[asterisk-users] Dail in modem

[ABBAS SHAKEEL]

Hello

I am required to do some thing like  Dail in modem .
User will have to call a modem just like we do in dail up connection
now we need to handle that request and retrieve some parameters
from that send a HTTp request to a web server and then after getting
http response send user a feed back ..


this is a requirement ..

Is it possible ??

what is the way forward ??


please give me a direction


Best Regards
Shakeel Abbas

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[asterisk-users] Asterisk Flite Problem

[DHAVAL INDRODIYA]

Hello

I found following issue while trying to load flite module from CLI

 module load app_flite.so
Unable to load module app_flite.so
Command 'module load app_flite.so ' failed.
[Jun 19 11:31:12] WARNING[23507]: loader.c:384 load_dynamic_module: Module
'app_flite.so' did not register itself during load
[Jun 19 11:31:12] WARNING[23507]: loader.c:662 load_resource: Module
'app_flite.so' could not be loaded.


even flite is also loaded and app_flite.so is located in
/usr/lib/asterisk/module

please help me to solve this problem and also give information regarding any
good tts engine which can be used with asterisk

regards
Dhaval
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Re: [asterisk-users] Recompiling dahdi-linux after kernel update - To minimize downtime

[Alex Samad]

On Thu, Jun 18, 2009 at 11:24:40PM -0500, Karl Fife wrote:
> After a kernel update (but before rebooting) Is there a way to recompile 
> Zap/Dahdi against the new kernel?
> 
> My objective is to eliminate the additional downtime that occurs while 
> recompiling/installing zap/dahdi after booting into the new kernel.
> 
> Please correct me if I'm wrong:  
> My understanding is that until you reboot (after a kernel update), 
> recompiling zap/dahdi still compiles against the OLD kernel, and that's why 
> zap/dahdi doesn't start after rebooting into the new kernel (even if you 
> recompiled it just before rebooting).  
> 
> So my question is: 
> Is there a method to recompile dahdi/zap against the new kernel such that the 
> only downtime is the actual server bounce itself?  OR is the current best 
> practice just to simply to reboot, recompile, restart?  

without trying to start a distro war under debian you can do 


m-a -t build -l  dahdi

aslong as you have the headers installed it will build a module against
it.

I think if you use the tgz tar ball you can actually specify KDIR to
point to the directory with the headers in it.



> 
> Thanks in advance.
> 
> -Karl 

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- George W. Bush
07/08/2002
White House press conference, Washington, D.C.


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[asterisk-users] Anonymous Connection form IP to use specific Context

[David Klaverstyn]

Hi All,

How can I force an anonymous SIP connection from a certain IP address to use a 
specific context rather than the default one defined in sip.conf.

I am using Asterisk 1.6.0.9

Regards
David Klaverstyn


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[asterisk-users] Recompiling dahdi-linux after kernel update - To minimize downtime

[Karl Fife]

After a kernel update (but before rebooting) Is there a way to recompile 
Zap/Dahdi against the new kernel?

My objective is to eliminate the additional downtime that occurs while 
recompiling/installing zap/dahdi after booting into the new kernel.

Please correct me if I'm wrong:  
My understanding is that until you reboot (after a kernel update), recompiling 
zap/dahdi still compiles against the OLD kernel, and that's why zap/dahdi 
doesn't start after rebooting into the new kernel (even if you recompiled it 
just before rebooting).  

So my question is: 
Is there a method to recompile dahdi/zap against the new kernel such that the 
only downtime is the actual server bounce itself?  OR is the current best 
practice just to simply to reboot, recompile, restart?  

Thanks in advance.

-Karl 
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[asterisk-users] Analogue card recommendation

[Alex Samad]

Hi

I have 2 digium cards (tdm410) with combination of fxs + multiple fxo
ports.

I have had a quick look at sangoma B series cards. I was wondering if
there is a card out there with 

hardware echo canceller
say max 4 ports (mix of fxs/fxo)
g729 encoding onboard


Alex


-- 
"More and more of our imports are coming from overseas."

- George W. Bush
09/26/2005
On NPR's Morning Edition


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Re: [asterisk-users] snom mass deploy help

[Alex Samad]

On Thu, Jun 18, 2009 at 02:21:47PM +0200, Philipp Kempgen wrote:
> > On Jun 18, 2009, at 7:25 AM, Alex Samad  wrote:
> >> I am trying to setup asterisk to do a mass deploy of some snom  
> >> phones. I
> >> can't find where i configure asteriks to listen to the multicast
> >> address, nor where to set the notify reply.
> >>
> >> I was hoping to not have to use dhcp options
> 
> Alex Balashov schrieb:
> > I thought TFTP (and therefore, DHCP option 66) is the only
> > autoprovisioning method Asterisk supports?
> 
> Asterisk is not involved here at all.
> Snom supports what they call "PnP config".
> Technically:
> ---cut---
> # SIP Event Notification:
> #   http://tools.ietf.org/html/rfc3265
> # SIP UA Profile Event Package:
> #   http://tools.ietf.org/html/draft-ietf-sipping-config-framework-15
> #   http://tools.ietf.org/html/draft-channabasappa-sipping-app-profile-type-03
> #
> # Snom 3xx:
> #   http://wiki.snom.com/SIP_Traces#PnP_Config
> 
> # other drafts:
> #   http://tools.ietf.org/html/draft-petrie-sip-config-framework-01
> #   
> http://www.cs.columbia.edu/sip/drafts/sip/draft-schulzrinne-sip-config-events-00.txt
> ---cut---
> 
> Gemeinschaft (Asterisk-based open-source PBX) comes with a SIP UA
> config responder.
> 
> http://www.amooma.de/gemeinschaft/
> https://svn.amooma.com/gemeinschaft/trunk/opt/gemeinschaft/sbin/gs-sip-ua-config-responder/gs-sip-ua-config-responder

I saw links to this on the voip-info site, but my german is non
existant. but on first glance through this seems to be what I want.



> 
> GNU GPL.
> 
> 
> Philipp Kempgen

-- 
"I always jest to people, the Oval Office is the kind of place where people 
stand outside, they're getting ready to come in and tell me what for, and they 
walk in and get overwhelmed in the atmosphere, and they say, man, you're 
looking pretty."

- George W. Bush
11/04/2004
Washington, DC


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Re: [asterisk-users] snom mass deploy help

[Alex Samad]

On Thu, Jun 18, 2009 at 11:57:20PM +0200, Philipp Kempgen wrote:
> Alex Samad schrieb:
> 
> > seems like the documentation from snom for V7, includes the pnp method
> > as well. it sends a subscribe to a multicast address (224.0.1.75) and the 
> > listener is
> > meant to respond with a notify which has the url which is normally sent
> > my dhcp, i was hoping to use that
> 
> > I think I would prefer this method, but I can't find where to set
> > asterisk to listen to the multicast address nor where to program the
> > notify reply
> 
> I have already told you that Asterisk is not involved in the process
> of configuring the phone.

sorry go the replies after I had sent the email

> In order to use Snom's PnP configuration method you have to write a
> daemon which opens a socket on 224.0.1.75 (sip.mcast.net), join the
> multicast group, read packets and send appropriate "ua-profile"
> notification events.

This is not what I had hopped for, I had hoped it was something asterisk
could handle

> Have a look at the code I mentioned to get the idea.

I will have a look 

> 
> 
> Philipp Kempgen

-- 
"I know what I believe. I will continue to articulate what I believe and what I 
believe I believe what I believe is right."

- George W. Bush
07/22/2001
Rome, Italy


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Re: [asterisk-users] help setting up transfering

[Alex Samad]

On Thu, Jun 18, 2009 at 08:06:24AM -0500, Danny Nicholas wrote:
> Have you tried #1103 or *2103?  The # would do a blind transfer, the * would
> initiate an attended transfer.

I tried combination of 

flash 
flash *67 



*67 


and re iterated with # instead of *.

The doco seemed to suggest after I press flash I should heard a dial
tone ! which i don't 


Alex

> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Samad
> Sent: Wednesday, June 17, 2009 9:41 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] help setting up transfering
> 
> Hi
> 
> I am trying to get transferring of calls working, I place a call from
> ext 101 => 103 and then from 101 I try and transfer the call to 102
> (such that it will be 102=>103), I have tried flash and *2 and nothing
> seems to work.
> 
> I have allowed transfers in sip.conf, I am expecting a dial tone when i
> hit flash
> 
> 101 -> dahdi/1 a uniden pots phone
> 
> 
> Thanks
> Alex
> 
> 
> 
> 
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> 

-- 
"I always jest to people, the Oval Office is the kind of place where people 
stand outside, they're getting ready to come in and tell me what for, and they 
walk in and get overwhelmed in the atmosphere, and they say, man, you're 
looking pretty."

- George W. Bush
11/04/2004
Washington, DC


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Re: [asterisk-users] Voicemail Password

[Jonathan Thurman]

I believe that 'externpasscheck' was added in the 1.6 branch.  Since we use
this, I wrote a quick perl script that checks for password length,
difficulty, repeated digits, etc. which are required for us.  If you get it
back-ported to the version you are on you can have the script, just contact
me off-list.  This is probably one of the best features added to
app_voicemail for 1.6.

