip, and they are separate issues,
> tied together on how many cpu cycles they use up:
>
> 1. Call setup/teardown (invites/hangups) -- limits the calls/sec
> asterisk can handle.
>   

one of the big issues in sip callsetup performance, that appears to me 
in current trunk, is about 500ms delay in propagation SIP/OK message 
between bridged parties
eg.: one party answers call, send SIP/OK with SDP to asterisk, asterisk 
then forwards it to other party, but with unacceptable delay about 500ms!
this is so much, that users complaining about lost first word of speech 
communication,
I posted info about this to bugreport, that seems to be related to this, 
look at my message:
http://bugs.digium.com/view.php?id=12708#91173
I also attach graph picture from wireshark, that clearly ilustrated, 
where is problem (OK-SDP-delay.png )
PJ






> 2. Sound processing: Moving around the sound data (frames) between
> channels,
> -- limits the number of simultaneous conversations asterisk can handle.
>
> For Call setup/teardown speed improvements:
>
> a. I would profile how much time is spent by the handlers in the 
>various situations (handling invites, etc).
>Theory: the cps (calls/sec) rate is currently limited by 
>the time it takes to process INVITE requests. You
>measure the time spent in handling INVITEs and
>find the average number of microseconds spent,
>and my suspicion is that the inverse of this
>time (in sec) would be the number of calls/sec
>chan_sip can handle. If this is the case, then
>find where the invite is spending all its time.
>Theory: The majority of chan_sip's INVITE
>processing time is spent in creating the
>thread for running the PBX.
>
>  If both theories above prove true, then to increase the
>  cps rating of asterisk, you institute a fairly large
>  thread pool (like what chan_iax does). Chan_iax uses
>  its thread pool to handle network request processing;
>  chan_sip can do this also, or just use its thread pool
>  for pbx threads. Going the pbx_start route might
>  be tactically better-- it could be used to speed
>  up EVERY channel driver, instead of just chan_sip.
>  Either way, thread pools would reduce the invite
>  time substantially, and allow a higher cps rating.
>
>  Also, if the above theories both prove true, then
>  I'd copy the thread pool stuff in chan_iax,
>  and make it pretty generic, and use
>  it as a basis for forming a thread pool for
>  running the pbx, then retrofit the chan_iax
>  code to also use the generic pool for 
>  network request handling...
>
>  If either or both of the above theories prove 
>  false, then the only path left is profile 
>  asterisk running near saturation, and optimize
>  the routines that are hogging the most cpu
>  cycles.
>
> To Enhance the Number of Simultaneous Conversations:
>
> b. Carefully profile asterisk while near saturation,
>find the chief cpu cycle absorbers, and optimize them.
>Theory: Optimizing maybe the top 5 cycle-burning routines
>could yield a noticeable improvement in how many
>simultaneous conversations asterisk could handle.
>Of course, for just numbers, you use the fastest
>codecs. If the codecs end up being the limiting
>factor, (and they may just be), optimizing those
>might be very rewarding, also, but then again,
>they are pretty optimized already (I hope!).
>
>When it comes to optimization, there are often
>surprising cases where improvements can be made!
>
> If some brave soul is interested in helping with this,
> feel free to dive in.
>
> murf
>
>   
> 
>
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Re: [asterisk-users] intercom/paging with grandstream gxp2000

[Fidel Garcia]

I added the configuration as you suggest but now the phone does not do
intercom. I tried Dial and Page in the gxp2000 but everything goes out as
Dial.

Here is the extensions.conf now
exten=s,1,SIPAddHeader(Alert-Info: \;info=Family)
exten=s,2,GotoIf($["${SIP_HEADER(Call-Info)}"="answer-after=0"]?3:4)
exten=s,3,SIPAddHeader(Call-Info: answer-after=0)
exten=s,4,Dial(${ARG2},20)
exten=s,5,Goto(s-${DIALSTATUS},1)
exten=s-NOANSWER,1,Voicemail(${ARG1},u)
exten=s-NOANSWER,2,Goto(default,s,1)
exten=s-BUSY,1,Voicemail(${ARG1},b)
exten=s-BUSY,2,Goto(default,s,1)
exten=_s-.,1,Goto(s-NOANSWER,1)
exten=a,1,VoicemailMain(${ARG1})

Any idea? I am very bad on this asterisk thing, sorry guys.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel van
Baak
Sent: Thursday, August 07, 2008 12:10 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] intercom/paging with grandstream gxp2000

On 10:59, Thu 07 Aug 08, Fidel Garcia wrote:
> Thanks for your reply!
> 
> Just so you have a better understanding of what I am trying to accomplish.
> The distinctive ring is working fine with "Family", however, the intercom
> configuration that I am currently testing makes all my calls and intercom
> call. It does not matter if I call using Dial or Page on the GXP2000, the
> call is always and intercom call. For some reason the GXP2000 is receiving
> the SipAddHeader when I do Dial and Page. Can you tell what is wrong with
> the configuration by looking at the configuration below?
> 
> exten=s,1,SIPAddHeader(Alert-Info: \;info=Family)
> exten=s,2,GotoIf($["${SIP_HEADER(Call-Info)}"="answer-after=0"]?2:3)
> exten=s,3,SIPAddHeader(Call-Info: answer-after=0)

if the sip header Call-Info has value answer-after=0 it goes to prio 2,
otherwise 3

Now let's have a closer look at those.
Hhmm, prio two is the gotoif, prio three adds the answer-after=0 ...

I think you mean:

exten=s,2,GotoIf($["${SIP_HEADER(Call-Info)}"="answer-after=0"]?3:4)

> exten=s,4,Dial(${ARG2},20)
> exten=s,5,Goto(s-${DIALSTATUS},1)
> exten=s-NOANSWER,1,Voicemail(${ARG1},u)
> exten=s-NOANSWER,2,Goto(default,s,1)
> exten=s-BUSY,1,Voicemail(${ARG1},b)
> exten=s-BUSY,2,Goto(default,s,1)
> exten=_s-.,1,Goto(s-NOANSWER,1)
> exten=a,1,VoicemailMain(${ARG1})
> 
> what would you do differently?
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Gordon
> Henderson
> Sent: Thursday, August 07, 2008 7:32 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] intercom/paging with grandstream gxp2000
> 
> On Wed, 6 Aug 2008, Fidel Garcia wrote:
> 
> > Guys I have been reading for days on how to get this to work with
asterisk
> > and for some reason every time I call the call goes to intercom.  I know
I
> > must be doing something wrong with the way I am adding the steps to my
> call;
> > I am not familiar with variables and flags.
> 
> What *exactly* are you trying to achieve?
> 
> I have used both paging and intercom mode in the Grandstreams with good 
> results.
> 
> You do need the settings in the phone set ON - ie.
> 
>   Allow Auto Answer by Call-Info:   No  Yes
>   Turn off speaker on remote disconnect:   No  Yes
> 
> These both need to be set to YES or ON.
> 
> That won't affect normal calls to that account on the phone - although the

> "turn off speaker" one does make the phone easier to use IMO...
> 
> So call the phone and the person answers normally, as before, but if you 
> rhen add the SIP header:
> 
>   SIPAddHeader(Call-Info: answer-after=0)
> 
> The phone will auto-answer - when the next Dial or Page command is 
> directed to it.
> 
> What next? If you want to Page the phone, use the Page() application.
> 
> So if the phone is SIP/100 then to Dial the phone normally..
> 
>  exten => 100,1,Dial(SIP/100)
> 
> but to page it:
> 
>  exten => 200,1,SIPAddHeader(Call-Info: answer-after=0)
>  exten => 200,n,Page(SIP/100)
> 
> and to intercom to it:
> 
>  exten => 300,1,SIPAddHeader(Call-Info: answer-after=0)
>  exten => 300,n,Page(SIP/100,d)
> 
> 
> So this has added 3 new extensions, 100, 200 and 300 - which all 'call' 
> SIP/100, but in 3 differet ways.
> 
> Gordon
> 
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> No virus found in this incoming message.
> Checked by AVG - http://www.avg.com 
> Version: 8.0.138 / Virus Database: 270.5.12/1596 - Release Date: 8/6/2008
> 4:55 PM
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Re: [asterisk-users] intercom/paging with grandstream gxp2000

[Michiel van Baak]

On 10:59, Thu 07 Aug 08, Fidel Garcia wrote:
> Thanks for your reply!
> 
> Just so you have a better understanding of what I am trying to accomplish.
> The distinctive ring is working fine with "Family", however, the intercom
> configuration that I am currently testing makes all my calls and intercom
> call. It does not matter if I call using Dial or Page on the GXP2000, the
> call is always and intercom call. For some reason the GXP2000 is receiving
> the SipAddHeader when I do Dial and Page. Can you tell what is wrong with
> the configuration by looking at the configuration below?
> 
> exten=s,1,SIPAddHeader(Alert-Info: \;info=Family)
> exten=s,2,GotoIf($["${SIP_HEADER(Call-Info)}"="answer-after=0"]?2:3)
> exten=s,3,SIPAddHeader(Call-Info: answer-after=0)

if the sip header Call-Info has value answer-after=0 it goes to prio 2,
otherwise 3

Now let's have a closer look at those.
Hhmm, prio two is the gotoif, prio three adds the answer-after=0 ...

I think you mean:

exten=s,2,GotoIf($["${SIP_HEADER(Call-Info)}"="answer-after=0"]?3:4)

> exten=s,4,Dial(${ARG2},20)
> exten=s,5,Goto(s-${DIALSTATUS},1)
> exten=s-NOANSWER,1,Voicemail(${ARG1},u)
> exten=s-NOANSWER,2,Goto(default,s,1)
> exten=s-BUSY,1,Voicemail(${ARG1},b)
> exten=s-BUSY,2,Goto(default,s,1)
> exten=_s-.,1,Goto(s-NOANSWER,1)
> exten=a,1,VoicemailMain(${ARG1})
> 
> what would you do differently?
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Gordon
> Henderson
> Sent: Thursday, August 07, 2008 7:32 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] intercom/paging with grandstream gxp2000
> 
> On Wed, 6 Aug 2008, Fidel Garcia wrote:
> 
> > Guys I have been reading for days on how to get this to work with asterisk
> > and for some reason every time I call the call goes to intercom.  I know I
> > must be doing something wrong with the way I am adding the steps to my
> call;
> > I am not familiar with variables and flags.
> 
> What *exactly* are you trying to achieve?
> 
> I have used both paging and intercom mode in the Grandstreams with good 
> results.
> 
> You do need the settings in the phone set ON - ie.
> 
>   Allow Auto Answer by Call-Info:   No  Yes
>   Turn off speaker on remote disconnect:   No  Yes
> 
> These both need to be set to YES or ON.
> 
> That won't affect normal calls to that account on the phone - although the 
> "turn off speaker" one does make the phone easier to use IMO...
> 
> So call the phone and the person answers normally, as before, but if you 
> rhen add the SIP header:
> 
>   SIPAddHeader(Call-Info: answer-after=0)
> 
> The phone will auto-answer - when the next Dial or Page command is 
> directed to it.
> 
> What next? If you want to Page the phone, use the Page() application.
> 
> So if the phone is SIP/100 then to Dial the phone normally..
> 
>  exten => 100,1,Dial(SIP/100)
> 
> but to page it:
> 
>  exten => 200,1,SIPAddHeader(Call-Info: answer-after=0)
>  exten => 200,n,Page(SIP/100)
> 
> and to intercom to it:
> 
>  exten => 300,1,SIPAddHeader(Call-Info: answer-after=0)
>  exten => 300,n,Page(SIP/100,d)
> 
> 
> So this has added 3 new extensions, 100, 200 and 300 - which all 'call' 
> SIP/100, but in 3 differet ways.
> 
> Gordon
> 
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> 4:55 PM
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GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer aficionados are both called users?"


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Re: [asterisk-users] intercom/paging with grandstream gxp2000

[Gordon Henderson]

On Thu, 7 Aug 2008, Fidel Garcia wrote:

> Thanks for your reply!
>
> Just so you have a better understanding of what I am trying to accomplish.
> The distinctive ring is working fine with "Family", however, the intercom
> configuration that I am currently testing makes all my calls and intercom
> call. It does not matter if I call using Dial or Page on the GXP2000, the
> call is always and intercom call. For some reason the GXP2000 is receiving
> the SipAddHeader when I do Dial and Page. Can you tell what is wrong with
> the configuration by looking at the configuration below?
>
> exten=s,1,SIPAddHeader(Alert-Info: \;info=Family)
> exten=s,2,GotoIf($["${SIP_HEADER(Call-Info)}"="answer-after=0"]?2:3)
> exten=s,3,SIPAddHeader(Call-Info: answer-after=0)
> exten=s,4,Dial(${ARG2},20)
> exten=s,5,Goto(s-${DIALSTATUS},1)
> exten=s-NOANSWER,1,Voicemail(${ARG1},u)
> exten=s-NOANSWER,2,Goto(default,s,1)
> exten=s-BUSY,1,Voicemail(${ARG1},b)
> exten=s-BUSY,2,Goto(default,s,1)
> exten=_s-.,1,Goto(s-NOANSWER,1)
> exten=a,1,VoicemailMain(${ARG1})
>
> what would you do differently?