-Jonathan


On Thu, Jun 18, 2009 at 11:40 AM, Darrin Henshaw wrote:

> As usual my manager comes up with some obscure reference I didn't find.
> There seems to be a parameter called minpassword described here:
>
> http://www.asterisk.org/doxygen/trunk/Config_vm.html
>

> But from further digging it looks like it's a 1.6.1.0 feature. Might see
> about a backport if possible.
>
> Cheers,
>
> Darrin Henshaw
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
> Sent: Thursday, June 18, 2009 15:19
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Voicemail Password
>
> On Thu, 18 Jun 2009, Darrin Henshaw wrote:
>
> > Does anyone know of a way to force the voicemail password for users to
> > be of a certain length? We've setup operator=yes within our
> > voicemail.conf and want to have the users use a long password to prevent
> > possible guessing by external parties. I'm not seeing any such option in
> > my research. If it doesn't exist it might be a decent feature. Thanks.
>
> Sounds like a cool feature. I started looking into it, checking out
> voicemail.conf (1.2) to get an idea of a good name to call the parameter
> and I found this:
>
> ; If you need to have an external program, i.e. /usr/bin/myapp called when
> ; a voicemail password is changed, uncomment this:
> externpass=/usr/bin/myapp
>
> Who knew?
>
> Thanks in advance,
> 
> Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
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Re: [asterisk-users] asterisk and openvpn and sip

[John A. Sullivan III]

On Thu, 2009-06-18 at 14:55 +0200, Giorgio Incantalupo wrote:
> Hi John,
> 
> I already have the ccd dir with the iroute (mandatory for routing to 
> pc/phone connected to vpn client). During the last test I could register 
> and  make a call but voice disappears after 1, 2 seconds. I'm trying to 
> understand if it is a bandwidth problem. At the moment I have my phone 
> connected to the openvpn client (which is its gateway) but I have to use 
> the vpn ip (10.0.0.1) to register the phone, the openvpn server local ip 
> (192.168.1.12) is not working. I suppose it is a  sip protocol problem: 
> I had to change the sip.conf setting nat=yes to make the phone dial and 
> domain = 10.0.0.1 to make the voice pass (or at least the first 2 seconds).
> I keep on working on the vpn since it seems so little is missing to have 
> a clear conversation. Let me know if your tests are successfull.
> 
> Thank you. 
> 
> Giorgio

Hi, Giorgio.  So far so good.  I have twinkle running on my laptop (the
VPN client), a Snom 320 and a Snom 360 on the internal network routing
through my laptop.  I haven't done much more than register and execute a
very basic dialplan but it is all working so far.

I hit a couple of small bumps but nothing to do with *.  I had forgotten
to tell my DNS to accept requests from the test network.  One of the
phones somehow decided the data center firewall was an outbound SIP
proxy.  Once I removed that setting, it all worked just fine.

I am using native addresses across the VPN; there is no NAT.

I've not yet had sustained conversations.  I'll be doing that in a while
hopefully - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] Wideband (G722) MeetMe

[Michael Graves]

I'm told that Asterisks wideband capability is exclusively based upon
16 khz sampling. Higher sampling rates, like you might find with CELT,
are downsampled for mixing at 16 khz.

My guess is that not everyone will be happy with this, especially
vendors trumpeting codecs with higher sampling rates. To handle their
output will be more CPU intensive.

I'm not an Asterisk insider. I just listen closely when the real
insiders speak up ;-)

Michael

--Original Message Text---
From: Doken, Serhad
Date: Thu, 18 Jun 2009 14:15:42 -0700



Thanks Michael. I guess prior to 1.6.2, Asterisk was downgrading
streams to SLIN before mixing and then mixed stream got upgraded to WB.


  

My question is, with this release, is Asterisk converting WB codecs to
SLIN16 and mix them that way ? That seems to be the logical way to me
just wanted an insider expert to confirm/deny that. 

  

Is this the right list to ask that question/find the right contact
before I delve into the code knee deep ? 

  

Serhad Doken 

  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael
Graves
Sent: Thursday, June 18, 2009 6:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Wideband (G722) MeetMe 



  

--Original Message Text---
From: Doken, Serhad
Date: Wed, 17 Jun 2009 16:07:12 -0700



Hi, 

I wanted to follow up on this thread about WB support on the MeetMe
bridge that is in 1.6.2. Does it only work for G722 or any WB codec ? 

I am working with another 16k WB codec that I can transcode to 722 and
vice versa. I was curious if the 1.6.2 MeetMe bridge can also mix 722
with any other WB codec natively(without downscaling). 

Thanks, 

Serhad Doken 

While not an expert in Asterisk internal, it seems unlikely that
Asterisk is mixing signals in encoded space. It's most likely
converting the stream to slin for mixing then encoding back into
whatever is most appropriate for each end-point.

Michael 

--
Michael Graves
mgravesmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:mgra...@mstvp.onsip.com
skype mjgraves
fwd 54245 



--
Michael Graves
mgravesmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:mgra...@mstvp.onsip.com
skype mjgraves
fwd 54245


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Re: [asterisk-users] Multiple Outgoing Lines: extensions.conf (Ioan Indreias)

[Barry L. Kline]

Clara Chan wrote:
> Loan, 
> 
> Thanks for your help in this matter.
> 
> Having never used astdb before, can you point me to an example on this??
> 
> Thanks hugely, 
> Clara

Clara --

You need to read "the book".  In it you'll find examples.

Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9)

It's downloadable at http://www.asteriskdocs.org

Barry


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Re: [asterisk-users] snom mass deploy help

[Daniel Hazelbaker]

On Jun 18, 2009, at 2:57 PM, Philipp Kempgen wrote:

>> I think I would prefer this method, but I can't find where to set
>> asterisk to listen to the multicast address nor where to program the
>> notify reply
>
> I have already told you that Asterisk is not involved in the process
> of configuring the phone.
> In order to use Snom's PnP configuration method you have to write a
> daemon which opens a socket on 224.0.1.75 (sip.mcast.net), join the
> multicast group, read packets and send appropriate "ua-profile"
> notification events.
> Have a look at the code I mentioned to get the idea.

As Philipp said, you don't.  However it would make a great 3rd party  
module that could be added to Asterisk.  I use a combination of the  
PnP and web redirects (early V6 versions did not support the PnP, but  
they do automatically request a file from the DHCP web server) and  
MySQL databases.  It is now set where we just add the MAC address to  
the database and plug the phone it.  It auto-configures the rest  
(along with firmware updates).

Daniel

>
>
>Philipp Kempgen
> -- 
> AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
> Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
> Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
> Videos of the AMOOCON VoIP conference 2009 ->  http://www.amoocon.de
> -- 
>
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Re: [asterisk-users] snom mass deploy help

[Philipp Kempgen]

Alex Samad schrieb:

> seems like the documentation from snom for V7, includes the pnp method
> as well. it sends a subscribe to a multicast address (224.0.1.75) and the 
> listener is
> meant to respond with a notify which has the url which is normally sent
> my dhcp, i was hoping to use that

> I think I would prefer this method, but I can't find where to set
> asterisk to listen to the multicast address nor where to program the
> notify reply

I have already told you that Asterisk is not involved in the process
of configuring the phone.
In order to use Snom's PnP configuration method you have to write a
daemon which opens a socket on 224.0.1.75 (sip.mcast.net), join the
multicast group, read packets and send appropriate "ua-profile"
notification events.
Have a look at the code I mentioned to get the idea.


Philipp Kempgen
-- 
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 ->  http://www.amoocon.de
-- 

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Re: [asterisk-users] asterisk-gui:http://id_address:8088/asterisk/static/config/cfgadvanced.html

[Danny Nicholas]

/var/lib/asterisk/static-http/config/cfgadvanced.html is the file location.
The root directory is /var/lib/asterisk unless you change it.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Thursday, June 18, 2009 4:48 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
asterisk-gui:http://id_address:8088/asterisk/static/config/cfgadvanced.html


Hello List;

Actually based on what I read at Guru that after I did the installation and
configuration of the asterisk-gui, I can access it using the link:

http://id_address:8088/asterisk/static/config/cfgadvanced.html

I tried to search for something like
/asterisk/static/config/cfgadvanced.html but did not find it at all, where
this cfgadvanced.html?

Another issue: if we look for the above link, the question is: do I
configure the httpd server and determine the root directory, so the root
directory should contain the /asterisk/static/config/cfgadvanced.html? Any
advise?

So how the installation will know the default httpd path and install the
asterisk/static/config/cfgadvanced.html under that default?

Any advise?

Regards
Bilal


  

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[asterisk-users] asterisk-gui: http://id_address:8088/asterisk/static/config/cfgadvanced.html

[bilal ghayyad]

Hello List;

Actually based on what I read at Guru that after I did the installation and 
configuration of the asterisk-gui, I can access it using the link:

http://id_address:8088/asterisk/static/config/cfgadvanced.html

I tried to search for something like /asterisk/static/config/cfgadvanced.html 
but did not find it at all, where this cfgadvanced.html?

Another issue: if we look for the above link, the question is: do I configure 
the httpd server and determine the root directory, so the root directory should 
contain the /asterisk/static/config/cfgadvanced.html? Any advise?

So how the installation will know the default httpd path and install the 
asterisk/static/config/cfgadvanced.html under that default?

Any advise?

Regards
Bilal


  

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Re: [asterisk-users] snom mass deploy help

[Philipp Kempgen]

Conrad Wood schrieb:
>> > On Jun 18, 2009, at 7:25 AM, Alex Samad  wrote:
>> >> I am trying to setup asterisk to do a mass deploy of some snom  
>> >> phones. I
>> >> can't find where i configure asteriks to listen to the multicast
>> >> address, nor where to set the notify reply.
> 
> FWIW I use a home-grown cgi script to configure the mass-deploy.
> (attached)

I fail to see how this script is useful in order to use Snom's
"Plug'n'play config".
A simple factory reset should enable the pnp_config setting.


Philipp Kempgen
-- 
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 ->  http://www.amoocon.de
-- 

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Re: [asterisk-users] snom mass deploy help

[Alex Samad]

On Thu, Jun 18, 2009 at 07:34:38AM -0400, Alex Balashov wrote:
> I thought TFTP (and therefore, DHCP option 66) is the only  
> autoprovisioning method Asterisk supports?

seems like the documentation from snom for V7, includes the pnp method
as well. it sends a subscribe to a multicast address (224.0.1.75) and the 
listener is
meant to respond with a notify which has the url which is normally sent
my dhcp, i was hoping to use that

this links talks a bit about multicast
http://www.voip-info.org/wiki/view/SIP+registrar+server

I think I would prefer this method, but I can't find where to set
asterisk to listen to the multicast address nor where to program the
notify reply


Alex


> 



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Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

[Andres]

 > Protocol Discriminator: Q.931 (8)  len=5
 > Call Ref: len= 2 (reference 27418/0x6B1A) (Originator)
 > Message type: CONNECT ACKNOWLEDGE (15)
!! Got reject for frame 69, but we have nothing -- resetting!
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate
Connect Request

The remote switch does not like your CONNECT ACKNOWLEDGE message.  I 
have no idea why but my first guess would be to play around with the 
'switchtype' in your chan_dahdi.conf.  Another thing to try is to 
enable/disable 'facilityenable' as well to see if it changes anything.