Well, I'd stop using numbers and start using labels to begin with.

Line 2:

I have no idea what this is doing, but it looks like You're saying that if 
a SIP_HEADER called Call-Info is already set to answer-after=0 then jump 
to ... step 2. This would cause an infinite loop.

then in the next step you explicitly set the Call-Info to answer-after=0, 
so what do you expect?

Your dialplan is just broken.

Create TWO extensions for this phone. One to make the phone ring as a 
normal phone and the other to make the phone go into Intercom mode. You 
can not select modes like this.

I'm assuming that's a macro, if-so, them make it:

[Macro-ringPhone]
exten => s,1,Dial(${ARG2},20)
exten => s,n,Goto(${DIALSTATUS})

exten => s,n(NOANSWER),Voicemail(${ARG1},u)
...and so on...

Then to call it:

exten => 100,1,Macro(ringPhone,100,100)

or to call it in intercom mode.

exten => 200,1,SIPAddHeader(Call-Info: answer-after=0)
exten => 200,n,Macro(ringPhone,100,100)

Gordon


>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Gordon
> Henderson
> Sent: Thursday, August 07, 2008 7:32 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] intercom/paging with grandstream gxp2000
>
> On Wed, 6 Aug 2008, Fidel Garcia wrote:
>
>> Guys I have been reading for days on how to get this to work with asterisk
>> and for some reason every time I call the call goes to intercom.  I know I
>> must be doing something wrong with the way I am adding the steps to my
> call;
>> I am not familiar with variables and flags.
>
> What *exactly* are you trying to achieve?
>
> I have used both paging and intercom mode in the Grandstreams with good
> results.
>
> You do need the settings in the phone set ON - ie.
>
>   Allow Auto Answer by Call-Info:   No  Yes
>   Turn off speaker on remote disconnect:   No  Yes
>
> These both need to be set to YES or ON.
>
> That won't affect normal calls to that account on the phone - although the
> "turn off speaker" one does make the phone easier to use IMO...
>
> So call the phone and the person answers normally, as before, but if you
> rhen add the SIP header:
>
>   SIPAddHeader(Call-Info: answer-after=0)
>
> The phone will auto-answer - when the next Dial or Page command is
> directed to it.
>
> What next? If you want to Page the phone, use the Page() application.
>
> So if the phone is SIP/100 then to Dial the phone normally..
>
> exten => 100,1,Dial(SIP/100)
>
> but to page it:
>
> exten => 200,1,SIPAddHeader(Call-Info: answer-after=0)
> exten => 200,n,Page(SIP/100)
>
> and to intercom to it:
>
> exten => 300,1,SIPAddHeader(Call-Info: answer-after=0)
> exten => 300,n,Page(SIP/100,d)
>
>
> So this has added 3 new extensions, 100, 200 and 300 - which all 'call'
> SIP/100, but in 3 differet ways.
>
> Gordon
>
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Re: [asterisk-users] Cisco 7970, CTLSEP.tlv

[Jason Parker]

Jason Parker wrote:
> I just wanted to post this so that it was out there and Googleable.  Hopefully
> it will save other people a bit of time.
> 
> If you have a Cisco phone (I was testing with a 7970, though presumably it 
> would
> affect 7960 and others as well) that is looping trying to fetch the CTL tlv 
> file
> - it may be because you are using Debians 'tftpd' (should be
> netkit-tftpd...*cough*hey, Debian developers*cough*) package, which is
> apparently not RFC 783 (tftp) compliant with "file not found" responses.  The
> whopping 18 page RFC states that Error Code should be 0x00,0x01 for "file not
> found" errors, but netkit-tftpd returns 0x00,0x00 which is "Not defined" -
> causing the phone to ignore it and request the file again a few seconds later.
> 
> Solution: Switch to any other tftpd.  The moment I switched to tftpd-hpa or
> atftpd, the phone stopped looping, picked up the SEP.cnf.xml file, and
> immediately registered to Asterisk.
> 
> Hopefully in the future Debian will rename, remove, or fix this package so it 
> is
> no longer the default tftpd.
> 

Responding to myself...

When I initially sent this, I had made several false assumptions.  The biggest
of which, was that the 'tftpd' package in Debian was no longer maintained
(upstream hadn't made a release in 8 years, and Debian hadn't made a release in
3 years - I think it was a fairly reasonable one).

Well, the maintainer of this package, Alberto, emailed me to let me know that
somebody pointed him to this post, and that less than 24 hours later, he had
fixed this bug (I've confirmed this) and made a new release - 0.17-16 - which is
currently in Sid, and will hopefully be put into Lenny.  This can be downloaded
from http://packages.debian.org/search?keywords=tftpd


Also, as Alberto correctly pointed out - I violated one of the most important
rules of Open Source Software.  If I may quote him: "You had perfectly traced
the problem, you perfectly described it, god! you even gave a reference to the
RFC.  You had the perfect bug report, but it was never going to make it to
me arrrggg  :)  Such a great loss!!"  I failed to complete one critical step
- reporting a bug.  It ended up working out, but only because somebody else took
the time to report the bug.

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Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 17

[dimitri . osler]

Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a 
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al 
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], 
altrimenti vi risponderò al mio rientro.

Dimitri Osler

I will be on vacation until Tuesday 19th of August with limited access to voice 
and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 
0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on 
my return.

Dimitri Osler





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[asterisk-users] Improving the speed of chan_sip

[Steve Murphy]

Hello--

Why do I target chan_sip for so much effort?  Because, 
it seems to me, chan_sip is probably the most used channel 
driver in the asterisk community!! (and, of course,
the zap/dahdi driver, is also pretty popular)

I haven't had time to follow up on chan_sip, and I probably 
won't for several months. 

But, if I had time, here is what I'd do:

There are two ways to speed up chan_sip, and they are separate issues,
tied together on how many cpu cycles they use up:

1. Call setup/teardown (invites/hangups) -- limits the calls/sec
asterisk can handle.

2. Sound processing: Moving around the sound data (frames) between
channels,
-- limits the number of simultaneous conversations asterisk can handle.

For Call setup/teardown speed improvements:

a. I would profile how much time is spent by the handlers in the 
   various situations (handling invites, etc).
   Theory: the cps (calls/sec) rate is currently limited by 
   the time it takes to process INVITE requests. You
   measure the time spent in handling INVITEs and
   find the average number of microseconds spent,
   and my suspicion is that the inverse of this
   time (in sec) would be the number of calls/sec
   chan_sip can handle. If this is the case, then
   find where the invite is spending all its time.
   Theory: The majority of chan_sip's INVITE
   processing time is spent in creating the
   thread for running the PBX.

 If both theories above prove true, then to increase the
 cps rating of asterisk, you institute a fairly large
 thread pool (like what chan_iax does). Chan_iax uses
 its thread pool to handle network request processing;
 chan_sip can do this also, or just use its thread pool
 for pbx threads. Going the pbx_start route might
 be tactically better-- it could be used to speed
 up EVERY channel driver, instead of just chan_sip.
 Either way, thread pools would reduce the invite
 time substantially, and allow a higher cps rating.

 Also, if the above theories both prove true, then
 I'd copy the thread pool stuff in chan_iax,
 and make it pretty generic, and use
 it as a basis for forming a thread pool for
 running the pbx, then retrofit the chan_iax
 code to also use the generic pool for 
 network request handling...

 If either or both of the above theories prove 
 false, then the only path left is profile 
 asterisk running near saturation, and optimize
 the routines that are hogging the most cpu
 cycles.

To Enhance the Number of Simultaneous Conversations:

b. Carefully profile asterisk while near saturation,
   find the chief cpu cycle absorbers, and optimize them.
   Theory: Optimizing maybe the top 5 cycle-burning routines
   could yield a noticeable improvement in how many
   simultaneous conversations asterisk could handle.
   Of course, for just numbers, you use the fastest
   codecs. If the codecs end up being the limiting
   factor, (and they may just be), optimizing those
   might be very rewarding, also, but then again,
   they are pretty optimized already (I hope!).

   When it comes to optimization, there are often
   surprising cases where improvements can be made!

If some brave soul is interested in helping with this,
feel free to dive in.

murf

-- 
Steve Murphy
Software Developer
Digium


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Re: [asterisk-users] CLI> show queues NOT WORKING WELL

[Daniel - Asterisk]

This problem was fixed when I upgraded my box to version 1.4.21.1

Thanks everyone,

Daniel

On Mon, Jun 30, 2008 at 2:01 PM, Chento Arohuanca <[EMAIL PROTECTED]>wrote:

> I forgot it!, I'm using Asterisk 1.4.19.1 version.
>
>
> On Mon, Jun 30, 2008 at 1:47 PM, Chento Arohuanca <[EMAIL PROTECTED]>
> wrote:
>
>> Hi Atis and friends,
>>
>> I think it has direct relation with calls not being delivered to available
>> agents, as you can see at following lines, I´ve confirmed i*n situ* there
>> were real available agents:
>>
>> [EMAIL PROTECTED] apps]# asterisk -rx "queue show"
>> QUEUE1  has 0 calls (max 100) in 'rrmemory' strategy (0s holdtime),
>> W:1, C:722, A:13, SL:83.7% within 0s
>>Members:
>>   Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has
>> taken no calls yet
>>   Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (In use) has
>> taken no calls yet
>>   Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has
>> taken no calls yet
>>   Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (In use) has
>> taken 1 calls (last was 2040 secs ago)
>>   Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (In use) has
>> taken no calls yet
>>   Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has
>> taken no calls yet
>>   Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (In use) has
>> taken no calls yet
>>   Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has
>> taken no calls yet
>>   Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has
>> taken no calls yet
>>   Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (In use) has
>> taken 3 calls (last was 1045 secs ago)
>>   Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has
>> taken no calls yet
>>   Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has
>> taken no calls yet
>>   Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (In use) has
>> taken 3 calls (last was 1207 secs ago)
>>   Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (In use) has
>> taken 8 calls (last was 351 secs ago)
>>   Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (In use) has
>> taken 3 calls (last was 450 secs ago)
>>No Callers
>>
>> QUEUE2  has 0 calls (max 100) in 'rrmemory' strategy (0s holdtime), W:4,
>> C:249, A:1, SL:87.6% within 0s
>>Members:
>> *[agents'  status are similar to QUEUE5]*
>>No Callers
>>
>> QUEUE3  has 0 calls (max 100) in 'rrmemory' strategy (3s holdtime), W:3,
>> C:582, A:6, SL:84.7% within 0s
>>Members:
>> *[agents'  status are similar to QUEUE5]*
>>No Callers
>>
>> QUEUE4 has 0 calls (max 100) in 'rrmemory' strategy (0s holdtime), W:3,
>> C:254, A:9, SL:78.3% within 0s
>>Members:
>> *[agents'  status are similar to QUEUE5]*
>>No Callers
>>
>>
>> QUEUE5 has 2 calls (max 100) in 'rrmemory' strategy (7s holdtime),
>> W:1, C:531, A:6, SL:79.3% within 0s
>>Members:
>> *  Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use)
>> has taken no calls yet*
>>Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (In use) has
>> taken 1 calls (last was 1253 secs ago)
>> *  Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use)
>> has taken 9 calls (last was 795 secs ago*
>>   Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (In use) has
>> taken 4 calls (last was 251 secs ago)
>>   Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (In use) has
>> taken 2 calls (last was 283 secs ago)
>> *  Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use)
>> has taken 5 calls (last was 264 secs ago*
>>   Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (In use) has
>> taken 1 calls (last was 15 secs ago)
>> *  Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use)
>> has taken no calls yet
>>   Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has
>> taken no calls yet
>> *
>>   Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (In use) has
>> taken 1 calls (last was 3376 secs ago)
>> *  Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use)
>> has taken no calls yet
>>   Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has
>> taken no calls yet
>> *
>>   Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (In use) has
>> taken no calls yet
>>   Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (In use) has
>> taken 3 calls (last was 1162 secs ago)
>>   Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (In use) has
>> taken 2 calls (last was 2046 secs ago)
>>Callers:
>> *  1. Zap/76-1 (wait: 0:28, prio: 0)
>>   2. Zap/63-1 (wait: 0:04, prio: 0)*
>>
>> QUEUE6 has 0 calls (max 100) in 'rrmemory' strategy (0s holdtime), W:4,
>> C:360, A:0, SL:84.2% within 0s
>>Members:
>> *[agents'  status are similar to QUEUE5]*
>>No Callers
>>
>> QUEUE7 has 0 calls (max 100) in 'rrmemory' strategy (0s holdtime), W:3,
>> C

Re: [asterisk-users] outgoing call file and agi detect busy

[Jerry Geis]

>
> Call files spawn a completely new channel that your AGI most likely 
> isn't going to be able to track.  Since the call is a completely new 
> channel, the DIALSTATUS variable for this channel will not be visible to 
> your original channel.  You may want to look at using the Originate 
> action from within the manager API.
>
> http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate
>
>   

So there is no way either in the dialplan or in the AGI that I can find 
out a status of my call?
As to why it did or did not complete?
Doesnt that seem like a defect?
There has to be a way around that?

I dont see off hand how this manager originate is any different.

Jerry

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Re: [asterisk-users] intercom/paging with grandstream gxp2000

[Fidel Garcia]

Thanks for your reply!