Andres
http://www.neuroredes.com

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Re: [asterisk-users] asterisk-gui: read/write in the conf files or db

[Tzafrir Cohen]

On Thu, Jun 18, 2009 at 11:27:18AM -0700, bilal ghayyad wrote:
> 
> Hi Danny;
> 
> Really I did not understand how I can determine if the IO will be DB or 
> conf files? Is it from the Asterisk manager?

Again, unless you make some pretty major changes in the way the
asterisk-gui[1] works, it will use Asterisk configuration files as its
database. E.g. users.conf to store extensions and trunks.

[1] http://svn.asterisk.org/svn/asterisk-gui , and not the generic
concept of "A GUI for Asterisk", of course.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Multiple Outgoing Lines: extensions.conf (Ioan Indreias)

[Clara Chan]

Loan, 

Thanks for your help in this matter.

Having never used astdb before, can you point me to an example on this??

Thanks hugely, 
Clara

>>
Hi Clara,
You could put some data into astdb and query for the outgoing line and
callerid based on internal callerid (extension).

something like

user/201/outline 89859715
user/201/outcallerid 89859715

and so on...

By the way: "_89859715" without the dot (".") is same like 89859715  -
maybe
you renounce to the underline ...

HTH,
Ioan.

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Re: [asterisk-users] Wideband (G722) MeetMe

[Doken, Serhad]

Thanks Michael. I guess prior to 1.6.2, Asterisk was downgrading streams to 
SLIN before mixing and then mixed stream got upgraded to WB.

My question is, with this release, is Asterisk converting WB codecs to SLIN16 
and mix them that way ? That seems to be the logical way to me just wanted an 
insider expert to confirm/deny that.

Is this the right list to ask that question/find the right contact before I 
delve into the code knee deep ?

Serhad Doken

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Graves
Sent: Thursday, June 18, 2009 6:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Wideband (G722) MeetMe

--Original Message Text---
From: Doken, Serhad
Date: Wed, 17 Jun 2009 16:07:12 -0700



Hi,

I wanted to follow up on this thread about WB support on the MeetMe bridge that 
is in 1.6.2. Does it only work for G722 or any WB codec ?

I am working with another 16k WB codec that I can transcode to 722 and vice 
versa. I was curious if the 1.6.2 MeetMe bridge can also mix 722 with any other 
WB codec natively(without downscaling).

Thanks,

Serhad Doken

While not an expert in Asterisk internal, it seems unlikely that Asterisk is 
mixing signals in encoded space. It's most likely converting the stream to slin 
for mixing then encoding back into whatever is most appropriate for each 
end-point.

Michael
--
Michael Graves
mgravesmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:mgra...@mstvp.onsip.com
skype mjgraves
fwd 54245
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Re: [asterisk-users] asterisk-gui: read/write in the conf files ordb

[Danny Nicholas]

Here's the .05 tour as I know it:
Every Asterisk installation reads /etc/asterisk/asterisk.conf and
/etc/extconfig.conf at startup.  Once these files are parsed, the remaining
IO is done based off of how these two files point.

When you run the GUI, it does a series of AMI calls to get it's information
and do it's "under the covers" work.  What is does is exactly what would
happen if you issued the AMI commands from telnet or any other source.

So in so many words, Asterisk will (hopefully) do what you tell it in the
.conf files and any supporting database tables.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Thursday, June 18, 2009 3:48 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk-gui: read/write in the conf files
ordb


I am not able to understand the relation between the AMI and the GUI? And
really I am not able to know where to determine if my Asterisk will
read/write with DB or with config files?

Regards
Bilal

---

Why not connect to the AMI via telnet?

On Thu, Jun 18, 2009 at 2:27 PM, bilal ghayyad  wrote:

>
> Hi Danny;
>
> Really I did not understand how I can determine if the IO will be DB or
> conf files? Is it from the Asterisk manager?
>
> Regards
> Bilal
>
> -
>
> It depends on how you are configured.  The gui interfaces using Asterisk
> Manager, so you get the Same IO from the gui that you would get from a
> native manager session.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal
> ghayyad
> Sent: Wednesday, June 17, 2009 5:36 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] asterisk-gui: read/write in the conf files or
db?
>
>
> Hi All;
>
> asterisk-gui read/write from the conf files or database?
>
> Any advise?
> Regards
> Bilal



  


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Re: [asterisk-users] asterisk-gui: read/write in the conf files or db

[bilal ghayyad]

I am not able to understand the relation between the AMI and the GUI? And 
really I am not able to know where to determine if my Asterisk will read/write 
with DB or with config files?

Regards
Bilal

---

Why not connect to the AMI via telnet?

On Thu, Jun 18, 2009 at 2:27 PM, bilal ghayyad  wrote:

>
> Hi Danny;
>
> Really I did not understand how I can determine if the IO will be DB or
> conf files? Is it from the Asterisk manager?
>
> Regards
> Bilal
>
> -
>
> It depends on how you are configured.  The gui interfaces using Asterisk
> Manager, so you get the Same IO from the gui that you would get from a
> native manager session.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal
> ghayyad
> Sent: Wednesday, June 17, 2009 5:36 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] asterisk-gui: read/write in the conf files or db?
>
>
> Hi All;
>
> asterisk-gui read/write from the conf files or database?
>
> Any advise?
> Regards
> Bilal



  


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Re: [asterisk-users] Installing LUA

[John A. Sullivan III]

On Thu, 2009-06-18 at 14:20 -0500, Tilghman Lesher wrote:
> On Wednesday 17 June 2009 16:54:53 John A. Sullivan III wrote:
> > On Wed, 2009-06-17 at 15:43 -0500, Tilghman Lesher wrote:
> > > On Wednesday 17 June 2009 14:55:55 John A. Sullivan III wrote:
> > > > On Wed, 2009-06-17 at 14:18 -0500, Tilghman Lesher wrote:
> > > > > On Wednesday 17 June 2009 11:56:28 John A. Sullivan III wrote:
> > > > > > On Wed, 2009-06-17 at 12:44 -0400, Sean Bright wrote:
> > > > > > > Watkins, Bradley wrote:
> > > > > > > > One more bit of magic necessary here, as pbx/pbx_lua.c has
> > > > > > > > includes for: #include 
> > > > > > > > #include 
> > > > > > > > #include 
> > > > > > > >
> > > > > > > > On Redhat-based systems, it needs to be:
> > > > > > > > #include 
> > > > > > > > #include 
> > > > > > > > #include 
> > > > > > >
> > > > > > > Gah.  OK.  So the patch I supplied will find LUA at configure
> > > > > > > time but not compile time.  This needs some more thought.
> > > > > >
> > > > > > Oops! Confirmed.  menuselect found it but make failed.  I'll make
> > > > > > the edits by hand and let you know how I fared - John
> > > > >
> > > > > Try this patch:
> > > > > http://asterisk.drunkcoder.com/patches/20090617__luafix.diff.txt
> > > >
> > > > Alas, the patch fails completely.  The configure revision numbers look
> > > > very different.  I am using the 1.6.1.1 tarball.  I copied the patch
> > > > into the source directory (asterisk-1.6.1.1) and ran patch -p0 <
> > > > 20090617__luafix.diff.txt.  It was a long list of failed hunks.  What
> > > > next? Thanks - John
> > >
> > > The patch is actually derived from the current 1.6.1 SVN source, so any
> > > changes made since 1.6.1.1 was branched are the difference.  Updated
> > > patch:
> > > http://asterisk.drunkcoder.com/patches/20090617__luafix__1.6.1.1.diff.txt
> >
> > The patch applied cleanly but Asterisk segfaulted on startup.  I did not
> > see anything particularly suspicious in the console output - mostly
> > warnings and errors about realtime database connectivity that we have
> > not yet setup other than this message:
> >
> > [Jun 17 17:47:24] WARNING[30682]: loader.c:375 load_dynamic_module:
> > Error loading module 'app_directory.so': /usr/lib64/libc-client.so.1:
> > undefined symbol:
> > mm_dlog
> >
> > As much as I'd like, I don't think I have the code knowledge and I don't
> > have the time on this project to step through it with a debugger.  What
> > next? Thanks - John
> 
> The error above has zilch to do with pbx_lua, but rather with the compilation
> of app_directory against IMAP and the probable inability of your linker to
> find the correct IMAP libraries.  You can prove this by adding 'noload =>
> app_directory.so' to your modules.conf and restarting Asterisk.
> 
Yes, I didn't suspect it was related but tossed it in just in case.
Nonetheless, it is clear (at least to me) that lua is causing the
segfault.  When I remove it, the segfault goes away.  When I add it, the
segfault returns.  Thanks - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] Installing LUA

[Tilghman Lesher]