Just so you have a better understanding of what I am trying to accomplish.
The distinctive ring is working fine with "Family", however, the intercom
configuration that I am currently testing makes all my calls and intercom
call. It does not matter if I call using Dial or Page on the GXP2000, the
call is always and intercom call. For some reason the GXP2000 is receiving
the SipAddHeader when I do Dial and Page. Can you tell what is wrong with
the configuration by looking at the configuration below?

exten=s,1,SIPAddHeader(Alert-Info: \;info=Family)
exten=s,2,GotoIf($["${SIP_HEADER(Call-Info)}"="answer-after=0"]?2:3)
exten=s,3,SIPAddHeader(Call-Info: answer-after=0)
exten=s,4,Dial(${ARG2},20)
exten=s,5,Goto(s-${DIALSTATUS},1)
exten=s-NOANSWER,1,Voicemail(${ARG1},u)
exten=s-NOANSWER,2,Goto(default,s,1)
exten=s-BUSY,1,Voicemail(${ARG1},b)
exten=s-BUSY,2,Goto(default,s,1)
exten=_s-.,1,Goto(s-NOANSWER,1)
exten=a,1,VoicemailMain(${ARG1})

what would you do differently?



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gordon
Henderson
Sent: Thursday, August 07, 2008 7:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] intercom/paging with grandstream gxp2000

On Wed, 6 Aug 2008, Fidel Garcia wrote:

> Guys I have been reading for days on how to get this to work with asterisk
> and for some reason every time I call the call goes to intercom.  I know I
> must be doing something wrong with the way I am adding the steps to my
call;
> I am not familiar with variables and flags.

What *exactly* are you trying to achieve?

I have used both paging and intercom mode in the Grandstreams with good 
results.

You do need the settings in the phone set ON - ie.

Allow Auto Answer by Call-Info:   No  Yes
Turn off speaker on remote disconnect:   No  Yes

These both need to be set to YES or ON.

That won't affect normal calls to that account on the phone - although the 
"turn off speaker" one does make the phone easier to use IMO...

So call the phone and the person answers normally, as before, but if you 
rhen add the SIP header:

SIPAddHeader(Call-Info: answer-after=0)

The phone will auto-answer - when the next Dial or Page command is 
directed to it.

What next? If you want to Page the phone, use the Page() application.

So if the phone is SIP/100 then to Dial the phone normally..

 exten => 100,1,Dial(SIP/100)

but to page it:

 exten => 200,1,SIPAddHeader(Call-Info: answer-after=0)
 exten => 200,n,Page(SIP/100)

and to intercom to it:

 exten => 300,1,SIPAddHeader(Call-Info: answer-after=0)
 exten => 300,n,Page(SIP/100,d)


So this has added 3 new extensions, 100, 200 and 300 - which all 'call' 
SIP/100, but in 3 differet ways.

Gordon

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No virus found in this incoming message.
Checked by AVG - http://www.avg.com 
Version: 8.0.138 / Virus Database: 270.5.12/1596 - Release Date: 8/6/2008
4:55 PM


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Re: [asterisk-users] outgoing call file and agi detect busy

[Rob Hillis]

Jerry Geis wrote:
> Jerry Geis wrote:
>   
>> I am using asterisk 1.4.21 with outgoing call files.
>>
>> I am call a line that is busy as you can see below.
>> How can my AGI ask what the status of the last call was
>> so I can tell if there was NO ANSWER or it was BUSY?
>>
>> Thanks,
>>
>> Jerry
>>
>>-- Attempting call on SIP/401 for 
>> [EMAIL PROTECTED]:1 (Retry 1)
>>-- Got SIP response 486 "Busy" back from 192.168.1.161
>>   > Channel SIP/401-15aa5ab0 was never answered.
>>-- Executing [EMAIL PROTECTED]:1] AGI("OutgoingSpoolFailed", 
>> "smvoice") in new stack
>>-- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice
>>  == Spawn extension (smvoice-dialout, failed, 1) exited non-zero on 
>> 'OutgoingSpoolFailed'
>> [Aug  7 09:20:56] NOTICE[11272]: pbx_spool.c:341 attempt_thread: Call 
>> failed to go through, reason (5) Remote end is Busy
>>
>>
>> 
> I am trying to inquire ${DIALSTATUS} from both the agi and the dial plan 
> and it is not set.
>
> exten => failed,1,noop(${DIALSTATUS})
> exten => failed,2,agi(smvoice,-digium_failed)
>
>
>   -- Attempting call on SIP/401 for 
> [EMAIL PROTECTED]:1 (Retry 1)
> -- Got SIP response 486 "Busy" back from 192.168.1.161
>> Channel SIP/401-15ac2dd0 was never answered.
> -- Executing [EMAIL PROTECTED]:1] NoOp("OutgoingSpoolFailed", 
> "") in new stack
> -- Executing [EMAIL PROTECTED]:2] AGI("OutgoingSpoolFailed", 
> "smvoice|-digium_failed") in new stack
> -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice
>   == Spawn extension (smvoice-dialout, failed, 2) exited non-zero on 
> 'OutgoingSpoolFailed'
>
>
> As you can see above the call was BUSY, the dialplan should be showing 
> ${DIALSTATUS}
> and it is empty. What am I not doing to get the resulting dial status?
>   

Call files spawn a completely new channel that your AGI most likely 
isn't going to be able to track.  Since the call is a completely new 
channel, the DIALSTATUS variable for this channel will not be visible to 
your original channel.  You may want to look at using the Originate 
action from within the manager API.

http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate

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Re: [asterisk-users] Capture digits, set as variable..., use for caller id?

[Tilghman Lesher]

On Wednesday 06 August 2008 22:35:01 Positively Optimistic wrote:
> We've searched but thus far have not successfully found a solution for
> this…
>
> We're looking for a way to set a variable using get digits for a DISA
> application.   Sometimes we're away from the office and get a voicemail
> that I need to respond to quickly and would prefer for the caller to be
> presented with the caller id of the office, or perhaps home….
>
> I would like to set up DISA so that we can dial into the switch, enter a
> password, provide the outgoing caller ID that we want to present, enter the
> number I want to dial, and PRESTO..   make a call…   Any ideas?

I don't know why not.  It's not terribly difficult.  You may need to record
several custom prompts, but that's as easy as Record(mynewprompt.wav).
Something along the lines of:

VMAuthenticate()
Read(newcid,entercid,10)
DISA(no-password,outgoing,${newcid})

-- 
Tilghman

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Re: [asterisk-users] outgoing call file and agi detect busy

[Jerry Geis]

Jerry Geis wrote:
> I am using asterisk 1.4.21 with outgoing call files.
>
> I am call a line that is busy as you can see below.
> How can my AGI ask what the status of the last call was
> so I can tell if there was NO ANSWER or it was BUSY?
>
> Thanks,
>
> Jerry
>
>-- Attempting call on SIP/401 for 
> [EMAIL PROTECTED]:1 (Retry 1)
>-- Got SIP response 486 "Busy" back from 192.168.1.161
>   > Channel SIP/401-15aa5ab0 was never answered.
>-- Executing [EMAIL PROTECTED]:1] AGI("OutgoingSpoolFailed", 
> "smvoice") in new stack
>-- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice
>  == Spawn extension (smvoice-dialout, failed, 1) exited non-zero on 
> 'OutgoingSpoolFailed'
> [Aug  7 09:20:56] NOTICE[11272]: pbx_spool.c:341 attempt_thread: Call 
> failed to go through, reason (5) Remote end is Busy
>
>
I am trying to inquire ${DIALSTATUS} from both the agi and the dial plan 
and it is not set.

exten => failed,1,noop(${DIALSTATUS})
exten => failed,2,agi(smvoice,-digium_failed)


  -- Attempting call on SIP/401 for 
[EMAIL PROTECTED]:1 (Retry 1)
-- Got SIP response 486 "Busy" back from 192.168.1.161
   > Channel SIP/401-15ac2dd0 was never answered.
-- Executing [EMAIL PROTECTED]:1] NoOp("OutgoingSpoolFailed", 
"") in new stack
-- Executing [EMAIL PROTECTED]:2] AGI("OutgoingSpoolFailed", 
"smvoice|-digium_failed") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice
  == Spawn extension (smvoice-dialout, failed, 2) exited non-zero on 
'OutgoingSpoolFailed'


As you can see above the call was BUSY, the dialplan should be showing 
${DIALSTATUS}
and it is empty. What am I not doing to get the resulting dial status?

Thanks,

Jerry


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[asterisk-users] outgoing call file and agi detect busy

[Jerry Geis]

I am using asterisk 1.4.21 with outgoing call files.

I am call a line that is busy as you can see below.
How can my AGI ask what the status of the last call was
so I can tell if there was NO ANSWER or it was BUSY?

Thanks,

Jerry

-- Attempting call on SIP/401 for 
[EMAIL PROTECTED]:1 (Retry 1)
-- Got SIP response 486 "Busy" back from 192.168.1.161
   > Channel SIP/401-15aa5ab0 was never answered.
-- Executing [EMAIL PROTECTED]:1] AGI("OutgoingSpoolFailed", 
"smvoice") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice
  == Spawn extension (smvoice-dialout, failed, 1) exited non-zero on 
'OutgoingSpoolFailed'
[Aug  7 09:20:56] NOTICE[11272]: pbx_spool.c:341 attempt_thread: Call 
failed to go through, reason (5) Remote end is Busy


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[asterisk-users] Voicemail on PRI

[Yann Derichard]

Hi,

I am trying to install a Voicemail on PRI after a redirection on an away or
a busy (a normal call which is redirected to voicemail in fact) but I can't
find the function in Asterisk which allow me using the phone number of the
callee (because I have only the number of asterisk and of the caller).

Is someone could give me a clue ?

Regards

-- 
Yann Derichard
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[asterisk-users] I can´t hear the warning sound in Dial command

[equis software]

Hi!
I´m using cmd Dial from an EAGI script
My problem is that I cant hear the warning sound

EAGI script:

SET VARIABLE LIMIT_WARNING_FILE beep
SET VARIABLE LIMIT_PLAYAUDIO_CALLEE yes
SET VARIABLE LIMIT_PLAYAUDIO_CALLER yes

EXEC DIAL Zap/g1/676354|20|HL(132000:3000:3000)

CLI:
-- AGI Script Executing Application: (DIAL) Options:
(Zap/g1/676354|20|HL(132000:3000:3000))
-- Limit Data for this call:
   > timelimit  = 132000
   > play_warning   = 3000
   > play_to_caller = yes
   > play_to_callee = yes
   > warning_freq   = 3000
   > start_sound= (null)
   > warning_sound  = beep
   > end_sound  = (null)
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/676354
-- Zap/1-1 is ringing
-- Zap/1-1 answered SIP/510093-08238830
-- Channel 0/1, span 1 got hangup request, cause 16
-- Hungup 'Zap/1-1'
-- AGI Script Prepago completed, returning 0


What I´m doing wrong??
Thanks!!


Asterisk 1.4.19 built by root @ a1 on a i686 running Linux
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Re: [asterisk-users] intercom/paging with grandstream gxp2000

[Gordon Henderson]

On Wed, 6 Aug 2008, Fidel Garcia wrote:

> Guys I have been reading for days on how to get this to work with asterisk
> and for some reason every time I call the call goes to intercom.  I know I
> must be doing something wrong with the way I am adding the steps to my call;
> I am not familiar with variables and flags.

What *exactly* are you trying to achieve?

I have used both paging and intercom mode in the Grandstreams with good 
results.

You do need the settings in the phone set ON - ie.

Allow Auto Answer by Call-Info:   No  Yes
Turn off speaker on remote disconnect:   No  Yes

These both need to be set to YES or ON.

That won't affect normal calls to that account on the phone - although the 
"turn off speaker" one does make the phone easier to use IMO...

So call the phone and the person answers normally, as before, but if you 
rhen add the SIP header:

SIPAddHeader(Call-Info: answer-after=0)

The phone will auto-answer - when the next Dial or Page command is 
directed to it.

What next? If you want to Page the phone, use the Page() application.

So if the phone is SIP/100 then to Dial the phone normally..

 exten => 100,1,Dial(SIP/100)

but to page it:

 exten => 200,1,SIPAddHeader(Call-Info: answer-after=0)
 exten => 200,n,Page(SIP/100)

and to intercom to it:

 exten => 300,1,SIPAddHeader(Call-Info: answer-after=0)
 exten => 300,n,Page(SIP/100,d)


So this has added 3 new extensions, 100, 200 and 300 - which all 'call' 
SIP/100, but in 3 differet ways.