On Wednesday 17 June 2009 16:54:53 John A. Sullivan III wrote:
> On Wed, 2009-06-17 at 15:43 -0500, Tilghman Lesher wrote:
> > On Wednesday 17 June 2009 14:55:55 John A. Sullivan III wrote:
> > > On Wed, 2009-06-17 at 14:18 -0500, Tilghman Lesher wrote:
> > > > On Wednesday 17 June 2009 11:56:28 John A. Sullivan III wrote:
> > > > > On Wed, 2009-06-17 at 12:44 -0400, Sean Bright wrote:
> > > > > > Watkins, Bradley wrote:
> > > > > > > One more bit of magic necessary here, as pbx/pbx_lua.c has
> > > > > > > includes for: #include 
> > > > > > > #include 
> > > > > > > #include 
> > > > > > >
> > > > > > > On Redhat-based systems, it needs to be:
> > > > > > > #include 
> > > > > > > #include 
> > > > > > > #include 
> > > > > >
> > > > > > Gah.  OK.  So the patch I supplied will find LUA at configure
> > > > > > time but not compile time.  This needs some more thought.
> > > > >
> > > > > Oops! Confirmed.  menuselect found it but make failed.  I'll make
> > > > > the edits by hand and let you know how I fared - John
> > > >
> > > > Try this patch:
> > > > http://asterisk.drunkcoder.com/patches/20090617__luafix.diff.txt
> > >
> > > Alas, the patch fails completely.  The configure revision numbers look
> > > very different.  I am using the 1.6.1.1 tarball.  I copied the patch
> > > into the source directory (asterisk-1.6.1.1) and ran patch -p0 <
> > > 20090617__luafix.diff.txt.  It was a long list of failed hunks.  What
> > > next? Thanks - John
> >
> > The patch is actually derived from the current 1.6.1 SVN source, so any
> > changes made since 1.6.1.1 was branched are the difference.  Updated
> > patch:
> > http://asterisk.drunkcoder.com/patches/20090617__luafix__1.6.1.1.diff.txt
>
> The patch applied cleanly but Asterisk segfaulted on startup.  I did not
> see anything particularly suspicious in the console output - mostly
> warnings and errors about realtime database connectivity that we have
> not yet setup other than this message:
>
> [Jun 17 17:47:24] WARNING[30682]: loader.c:375 load_dynamic_module:
> Error loading module 'app_directory.so': /usr/lib64/libc-client.so.1:
> undefined symbol:
> mm_dlog
>
> As much as I'd like, I don't think I have the code knowledge and I don't
> have the time on this project to step through it with a debugger.  What
> next? Thanks - John

The error above has zilch to do with pbx_lua, but rather with the compilation
of app_directory against IMAP and the probable inability of your linker to
find the correct IMAP libraries.  You can prove this by adding 'noload =>
app_directory.so' to your modules.conf and restarting Asterisk.

-- 
Tilghman

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Re: [asterisk-users] snom mass deploy help

[Conrad Wood]

On Thu, 2009-06-18 at 14:21 +0200, Philipp Kempgen wrote:
> > On Jun 18, 2009, at 7:25 AM, Alex Samad  wrote:
> >> I am trying to setup asterisk to do a mass deploy of some snom  
> >> phones. I
> >> can't find where i configure asteriks to listen to the multicast
> >> address, nor where to set the notify reply.

FWIW I use a home-grown cgi script to configure the mass-deploy.
(attached)

Conrad


snom
Description: Perl program
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Re: [asterisk-users] Nagios under *

[Steve Edwards]

On Thu, 18 Jun 2009, Sriram wrote:

> I am trying to implement monitoring of asterisk (all 4 spans-i want to 
> show them line by line Up or down) using nagios using below script, but 
> i always get the status as down and red..can anyone let me know how to 
> read an output from nagios plugin ? nagios etc is configured already and 
> is working

I'm a 1.2 Luddite, but here's my $0.02.

> PATH=/bin:/sbin:/usr/bin:/usr/sbin
> FAILS=""
>   STATUS=$(asterisk -rnx "pri show span 1" | grep -a Status | awk '{print 
> $3;}' | cut -d, -f1)
>   if [ "${STATUS}" == "Up" ]; then
>  echo "PRI UP"
>  exit 0
>else
>  echo "PRI DOWN"
>  exit 2
>   fi

Does the user running Nagios have access to /var/run/asterisk.ctl?

Using sudo will work. You could change the ownership or permissions of 
asterisk.ctl, but using sudo should survive upgrades better and has less 
chance of breaking something else.

FAILS is never used.

You don't need grep or cut. Since (I'm assuming) this script will run 
frequently, cutting the number of processes created in half seems like a 
good idea.

STATUS=$(sudo asterisk -rnx "pri show span 1"\
 | awk '/Status/ {print $3}'\
 )

 if  [ "Up," == "${STATUS}" ]
 thenecho "PRI UP"
 exit 0
 elseecho "PRI DOWN"
 exit 2
 fi

Keep in mind that if the span has multiple D channels (like an NFAS group) 
you will get multiple statuses.

I'm not a big fan of using "-rx." I think AMI should be used for all new 
code.

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] Voicemail Password

[Darrin Henshaw]

As usual my manager comes up with some obscure reference I didn't find. There 
seems to be a parameter called minpassword described here:

http://www.asterisk.org/doxygen/trunk/Config_vm.html

But from further digging it looks like it's a 1.6.1.0 feature. Might see about 
a backport if possible.

Cheers,

Darrin Henshaw

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, June 18, 2009 15:19
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail Password

On Thu, 18 Jun 2009, Darrin Henshaw wrote:

> Does anyone know of a way to force the voicemail password for users to
> be of a certain length? We've setup operator=yes within our
> voicemail.conf and want to have the users use a long password to prevent
> possible guessing by external parties. I'm not seeing any such option in
> my research. If it doesn't exist it might be a decent feature. Thanks.

Sounds like a cool feature. I started looking into it, checking out
voicemail.conf (1.2) to get an idea of a good name to call the parameter
and I found this:

; If you need to have an external program, i.e. /usr/bin/myapp called when
; a voicemail password is changed, uncomment this:
externpass=/usr/bin/myapp

Who knew?

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

[James A. Shigley]

It errors the same whether I use g or G. 

 

James Shigley

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Totaro
Sent: Thursday, June 18, 2009 1:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

 

I don't feel like looking it up but does a capital G and lowercase g in
your DAHDI/group make a difference?

Just a thought.

On Thu, Jun 18, 2009 at 2:28 PM, James A. Shigley
 wrote:

I didn't have a limit set, but I put one on of 5 for testing sake that
didn't change a thing.

 

James Shigley

Monroe Telephone Answering Service

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Wednesday, June 17, 2009 2:55 PM


To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Subject: Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

 

Is your SIP call-limit set to 1?  That might explain the busy/congest
message.

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Wednesday, June 17, 2009 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

 

Never saw this appear on the list. So just resending it.

 

 

Alright I've been having an issue when trying to dial out locally when
coming from SIP. This used to work no problem, now it doesn't. Now the
local PRI to Bell Is working fine I have calls coming in and out of it
constantly right now. BUT if I try and make a local call from SIP (from
X-Lite or one of our Linksys SPA2102s) It fails every time with errors
like these

 

 

== Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Dial("SIP/test-b636a620",
"DAHDI/G3/9819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/9819213

-- Channel 0/23, span 3 got hangup, cause 50

-- Hungup 'DAHDI/71-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-b636a620' status is
'CHANUNAVAIL'

 

  == Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Dial("SIP/test-b6369010",
"DAHDI/G3/4099819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/4099819213

-- Channel 0/23, span 3 got hangup, cause 50

-- Hungup 'DAHDI/71-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-b6369010' status is
'CHANUNAVAIL'

 

== Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Set("SIP/test-09f23d18",
"CALLERID(name)=James Shigley") in new stack

-- Executing [9819...@from_test:2] Set("SIP/test-09f23d18",
"CALLERID(number)=4099819213") in new stack

-- Executing [9819...@from_test:3] Set("SIP/test-09f23d18",
"CALLERID(all)=James Shigley<4099819213>") in new stack

-- Executing [9819...@from_test:4] Dial("SIP/test-09f23d18",
"DAHDI/G3/9819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/9819213

-- Channel 0/22, span 3 got hangup, cause 50

-- Hungup 'DAHDI/70-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-09f23d18' status is
'CHANUNAVAIL'

 

Oh and sometimes it will also have this in the errors though no always

 

[Jun 17 10:52:53] WARNING[27549]: app_dial.c:842 wait_for_answer: Unable
to forward voice or dtmf 

 

On the second error above has the 409 added by the dialplan to see if
Bell wanted full 10 digits.

 

For the third I've tried a variety of ways of setting the CID thinking
maybe that was the issue this was just my most recent.

 

 

The odd thing is that I can send the call down one of my other PRI ports
to our Amtelco Infinity system. (via exten=>
9819213,1,Dial(${inf}/409${EXTEN}). I've tried everything I can think of
and googled for a good while trying to find an explanation for "got
hangup, cause 50". What is cause 50?

 

Sip Login information

 

[test]

username=test

type=friend

secret=X

callerid=

host=dynamic

nat=no

canreinvite=no

context=from_test

;codecs

disallow=all

allow=ulaw

 

Also had it as

 

[test]

username=test

type=friend

secret= X

callerid= "James Shigley" <4099819213>

host=dynamic

nat=no

canreinvite=no

context=from_test

;codecs

disallow=all

allow=ulaw

 

My From Context has changed several times here is some of the iterations
I've tried.

 

 

inf=DAHDI/g2  

bell=DAHDI/G3

 

[from_test] ; noted but not repaired.

exten=> _NXX,1,Dial(${belltd}/409${EXTEN})

exten=> 9819213,1,Dial(${inf}/409${EXTEN}

 

[from_test] ; noted but not repaired.

exten=> _NXX,1,Set(CALLERID(name)=James Shigley)

exten=> _NXX,2,Set(CALLE

Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

[Danny Nicholas]

G looks up 1,2,3,4,5, g looks up 5,4,3,2,1 so yes, at least in theory.  If
you only have one open line, no harm no foul.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
Sent: Thursday, June 18, 2009 1:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

 

I don't feel like looking it up but does a capital G and lowercase g in your
DAHDI/group make a difference?

Just a thought.

On Thu, Jun 18, 2009 at 2:28 PM, James A. Shigley 
wrote:

I didn't have a limit set, but I put one on of 5 for testing sake that
didn't change a thing.

 

James Shigley

Monroe Telephone Answering Service

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, June 17, 2009 2:55 PM


To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Subject: Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

 

Is your SIP call-limit set to 1?  That might explain the busy/congest
message.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Wednesday, June 17, 2009 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

 

Never saw this appear on the list. So just resending it.