Gordon

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Re: [asterisk-users] Randulo: An open suggestion for the VOIP users Conference

[mgraves]

On Thu, 07 Aug 2008 00:28:29 -0500, Karl Fife wrote:

>Example:   Last week there was talk about Polycom's "HDVoice"
>technology, and the term was being used interchangeably with G.722.  In
>fact there are important distinctions, but someone listening might
>presume that the information was correct and leave short-changed.  There
>are other examples even from last week, one involving someone's claim
>that there's not a way to pick up a phone and directly interface with a
>voice recognition directory application without needing to press some
>digits first.  As it turns out, it's easy if you know the trick.  
>
>Id' be happy to put my money where my mouth is and kick off this
>Friday's show with these examples & any others I'm not remembering at
>this moment if you think it would be well received.  Perhaps others will
>do the same.
>What do you think?  
>
>Thanks!
>-Karl Fife
>
>If you want to discuss this off-list, you can email me at
>[EMAIL PROTECTED]
>
>p.s.
>As it turns out, HDVoice CAN use G.722, but it can also be overlain onto
>other codec's such as use G.722.1 and even G.711µ [sic].  That's right,
>you can have an "HDVvoice" call over the PSTN using G.711, using a
>special companding overlay on top G.711.  As I understand it, the two
>HDVoice compliant endpoints (Polycom, Cisco & others that license the
>technology) have an in-band (but inaudible) handshake, and then begin
>applying the proprietary companding overlay which extends the dynamic
>range of the audio.  It sounds great even though the underlying codec is
>not a wideband codec.  Certainly the sound is not as good as HDVoice
>over a modern adaptive-transform codec like G.722 (1987) or even better
>over G.722.1 (1999), but it's definitely a big improvement over the
>"Toll-Quality" (Read: AM-Radio-Through-A-Pillow) that we're all used to,
>and it is not dependent upon having a pure-IP connection involving ENUM,
>DUNDI, or other non e.164 namespaces such as SIP URI's, ITAD Subscriber
>Numbers etc.  In my opinion HDVoice is it's a brilliant transition
>technology.  
>

Karl,

This is very interesting. Did you see that Polycom made G.722.1
available through a royalty free license earlier th
week?

http://www.polycom.com/usa/en/company/news_room/press_releases/2008/2008
0805.html

In tinkering with the three phones that I have (ip650/550 & Siemens
S685IP) they all support only G.722. At least according to the
datasheets even the Polycom models don't handle G.722.1 as yet. 

Mind you I haven' gone so far as to use Wireshark to analyse the
traffic. Just measure the bandwith used across my router and note when
the phone indicates "HD" engaged on the line button.

Do you know if the companded processing you mention is implemented in
the Soundpoint models? Just from the sound of it it could improve S/N
ratio, be perhaps not frequency response. Still, it's good that they
can improve a call over the PSTN.

So, perhaps there's more to HDVoice than just G.722. Even so, G.722 is
all that I have experienced of HDVoice in their current Soundpoint IP
phones. I suspect that some of their technology is only deployed in
their larger conferencing systems, and not in the Soundpoint lineup.

Michael
--
Michael Graves
mgravesmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves





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Re: [asterisk-users] Max amount of concurrent calls on and iax trunk

[Gordon Henderson]

On Thu, 7 Aug 2008, Mattias Andersson wrote:

> I agree bandwidth is the limit, however the reason to use IAX is it is
> saving bandwidth.
> I am runging 2 Trixbox CE with IAX over a 2 Mbit line.
> I have never had any isues with the IAX trunk.
> I wish that I could get son good IAX phones to the office, tan would we skip
> on Trixbox and run the phones directly over a VPN Chanel. SIP over VPN are
> giving more hassle then IAX sound wise.

So As I Understand It... The bandwidth saving on using IAX is using it in 
a trunk mode - using it on separate phones will use almost as much as SIP 
phones will.

In my (rough calculations!) world, a SIP call uses 80Kb/sec b/w (each way, 
but we'll quietly ignore thant for now) A single IAX call uses almost the 
same, (78 according to a quick check on iftop) but in an IAX trunk, each 
additional call is only adding on about 65Kb/sec - whereas in a SIP world, 
each additional call is adding on another 80Kb/sec... Or you can have 
approx. 15 to 20% more IAX calls than SIP calls over the same Internet 
path, if they're inside an IAX trunk.

Lets see if that works:

On a 2Mb line:

   SIP is: 2048/80  = 25 concurrent SIP calls.
   IAX is: ((2048-80)/65)+1 = 31 concurrent IAX calls.

(And 31 calls is 31*50 = 1550 packets a seconds each way, so make sure the 
router can handle that!)

Personally, I tend to suggest to my customers that if they have more than 
4 or 5 extensions in one location that they're better off with a "micro" 
PBX to trunk the calls out - especially if they're making a lots of calls, 
and a few internal calls (as on NAT it'll bounce the call out & in again 
)-: This is based on typical UK ADSL line speeds though (up to 420Kb/sec , 
outgoing speeds on normal lines, 830Kbps on 'business' lines) Although 
typically what happens is that they want to keep their existing 
branch-office POTS lines and integrate them into the "system", so they get 
a mini PBX rather than a micro one!

However I did see some more IAX phones recently when looking:

Atcom -
   http://www.voipon.co.uk/voip-ip-telephones-atcom-iax-ip-telephone-c-1_61.html

(thats a UK site, obvously)

It also irritates me when I see the likes of Snom (and Doro) phones which 
are Linux based not supporting IAX ...

Gordon

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[asterisk-users] problem controlling dialplan order

[Adrian Marsh]

Hi All,

 

On a 1.4.15 system, I've a context as below, where I need to catch some
specific US ranges and dial direct via SIP rather than a PSTN trunk.
But the logic always goes via the International Trunk and I cant see
why...

 

[local]

exten => _00165011091[45]0-9],1,NoOp(I AM HERE)

exten => _00165011091[45]0-9],n,Macro(setcli)

exten => _00165011091[45]0-9],n,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

exten => _00165011091[45]0-9],n,Hangup

 

. (same context)

 

Catch local (UK) numbers

exten => _0[1-9]X.,1,NoOp(Dialling UK number)

exten => _0[1-9]X.,n,Macro(setcli)

exten => _0[1-9]X.,n(jumpdial),Dial(SIP/+44${EXTEN:[EMAIL PROTECTED])

exten => _0[1-9]X.,jumpdial+101,Dial(${TRUNK}/${EXTEN},,Wr)

exten => _0[1-9]X.,n+101,Busy

 

;Catch any (00xx) numbers

exten => _00X.,1,NoOp(Dialling International number)

exten => _00X.,n,Macro(setcli)

exten => _00X.,n(jumpdial),Dial(SIP/+${EXTEN:[EMAIL PROTECTED])

exten => _00X.,jumpdial+101,Dial(${TRUNK}/${EXTEN},,Wr)

exten => _00X.,n+101,Busy

 

 

I've tried putting the Catch codes above into a sub-context, and then
put an include into the [local], but it still dials via the Catch
international...