 

 

Alright I've been having an issue when trying to dial out locally when
coming from SIP. This used to work no problem, now it doesn't. Now the local
PRI to Bell Is working fine I have calls coming in and out of it constantly
right now. BUT if I try and make a local call from SIP (from X-Lite or one
of our Linksys SPA2102s) It fails every time with errors like these

 

 

== Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Dial("SIP/test-b636a620",
"DAHDI/G3/9819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/9819213

-- Channel 0/23, span 3 got hangup, cause 50

-- Hungup 'DAHDI/71-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-b636a620' status is 'CHANUNAVAIL'

 

  == Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Dial("SIP/test-b6369010",
"DAHDI/G3/4099819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/4099819213

-- Channel 0/23, span 3 got hangup, cause 50

-- Hungup 'DAHDI/71-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-b6369010' status is 'CHANUNAVAIL'

 

== Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Set("SIP/test-09f23d18",
"CALLERID(name)=James Shigley") in new stack

-- Executing [9819...@from_test:2] Set("SIP/test-09f23d18",
"CALLERID(number)=4099819213") in new stack

-- Executing [9819...@from_test:3] Set("SIP/test-09f23d18",
"CALLERID(all)=James Shigley<4099819213>") in new stack

-- Executing [9819...@from_test:4] Dial("SIP/test-09f23d18",
"DAHDI/G3/9819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/9819213

-- Channel 0/22, span 3 got hangup, cause 50

-- Hungup 'DAHDI/70-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-09f23d18' status is 'CHANUNAVAIL'

 

Oh and sometimes it will also have this in the errors though no always

 

[Jun 17 10:52:53] WARNING[27549]: app_dial.c:842 wait_for_answer: Unable to
forward voice or dtmf 

 

On the second error above has the 409 added by the dialplan to see if Bell
wanted full 10 digits.

 

For the third I've tried a variety of ways of setting the CID thinking maybe
that was the issue this was just my most recent.

 

 

The odd thing is that I can send the call down one of my other PRI ports to
our Amtelco Infinity system. (via exten=>
9819213,1,Dial(${inf}/409${EXTEN}). I've tried everything I can think of and
googled for a good while trying to find an explanation for "got hangup,
cause 50". What is cause 50?

 

Sip Login information

 

[test]

username=test

type=friend

secret=X

callerid=

host=dynamic

nat=no

canreinvite=no

context=from_test

;codecs

disallow=all

allow=ulaw

 

Also had it as

 

[test]

username=test

type=friend

secret= X

callerid= "James Shigley" <4099819213>

host=dynamic

nat=no

canreinvite=no

context=from_test

;codecs

disallow=all

allow=ulaw

 

My From Context has changed several times here is some of the iterations
I've tried.

 

 

inf=DAHDI/g2  

bell=DAHDI/G3

 

[from_test] ; noted but not repaired.

exten=> _NXX,1,Dial(${belltd}/409${EXTEN})

exten=> 9819213,1,Dial(${inf}/409${EXTEN}

 

[from_test] ; noted but not repaired.

exten=> _NXX,1,Set(CALLERID(n

Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

[Steve Totaro]

I don't feel like looking it up but does a capital G and lowercase g in your
DAHDI/group make a difference?

Just a thought.

On Thu, Jun 18, 2009 at 2:28 PM, James A. Shigley wrote:

>  I didn’t have a limit set, but I put one on of 5 for testing sake that
> didn’t change a thing.
>
>
>
> James Shigley
>
> *Monroe Telephone Answering Service*
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas
> *Sent:* Wednesday, June 17, 2009 2:55 PM
> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
> *Subject:* Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue
>
>
>
> Is your SIP call-limit set to 1?  That might explain the busy/congest
> message.
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *James A. Shigley
> *Sent:* Wednesday, June 17, 2009 2:59 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue
>
>
>
> Never saw this appear on the list. So just resending it.
>
>
>
>
>
> Alright I’ve been having an issue when trying to dial out locally when
> coming from SIP. This used to work no problem, now it doesn’t. Now the local
> PRI to Bell Is working fine I have calls coming in and out of it constantly
> right now. BUT if I try and make a local call from SIP (from X-Lite or one
> of our Linksys SPA2102s) It fails every time with errors like these
>
>
>
>
>
> == Using SIP RTP CoS mark 5
>
> -- Executing [9819...@from_test:1] Dial("SIP/test-b636a620",
> "DAHDI/G3/9819213") in new stack
>
> -- Requested transfer capability: 0x00 - SPEECH
>
> -- Called G3/9819213
>
> -- Channel 0/23, span 3 got hangup, cause 50
>
> -- Hungup 'DAHDI/71-1'
>
>   == Everyone is busy/congested at this time (1:0/0/1)
>
> -- Auto fallthrough, channel 'SIP/test-b636a620' status is
> 'CHANUNAVAIL'
>
>
>
>   == Using SIP RTP CoS mark 5
>
> -- Executing [9819...@from_test:1] Dial("SIP/test-b6369010",
> "DAHDI/G3/4099819213") in new stack
>
> -- Requested transfer capability: 0x00 - SPEECH
>
> -- Called G3/4099819213
>
> -- Channel 0/23, span 3 got hangup, cause 50
>
> -- Hungup 'DAHDI/71-1'
>
>   == Everyone is busy/congested at this time (1:0/0/1)
>
> -- Auto fallthrough, channel 'SIP/test-b6369010' status is
> 'CHANUNAVAIL'
>
>
>
> == Using SIP RTP CoS mark 5
>
> -- Executing [9819...@from_test:1] Set("SIP/test-09f23d18",
> "CALLERID(name)=James Shigley") in new stack
>
> -- Executing [9819...@from_test:2] Set("SIP/test-09f23d18",
> "CALLERID(number)=4099819213") in new stack
>
> -- Executing [9819...@from_test:3] Set("SIP/test-09f23d18",
> "CALLERID(all)=James Shigley<4099819213>") in new stack
>
> -- Executing [9819...@from_test:4] Dial("SIP/test-09f23d18",
> "DAHDI/G3/9819213") in new stack
>
> -- Requested transfer capability: 0x00 - SPEECH
>
> -- Called G3/9819213
>
> -- Channel 0/22, span 3 got hangup, cause 50
>
> -- Hungup 'DAHDI/70-1'
>
>   == Everyone is busy/congested at this time (1:0/0/1)
>
> -- Auto fallthrough, channel 'SIP/test-09f23d18' status is
> 'CHANUNAVAIL'
>
>
>
> Oh and sometimes it will also have this in the errors though no always
>
>
>
> [Jun 17 10:52:53] WARNING[27549]: app_dial.c:842 wait_for_answer: Unable to
> forward voice or dtmf
>
>
>
> On the second error above has the 409 added by the dialplan to see if Bell
> wanted full 10 digits.
>
>
>
> For the third I’ve tried a variety of ways of setting the CID thinking
> maybe that was the issue this was just my most recent.
>
>
>
>
>
> The odd thing is that I can send the call down one of my other PRI ports to
> our Amtelco Infinity system. (via exten=>
> 9819213,1,Dial(${inf}/409${EXTEN}). I’ve tried everything I can think of
> and googled for a good while trying to find an explanation for “got hangup,
> cause 50”. What is cause 50?
>
>
>
> Sip Login information
>
>
>
> [test]
>
> username=test
>
> type=friend
>
> secret=X
>
> callerid=
>
> host=dynamic
>
> nat=no
>
> canreinvite=no
>
> context=from_test
>
> ;codecs
>
> disallow=all
>
> allow=ulaw
>
>
>
> Also had it as
>
>
>
> [test]
>
> username=test
>
> type=friend
>
> secret= X
>
> callerid= "James Shigley" <4099819213>
>
> host=dynamic
>
> nat=no
>
> canreinvite=no
>
> context=from_test
>
> ;codecs
>
> disallow=all
>
> allow=ulaw
>
>
>
> My From Context has changed several times here is some of the iterations
> I’ve tried.
>
>
>
>
>
> inf=DAHDI/g2
>
> bell=DAHDI/G3
>
>
>
> [from_test] ; noted but not repaired.
>
> exten=> _NXX,1,Dial(${belltd}/409${EXTEN})
>
> exten=> 9819213,1,Dial(${inf}/409${EXTEN}
>
>
>
> [from_test] ; noted but not repaired.
>
> exten=> _NXX,1,Set(CALLERID(name)=James Shigley)
>
> exten=> _NXX,2,Set(CALLERID(number)=4099819213)
>
> exten=> _NXX,3,Set(CALLERID(all)=${CALLERID(name)}

Re: [asterisk-users] asterisk-gui: read/write in the conf files or db

[bilal ghayyad]

Hi Danny;

Really I did not understand how I can determine if the IO will be DB or conf 
files? Is it from the Asterisk manager?

Regards
Bilal

-

It depends on how you are configured.  The gui interfaces using Asterisk
Manager, so you get the Same IO from the gui that you would get from a
native manager session.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Wednesday, June 17, 2009 5:36 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk-gui: read/write in the conf files or db?


Hi All;

asterisk-gui read/write from the conf files or database?

Any advise?
Regards
Bilal




  


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Re: [asterisk-users] asterisk-gui: read/write in the conf files or db

[Steve Totaro]

Why not connect to the AMI via telnet?

On Thu, Jun 18, 2009 at 2:27 PM, bilal ghayyad  wrote:

>
> Hi Danny;
>
> Really I did not understand how I can determine if the IO will be DB or
> conf files? Is it from the Asterisk manager?
>
> Regards
> Bilal
>
> -
>
> It depends on how you are configured.  The gui interfaces using Asterisk
> Manager, so you get the Same IO from the gui that you would get from a
> native manager session.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal
> ghayyad
> Sent: Wednesday, June 17, 2009 5:36 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] asterisk-gui: read/write in the conf files or db?
>
>
> Hi All;
>
> asterisk-gui read/write from the conf files or database?
>
> Any advise?
> Regards
> Bilal
>
>
>
>
>
>
>
> ___
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>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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[asterisk-users] SOLVED: Re: dahdi and overlapdial problem

[Bjoern Metzdorf]

Hi,

after further investigation we found a solution:

overlapdial=incoming

See also https://issues.asterisk.org/view.php?id=7511

Regards,
Bjoern

Bjoern Metzdorf wrote:
> Forgot to add:
> 
> Asterisk full log only shows no anomalies. Normal call clearing when you 
> hangup, nothing else.
> 
> Regards
> 
> Bjoern Metzdorf wrote:
>> Hi there,
>>
>> we have a problem with dahdi and overlapdial. We are running an E1 in 
>> Germany and are in need of overlapdial. The E1 is connected to a Sangoma 
>> A101.
>>
>> As soon as overlapdial is set to "yes" we have problems with incoming 
>> audio on the dahdi channels. When set to "no" all audio is fine.
>>
>> Basically we can choose between being able to receive calls or to place 
>> calls :)
>>
>> Details:
>>
>> Software:
>> asterisk-1.6.1.0.tar.gz
>> libpri-1.4.9.tar.gz
>> dahdi-linux-2.1.0.4.tar.gz
>> dahdi-tools-2.1.0.2.tar.gz
>> wanpipe-3.4.1.tgz
>>
>> chan_dadhi.conf:
>> [trunkgroups]
>> [channels]
>> context=from-pstn
>> usecallerid=yes
>> hidecallerid=no
>> callwaiting=yes
>> usecallingpres=yes
>> callwaitingcallerid=yes
>> threewaycalling=yes
>> transfer=yes
>> canpark=yes
>> cancallforward=yes
>> callreturn=yes
>> echocancel=yes
>> echocancelwhenbridged=yes
>> relaxdtmf=yes
>> rxgain=0.0
>> txgain=0.0
>> group=1
>> callgroup=1
>> pickupgroup=1
>> immediate=no
>> internationalprefix=00
>> nationalprefix=0
>> overlapdial=no
>> ;Sangoma A101 port 1 [slot:5 bus:3 span:1] 
>> switchtype=euroisdn
>> echocancel=yes
>> signalling=pri_cpe
>> channel =>1-15,17-31
>>
>> /etc/dahdi/system.conf:
>> # Span 1: WPE1/0 "wanpipe1 card 0" (MASTER) HDB3/CCS/CRC4
>> span=1,1,0,ccs,hdb3,crc4
>> # termtype: te
>> bchan=1-15,17-31
>> echocanceller=mg2,1-15,17-31
>> hardhdlc=16
>> # Global data
>> loadzone= de
>> defaultzone = de
>>
>> Any hints?
>>
>> Thanks!
>>
>> Regards,
>> Bjoern
>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
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Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

[Steve Totaro]

I am calling CHARLOT,DANIEL @ 4099819921 to tell him that you posted his
name and phone number on the interweb

Sometimes a redaction is prudent.

On Thu, Jun 18, 2009 at 2:24 PM, James A. Shigley wrote:

> !! Got reject for frame 61, but we have nothing -- resetting!
> !! Got reject for frame 63, but we have nothing -- resetting!
> !! Got reject for frame 65, but we have nothing -- resetting!
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate
> Connect Request
> q931.c:3009 q931_disconnect: call 60185 on channel 3 enters state 11
> (Disconnect Request)
> > Protocol Discriminator: Q.931 (8)  len=9
> > Call Ref: len= 2 (reference 27417/0x6B19) (Originator)
> > Message type: DISCONNECT (69)
> > [08 02 81 90]
> > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
> Location: Private network serving the local user (1)
> >  Ext: 1  Cause: Normal Clearing (16), class = Normal
> Event (1) ]
> < Protocol Discriminator: Q.931 (8)  len=5
> < Call Ref: len= 2 (reference 27417/0x6B19) (Terminator)
> < Message type: RELEASE (77)
> q931.c:3795 q931_receive: call 60185 on channel 3 enters state 0 (Null)
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release
> Request
> > Protocol Discriminator: Q.931 (8)  len=9
> > Call Ref: len= 2 (reference 27417/0x6B19) (Originator)
> > Message type: RELEASE COMPLETE (90)
> > [08 02 81 90]
> > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
> Location: Private network serving the local user (1)
> >  Ext: 1  Cause: Normal Clearing (16), class = Normal
> Event (1) ]
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
> NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
> -- Making new call for cr 60186
> > Protocol Discriminator: Q.931 (8)  len=101
> > Call Ref: len= 2 (reference 27418/0x6B1A) (Originator)
> > Message type: SETUP (5)
> > [04 03 80 90 a2]
> > Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
> capability: Speech (0)
> >  Ext: 1  Trans mode/rate: 64kbps,
> circuit-mode (16)
> >User information layer 1: u-Law (34)
> > [18 03 a1 83 82]
> > Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0
> Preferred  Dchan: 0
> >ChanSel: As indicated in following octets
> >   Ext: 1  Coding: 0  Number Specified  Channel
> Type: 3
> >   Ext: 1  Channel: 2 ]
> > [1c 15 9f 8b 01 00 a1 0f 02 01 fb 06 07 2a 86 48 ce 15 00 04 0a 01 00]
> > Facility (len=23, codeset=0) [ 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x0F,
> 0x02, 0x01, 0xFB, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x04,
> 0x0A, 0x01, 0x00 ]
> PROTOCOL 1F
> 8B 0001 00 (CONTEXT SPECIFIC [11])
> A1 000F (CONTEXT SPECIFIC [1])
>  02 0001 FB (INTEGER: 251)
>  06 0007 2A 86 48 CE 15 00 04 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 04)
>  0A 0001 00 (ENUMERATED: 0)
> > [1e 02 80 83]
> > Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)
> 0: 0  Location: User (0)
> >   Ext: 1  Progress Description: Calling
> equipment is non-ISDN. (3) ]
> > [28 0f b1 43 48 41 52 4c 4f 54 2c 44 41 4e 49 45 4c]
> > Display (len=15) Charset: 31 [ CHARLOT,DANIEL ]
> > [6c 0c 21 83 34 30 39 35 34 33 33 31 31 30]
> > Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI:
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> >   Presentation: Presentation allowed of
> network provided number (3)  '4095433110' ]
> > [70 0b a1 34 30 39 39 38 31 39 39 32 31]
> > Called Number (len=13) [ Ext: 1  TON: National Number (2)  NPI:
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)  '4099819921' ]
> > [74 0d 21 01 8f 34 30 39 39 38 35 38 38 36 35]
> > Redirecting Number (len=15) [ Ext: 0  TON: National Number (2)  NPI:
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> >   Ext: 0  Presentation: Presentation
> permitted, user number passed network screening (1)
> >   Ext: 1  Reason: Forwarded
> unconditionally (15)  '4099858865' ]
> q931.c:3128 q931_setup: call 60186 on channel 2 enters state 1 (Call
> Initiated)
> < Protocol Discriminator: Q.931 (8)  len=11
> < Call Ref: len= 2 (reference 27418/0x6B1A) (Terminator)
> < Message type: CALL PROCEEDING (2)
> < [18 04 e9 80 83 82]
> < Channel ID (len= 6) [ Ext: 1  IntID: Explicit  PRI  Spare: 0
> Exclusive  Dchan: 0
>  <   Ext: 1  DS1 Identifier: 0
> <   Ext: 1  Coding: 0  Number Specified  Channel
> Type: 3
> <   Ext: 1  Channel: 2 ]
> -- Processing IE 24 (cs0, Channel Identification)
> q931.c:3677 q931_receive: call 60186 on channel 2 enters state 3
> (Outgoing call  Proceeding)
> > Protocol Discriminator: Q.931 (8)  len=35
> > Call Ref: len= 2 (reference 27418/0x6B1A) (Originator)
> > Mes

Re: [asterisk-users] Scaling

[Steve Totaro]

On Wed, Jun 17, 2009 at 7:41 PM, Steve Totaro <
stot...@totarotechnologies.com> wrote:

>
>
> On Wed, Jun 17, 2009 at 3:18 PM, John Todd  wrote:
>
>>
>> On Jun 17, 2009, at 8:16 AM, Steve Totaro wrote:
>>
>> > Hi,
>> >
>> > Quick question to the real world.
>> >
>> > Approx what specs would I need on server to handle 95 ZAP or Dahdi -
>> > > SIP gateway using G729 on the SIP to carrier side (nothing else,
>> > just media conversion)?
>> >
>> > Does the latest Asterisk/DAHDI significantly improve these numbers
>> > over say, Asterisk 1.2.X?
>> >
>> > Sure, there is plenty to read but nothing I could find quickly on my
>> > exact needs that was clear and I want to be fairly sure before
>> > ordering a server.
>> >
>> > Obviously load avg has something to do with it but CPU and mem seems
>> > to be the biggest factors.
>> >
>> > --
>> > Thanks,
>> > Steve Totaro
>> > +18887771888 (Toll Free)
>> > +12409381212 (Cell)
>> > +12024369784 (Skype)
>>
>>
>> [Digium hat off, ITSP (previous employer) hat on.]
>>
>> Not speaking for Digium on this one, but speaking from personal
>> experience at another company.
>>
>> We could get 100 G.729 channels in a 2x3ghz P4 machine with plenty of
>> CPU room to spare, 4+ years ago, and that rule of thumb served us
>> well.  This was for SIP (G.711<->G.729) but I can't imagine that DAHDI
>> or Asterisk takes significantly more or less horsepower for this task
>> now for the base transcoding load.
>>
>> JT
>>
>> ---
>> John Todd   
>> email:jt...@digium.com
>> Digium, Inc. | Asterisk Open Source Community Director
>> 445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
>> direct: +1-256-428-6083 http://www.digium.com/
>>
>>
>>
>>
> John,
>
> Thanks for the info.
>
> Can't go wrong with these in a cluster if you are correct.
> http://www.surpluscomputers.com/348663/hp-dl140-proliant-dual-xeon.html
>
> I used to have a garden of seven (too small to be a farm) of Asterisk boxen
> that were simple PRI to SIP gateways doing ulaw (no transcoding as far as
> codec), and nothing else running, just a few lines in each conf file.
>
> They would run at ~65% CPU utilization with 95 channels in use on each box.
>
> This was the Asterisk, Zaptel, Libpri 1.2.X flavor.
>
> I wonder if anyone else has input on the CPU utilization of of a simple
> PSTN <-> VoIP gateway based on Asterisk?
>
>
I wonder how much of my observed 65% CPU utilization was because of software
echo cancellation?