The odd thing is that in either, the show dialplan seems to suggest the
correct order :

 

 

 

  '_00165011091[45]0-9]' => 1. NoOp(I AM HERE)
[pbx_config]

2. Macro(setcli)
[pbx_config]

3. Dial(SIP/${EXTEN:[EMAIL PROTECTED])  [pbx_config]

4. Hangup()
[pbx_config]

 (some others)

  '_00X.' =>1. NoOp(Dialling International number)
[pbx_config]

2. Macro(setcli)
[pbx_config]

 [jumpdial] 3. Dial(SIP/+${EXTEN:[EMAIL PROTECTED])
[pbx_config]

104. Dial(${TRUNK}/${EXTEN}||Wr)
[pbx_config]

206. Busy()
[pbx_config]

 

 

The page at voip-info isn't too clear in the differences between 1.2 and
1.4
(http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf+sort
ing) so I'm not sure where I've gone wrong.

 

 

Adrian Marsh

 

 

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Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 16

[dimitri . osler]

Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a 
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al 
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], 
altrimenti vi risponderò al mio rientro.

Dimitri Osler

I will be on vacation until Tuesday 19th of August with limited access to voice 
and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 
0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on 
my return.

Dimitri Osler





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Re: [asterisk-users] Max amount of concurrent calls on and iax trunk

[Thomas Kenyon]

Mattias Andersson wrote:
> I agree bandwidth is the limit, however the reason to use IAX is it is 
> saving bandwidth.
> I am runging 2 Trixbox CE with IAX over a 2 Mbit line.
> I have never had any isues with the IAX trunk.
> I wish that I could get son good IAX phones to the office, tan would we 
> skip on Trixbox and run the phones directly over a VPN Chanel. SIP over 
> VPN are giving more hassle then IAX sound wise.
> 
In this setup, there can be a bandwidth saving with keeping both the 
servers so that you can use trunking=yes in the trunk definition in 
iax.conf.

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[asterisk-users] [HELP] Regarding stripping of fmtp parameters for Video.

[SiM]

Hello All,
   I'am doing a video call between two Video Phones, and i see
that Asterisk is stripping the fmtp parameters for the h263 video line in
SDP.
For example a line similar to the below is stripped,

 a=fmtp:xx CIF=4;QCIF=2;F=1;K=1

Asterisk is configured NOT to be present in the Media path (My version :
Asterisk 1.4.19.1 ).
I have the following enabled in my sip.conf.

canreinvite=yes
directrtpsetup=yes

>From what i have read on the internet, i feel fmtp parameters are not
supported by Asterisk for Video.
I also find that video_caps branch has a fix for this problem, please can
someone share more information
about this and where i can find it ?

I DO NOT want those fmtp lines to be stripped. Suggestions to change the
Asterisk config files, to achieve this are also welcome.

Thank you.


Best regards,
Simith
PS: I see that my last e-mail has been truncated when i check the archives.
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Re: [asterisk-users] Max amount of concurrent calls on and iax trunk

[Mattias Andersson]

I agree bandwidth is the limit, however the reason to use IAX is it is
saving bandwidth.
I am runging 2 Trixbox CE with IAX over a 2 Mbit line.
I have never had any isues with the IAX trunk.
I wish that I could get son good IAX phones to the office, tan would we skip
on Trixbox and run the phones directly over a VPN Chanel. SIP over VPN are
giving more hassle then IAX sound wise.

On Thu, Aug 7, 2008 at 1:45 AM, Chris Brentano <
[EMAIL PROTECTED]> wrote:

> I have two Asterisk 1.4 boxes connected via IAX over a VPN tunnel on a
> 10Mbit link. We never did any stress testing as it's a temporary
> arrangement, but we've never had any call quality issues or run up
> against concurrent ca, holl limitations. I'm mostly routing internal
> extensions over the trunk, and in the case of two floating users I
> have their extensions at each office ring when their DID is called.
> One server is an older Pentium 4 1.7 GHz with 1GB Ram, and the other
> is a Dual Xeon 2.33 GHz with 4GB Ram. As for codec, I'm disallowing
> all except ulaw and gsm, with ulaw the priority codec for hardphones
> (Polycom) and gsm the priority for softphones (X-Lite, Zoiper).
>
> I would expect the limitation you're going to run up against is not
> Asterisk, but the bandwidth between your two systems.
>
>
> On 6 Aug, 2008, at 10:40 AM, Rosli Sukri wrote:
>
> > hi,
> > wanted to ask if anybody has experienced setting up two asterisk 1.2
> > boxes connected via iax trunk. have u guys ever stress tested the
> > trunks i.e how many concurrent calls can a trunk handle and whether
> > codec has any effect on it.
> > 
>
>
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-- 
Mattias Andersson

Storskiftesvägen 6
145 60 Norsborg

m. +46-70-799 44 41
h. +46-8-641 38 97

Email: [EMAIL PROTECTED]
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Re: [asterisk-users] Digium B410P: problematic Bri connection between * and a legacy Philips PBX

[Mr Shunz]

> Hi all,
>
[snip]

> For this reason I set all the 4 ports of Digium's card in NT mode
> (Philips can not do this). Then i opportunely
> edited /etc/misdn-init.conf and /etc/asterisk/misdn.conf. In fact, when
> I run the command "misdn shows stacks" in * CLI, I can see all ports in
> NT (PTP) mode.

have you tried to configure the ports in PTMP mode instead of PTP?
I found that for some PBX works better...

[snip]

> p.s.: I already attempt to change some options
> in /etc/asterisk/misdn.conf file. I tried to put
> incoming_early_audio=yes (explanation: Rarely used. If turned on, sends
> Tone Indications on TE Port for Incoming isdn channel. Normally the
> telcos send that informaton. By default is 'no'). It didn't work...

can you post your misdn-init.conf and misdn.conf?

cheers

-- 

Daniele Santi   .o.
[EMAIL PROTECTED] ..o () ascii ribbon campaign
Linux User #415108  ooo /\ www.asciiribbon.org


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Re: [asterisk-users] shared mysql connection in dialplan

[Rizwan Hisham]

have done it, and its working fine. but still expecting to receive some new
ideas.

On Wed, Aug 6, 2008 at 2:12 PM, Rizwan Hisham <[EMAIL PROTECTED]>wrote:

> hi all,
> i just finished developing some incoming call features in a macro. that
> macro gets executed everytime an incoming call is received and a new mysql
> connection is made using the MYSQL cmd in dialplan. i want to use a single
> mysql connection for every incoming call.
>
> my idea of doing it is like this, i want to get a mysql connection in a
> global variable, just to share the connection with different incoming calls.
> Im not sure if this can be done. I am going to try doing it somehow,
> meanwhile i want your suggestions about how i can share a mysql connection
> with different calls in a dialplan.
>
> I am using asterisk1.4.2 and asterisk addon1.4.0 package for mysql
> connectivity.
>
> Thanx in advance
>
> --
> Best Regards
> Rizwan Hisham
>
>


-- 
Best Regards
Rizwan Hisham
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