If someone knows, I would appreciate any input.  If not, I will follow-up
with my own testing in this thread.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] Voicemail Password

[Steve Edwards]

On Thu, 18 Jun 2009, Darrin Henshaw wrote:

> Does anyone know of a way to force the voicemail password for users to 
> be of a certain length? We've setup operator=yes within our 
> voicemail.conf and want to have the users use a long password to prevent 
> possible guessing by external parties. I'm not seeing any such option in 
> my research. If it doesn't exist it might be a decent feature. Thanks.

Sounds like a cool feature. I started looking into it, checking out 
voicemail.conf (1.2) to get an idea of a good name to call the parameter 
and I found this:

; If you need to have an external program, i.e. /usr/bin/myapp called when
; a voicemail password is changed, uncomment this:
externpass=/usr/bin/myapp

Who knew?

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] Nagios under *

[Diego Aguirre (DagMoller)]

Check the script permissions for nagios user

Sriram escreveu:
>  
> Hi
> I am trying to implement monitoring of asterisk (all 4 spans-i want to
> show them line by line Up or down) using nagios using below script, but
> i always get the status as down and red..can anyone let me know how to
> read an output from nagios plugin ? nagios etc is configured already and
> is working
> PATH=/bin:/sbin:/usr/bin:/usr/sbin
> FAILS=""
>  
>STATUS=$(asterisk -rnx "pri show span 1" | grep -a Status | awk
> '{print $3;}' | cut -d, -f1)
>if [ "${STATUS}" == "Up" ]; then
>   echo "PRI UP"
>   exit 0
> else
>   echo "PRI DOWN"
>   exit 2
>fi
>  
> if i execute the script from command line i get the correct output i.e
> OK for span 1 but on nagios web interface i get it as down...
> If anyone can share the above script for asterisk monitoring then i wud
> be grateful
>  
> rgds
> Sriram
> 
> 
> 
> 
> ___
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> 
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Diego Aguirre (DagMoller)
Infodag Consultoria
FWD#: 459696
Enum#: +55 21 8871-4916 (e164.org)
DUNDi-br#: 21 8871-4916

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Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

[James A. Shigley]

I didn't have a limit set, but I put one on of 5 for testing sake that
didn't change a thing.

 

James Shigley

Monroe Telephone Answering Service

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Wednesday, June 17, 2009 2:55 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

 

Is your SIP call-limit set to 1?  That might explain the busy/congest
message.

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Wednesday, June 17, 2009 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

 

Never saw this appear on the list. So just resending it.

 

 

Alright I've been having an issue when trying to dial out locally when
coming from SIP. This used to work no problem, now it doesn't. Now the
local PRI to Bell Is working fine I have calls coming in and out of it
constantly right now. BUT if I try and make a local call from SIP (from
X-Lite or one of our Linksys SPA2102s) It fails every time with errors
like these

 

 

== Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Dial("SIP/test-b636a620",
"DAHDI/G3/9819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/9819213

-- Channel 0/23, span 3 got hangup, cause 50

-- Hungup 'DAHDI/71-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-b636a620' status is
'CHANUNAVAIL'

 

  == Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Dial("SIP/test-b6369010",
"DAHDI/G3/4099819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/4099819213

-- Channel 0/23, span 3 got hangup, cause 50

-- Hungup 'DAHDI/71-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-b6369010' status is
'CHANUNAVAIL'

 

== Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Set("SIP/test-09f23d18",
"CALLERID(name)=James Shigley") in new stack

-- Executing [9819...@from_test:2] Set("SIP/test-09f23d18",
"CALLERID(number)=4099819213") in new stack

-- Executing [9819...@from_test:3] Set("SIP/test-09f23d18",
"CALLERID(all)=James Shigley<4099819213>") in new stack

-- Executing [9819...@from_test:4] Dial("SIP/test-09f23d18",
"DAHDI/G3/9819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/9819213

-- Channel 0/22, span 3 got hangup, cause 50

-- Hungup 'DAHDI/70-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-09f23d18' status is
'CHANUNAVAIL'

 

Oh and sometimes it will also have this in the errors though no always

 

[Jun 17 10:52:53] WARNING[27549]: app_dial.c:842 wait_for_answer: Unable
to forward voice or dtmf 

 

On the second error above has the 409 added by the dialplan to see if
Bell wanted full 10 digits.

 

For the third I've tried a variety of ways of setting the CID thinking
maybe that was the issue this was just my most recent.

 

 

The odd thing is that I can send the call down one of my other PRI ports
to our Amtelco Infinity system. (via exten=>
9819213,1,Dial(${inf}/409${EXTEN}). I've tried everything I can think of
and googled for a good while trying to find an explanation for "got
hangup, cause 50". What is cause 50?

 

Sip Login information

 

[test]

username=test

type=friend

secret=X

callerid=

host=dynamic

nat=no

canreinvite=no

context=from_test

;codecs

disallow=all

allow=ulaw

 

Also had it as

 

[test]

username=test

type=friend

secret= X

callerid= "James Shigley" <4099819213>

host=dynamic

nat=no

canreinvite=no

context=from_test

;codecs

disallow=all

allow=ulaw

 

My From Context has changed several times here is some of the iterations
I've tried.

 

 

inf=DAHDI/g2  

bell=DAHDI/G3

 

[from_test] ; noted but not repaired.

exten=> _NXX,1,Dial(${belltd}/409${EXTEN})

exten=> 9819213,1,Dial(${inf}/409${EXTEN}

 

[from_test] ; noted but not repaired.

exten=> _NXX,1,Set(CALLERID(name)=James Shigley)

exten=> _NXX,2,Set(CALLERID(number)=4099819213)

exten=>
_NXX,3,Set(CALLERID(all)=${CALLERID(name)}<${CALLERID(num)}>)

exten=> _NXX,4,Dial(${bell}/${EXTEN})

 

[from_test] ; noted but not repaired.

exten=> _NXX,1,Set(CALLERID(name)=James Shigley)

exten=> _NXX,2,Set(CALLERID(number)=4099819213)

exten=> _NXX,3,Dial(${bell}/${EXTEN})

 

 

Note I didn't include the full context only the lines relevant to local
dialing. LD dialing which is sent out sip works just fine. Also I tried
using g3 instead of G3 thinking maybe there was an issue wi

Re: [asterisk-users] Speex problem installing on CentOS 5.3

[John A. Sullivan III]

On Thu, 2009-06-18 at 13:17 -0400, Steve Totaro wrote:
> 
> 
> On Thu, Jun 18, 2009 at 12:46 PM, Brent Davidson
>  wrote:
> John A. Sullivan III wrote:
> 
> > Hello, all.  I am delightfully slogging my way through installing 
> and
> > configuring Asterisk 1.6.1.1 on CentOS 5.3.  I'm learning lots and
> > admiring the product but I'm having a problem getting speex to 
> install
> > and I would very much like to use it.  It is not available in 
> menuselect
> > and the problem appears to be with speex_preprocess_ctl:
> > 
> > [r...@pbx01 asterisk-1.6.1.1]# grep -i speex config.log
> > configure:43813: checking for speex_encode in -lspeex
> > configure:43848: gcc -o conftest -g -O2   conftest.c -lspeex  -lm  
> >&5
> > configure:43906: checking speex/speex.h usability
> > configure:43947: checking speex/speex.h presence
> > configure:44015: checking for speex/speex.h
> > configure:44076: checking for speex_preprocess_ctl in -lspeex
> > configure:44111: gcc -o conftest -g -O2   conftest.c -lspeex  -lm  
> >&5
> > /home/compuser/Asterisk/asterisk-1.6.1.1/conftest.c:306: undefined
> > reference to `speex_preprocess_ctl'
> > | #define HAVE_SPEEX 1
> > | #define HAVE_SPEEX_VERSION
> > | char speex_preprocess_ctl ();
> > | return speex_preprocess_ctl ();
> > configure:44341: checking for speex_preprocess_ctl in -lspeexdsp
> > configure:44376: gcc -o conftest -g -O2   conftest.c -lspeexdsp  -lm
> >   
> > > &5
> > > 
> > /usr/bin/ld: cannot find -lspeexdsp
> > | #define HAVE_SPEEX 1
> > | #define HAVE_SPEEX_VERSION
> > | char speex_preprocess_ctl ();
> > | return speex_preprocess_ctl ();
> > 
> > Internet searches have only further confused the issue for me.  It 
> seems
> > this is part of libspeex which in the RedHat world is provided by 
> the
> > speex-devel package (which I have installed):
> > 
> > [r...@pbx01 ~]# rpm -qa | grep speex
> > speex-devel-1.0.5-4.el5_1.1
> > speex-1.0.5-4.el5_1.1
> > 
> > What is the magic to make speex available to Asterisk on CentOS 
> 5.3? Or
> > am I stuck having to uninstall the speex packages and install speex 
> from
> > source?  Thanks - John
> > 
> >   
> 
> 
> I ended up having to install from source.  There are
> apparently bits of speex that are not included in the RPM's.
> It's a farily simple install though.
> 
> Good luck,
> -Brent
> 
> 
> 
> I am curious if a "yum -y install speex*" would have worked for you?
> I will give it a try on my next 5.3 box.

Alas not.  That is, in effect, what I first tried.  From what I can
tell, it is against RedHat policy to issue RPMs for the newer versions
(I'm guessing because it is still rc rather than ga).  I just installed
from source and, once I remembered to add /usr/local/lib to ldconfig, it
all config'd, compiled, and loaded just fine.  Thanks, all
> 
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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[asterisk-users] Nagios under *

[Sriram]

Hi 
I am trying to implement monitoring of asterisk (all 4 spans-i want to show 
them line by line Up or down) using nagios using below script, but i always get 
the status as down and red..can anyone let me know how to read an output from 
nagios plugin ? nagios etc is configured already and is working 
PATH=/bin:/sbin:/usr/bin:/usr/sbin
FAILS=""

   STATUS=$(asterisk -rnx "pri show span 1" | grep -a Status | awk '{print 
$3;}' | cut -d, -f1)
   if [ "${STATUS}" == "Up" ]; then
  echo "PRI UP"
  exit 0
else
  echo "PRI DOWN"
  exit 2
   fi

if i execute the script from command line i get the correct output i.e OK for 
span 1 but on nagios web interface i get it as down...
If anyone can share the above script for asterisk monitoring then i wud be 
grateful

rgds
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Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

[James A. Shigley]

!! Got reject for frame 61, but we have nothing -- resetting!
!! Got reject for frame 63, but we have nothing -- resetting!
!! Got reject for frame 65, but we have nothing -- resetting!
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate
Connect Request
q931.c:3009 q931_disconnect: call 60185 on channel 3 enters state 11
(Disconnect Request)
> Protocol Discriminator: Q.931 (8)  len=9
> Call Ref: len= 2 (reference 27417/0x6B19) (Originator)
> Message type: DISCONNECT (69)
> [08 02 81 90]
> Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
Location: Private network serving the local user (1)
>  Ext: 1  Cause: Normal Clearing (16), class = Normal
Event (1) ]
< Protocol Discriminator: Q.931 (8)  len=5
< Call Ref: len= 2 (reference 27417/0x6B19) (Terminator)
< Message type: RELEASE (77)
q931.c:3795 q931_receive: call 60185 on channel 3 enters state 0 (Null)
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release
Request
> Protocol Discriminator: Q.931 (8)  len=9
> Call Ref: len= 2 (reference 27417/0x6B19) (Originator)
> Message type: RELEASE COMPLETE (90)
> [08 02 81 90]
> Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
Location: Private network serving the local user (1)
>  Ext: 1  Cause: Normal Clearing (16), class = Normal
Event (1) ]
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
-- Making new call for cr 60186
> Protocol Discriminator: Q.931 (8)  len=101
> Call Ref: len= 2 (reference 27418/0x6B1A) (Originator)
> Message type: SETUP (5)
> [04 03 80 90 a2]
> Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: Speech (0)
>  Ext: 1  Trans mode/rate: 64kbps,
circuit-mode (16)
>User information layer 1: u-Law (34)
> [18 03 a1 83 82]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0
Preferred  Dchan: 0
>ChanSel: As indicated in following octets
>   Ext: 1  Coding: 0  Number Specified  Channel
Type: 3
>   Ext: 1  Channel: 2 ]
> [1c 15 9f 8b 01 00 a1 0f 02 01 fb 06 07 2a 86 48 ce 15 00 04 0a 01 00]
> Facility (len=23, codeset=0) [ 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x0F,
0x02, 0x01, 0xFB, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x04,
0x0A, 0x01, 0x00 ]
PROTOCOL 1F
8B 0001 00 (CONTEXT SPECIFIC [11])
A1 000F (CONTEXT SPECIFIC [1])
  02 0001 FB (INTEGER: 251)
  06 0007 2A 86 48 CE 15 00 04 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 04)
  0A 0001 00 (ENUMERATED: 0)
> [1e 02 80 83]
> Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)
0: 0  Location: User (0)
>   Ext: 1  Progress Description: Calling
equipment is non-ISDN. (3) ]
> [28 0f b1 43 48 41 52 4c 4f 54 2c 44 41 4e 49 45 4c]
> Display (len=15) Charset: 31 [ CHARLOT,DANIEL ]
> [6c 0c 21 83 34 30 39 35 34 33 33 31 31 30]
> Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>   Presentation: Presentation allowed of
network provided number (3)  '4095433110' ]
> [70 0b a1 34 30 39 39 38 31 39 39 32 31]
> Called Number (len=13) [ Ext: 1  TON: National Number (2)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)  '4099819921' ]
> [74 0d 21 01 8f 34 30 39 39 38 35 38 38 36 35]
> Redirecting Number (len=15) [ Ext: 0  TON: National Number (2)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>   Ext: 0  Presentation: Presentation
permitted, user number passed network screening (1)
>   Ext: 1  Reason: Forwarded
unconditionally (15)  '4099858865' ]
q931.c:3128 q931_setup: call 60186 on channel 2 enters state 1 (Call
Initiated)
< Protocol Discriminator: Q.931 (8)  len=11
< Call Ref: len= 2 (reference 27418/0x6B1A) (Terminator)
< Message type: CALL PROCEEDING (2)
< [18 04 e9 80 83 82]
< Channel ID (len= 6) [ Ext: 1  IntID: Explicit  PRI  Spare: 0
Exclusive  Dchan: 0
 Protocol Discriminator: Q.931 (8)  len=35
> Call Ref: len= 2 (reference 27418/0x6B1A) (Originator)
> Message type: FACILITY (98)
> [1c 1c 9f 8b 01 00 a1 16 02 01 fc 02 01 00 80 0e 43 48 41 52 4c 4f 54
2c 44 41 4e 49 45 4c]
> Facility (len=30, codeset=0) [ 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x16,
0x02, 0x01, 0xFC, 0x02, 0x01, 0x00, 0x80, 0x0E, 'CHARLOT,DANIEL' ]
PROTOCOL 1F
8B 0001 00 (CONTEXT SPECIFIC [11])
A1 0016 (CONTEXT SPECIFIC [1])
  02 0001 FC (INTEGER: 252)
  02 0001 00 (INTEGER: 0)
  80 000E 43 48

Re: [asterisk-users] dahdi and overlapdial problem

[Bjoern Metzdorf]

Forgot to add:

Asterisk full log only shows no anomalies. Normal call clearing when you 
hangup, nothing else.

Regards

Bjoern Metzdorf wrote:
> Hi there,
> 
> we have a problem with dahdi and overlapdial. We are running an E1 in 
> Germany and are in need of overlapdial. The E1 is connected to a Sangoma 
> A101.
> 
> As soon as overlapdial is set to "yes" we have problems with incoming 
> audio on the dahdi channels. When set to "no" all audio is fine.
> 
> Basically we can choose between being able to receive calls or to place 
> calls :)
> 
> Details:
> 
> Software:
> asterisk-1.6.1.0.tar.gz
> libpri-1.4.9.tar.gz
> dahdi-linux-2.1.0.4.tar.gz
> dahdi-tools-2.1.0.2.tar.gz
> wanpipe-3.4.1.tgz
> 
> chan_dadhi.conf:
> [trunkgroups]
> [channels]
> context=from-pstn
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> relaxdtmf=yes
> rxgain=0.0
> txgain=0.0
> group=1
> callgroup=1
> pickupgroup=1
> immediate=no
> internationalprefix=00
> nationalprefix=0
> overlapdial=no
> ;Sangoma A101 port 1 [slot:5 bus:3 span:1] 
> switchtype=euroisdn
> echocancel=yes
> signalling=pri_cpe
> channel =>1-15,17-31
> 
> /etc/dahdi/system.conf:
> # Span 1: WPE1/0 "wanpipe1 card 0" (MASTER) HDB3/CCS/CRC4
> span=1,1,0,ccs,hdb3,crc4
> # termtype: te
> bchan=1-15,17-31
> echocanceller=mg2,1-15,17-31
> hardhdlc=16
> # Global data
> loadzone= de
> defaultzone = de
> 
> Any hints?
> 
> Thanks!
> 
> Regards,
> Bjoern
> 
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Re: [asterisk-users] Voicemail Password

[Danny Nicholas]

AFAIK, this doesn't exist.  However,  you could disable password changing in
the voicemail application and set it from the dialplan and force a minimum
length there.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrin Henshaw
Sent: Thursday, June 18, 2009 12:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Voicemail Password

 

Does anyone know of a way to force the voicemail password for users to be of
a certain length? We've setup operator=yes within our voicemail.conf and
want to have the users use a long password to prevent possible guessing by
external parties. I'm not seeing any such option in my research. If it
doesn't exist it might be a decent feature. Thanks.

 

Running:  1.4.25, on CentOS 4.7

 

Cheers,

 

Darrin Henshaw

 

  _  

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[asterisk-users] Voicemail Password

[Darrin Henshaw]

Does anyone know of a way to force the voicemail password for users to be of a 
certain length? We've setup operator=yes within our voicemail.conf and want to 
have the users use a long password to prevent possible guessing by external 
parties. I'm not seeing any such option in my research. If it doesn't exist it 
might be a decent feature. Thanks.

Running:  1.4.25, on CentOS 4.7

Cheers,

Darrin Henshaw


This email and its attachments may be confidential and are intended solely for 
the use of the individual or parties' to whom it is addressed. All comments are 
solely those of the author and do not necessarily represent those of Ignition. 
If you are not the intended recipient of this email and its attachments, you 
must take no action based upon them, nor must you copy or show them to anyone. 
Please contact the sender if you believe you have received this email in error. 
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[asterisk-users] dahdi and overlapdial problem

[Bjoern Metzdorf]

Hi there,

we have a problem with dahdi and overlapdial. We are running an E1 in 
Germany and are in need of overlapdial. The E1 is connected to a Sangoma 
A101.

As soon as overlapdial is set to "yes" we have problems with incoming 
audio on the dahdi channels. When set to "no" all audio is fine.

Basically we can choose between being able to receive calls or to place 
calls :)

Details:

Software:
asterisk-1.6.1.0.tar.gz
libpri-1.4.9.tar.gz
dahdi-linux-2.1.0.4.tar.gz
dahdi-tools-2.1.0.2.tar.gz
wanpipe-3.4.1.tgz

chan_dadhi.conf:
[trunkgroups]
[channels]
context=from-pstn
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
internationalprefix=00
nationalprefix=0
overlapdial=no
;Sangoma A101 port 1 [slot:5 bus:3 span:1] 
switchtype=euroisdn
echocancel=yes
signalling=pri_cpe
channel =>1-15,17-31

/etc/dahdi/system.conf:
# Span 1: WPE1/0 "wanpipe1 card 0" (MASTER) HDB3/CCS/CRC4
span=1,1,0,ccs,hdb3,crc4
# termtype: te
bchan=1-15,17-31
echocanceller=mg2,1-15,17-31
hardhdlc=16
# Global data
loadzone= de
defaultzone = de

Any hints?

Thanks!

Regards,
Bjoern

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Re: [asterisk-users] Speex problem installing on CentOS 5.3

[Steve Totaro]

On Thu, Jun 18, 2009 at 12:46 PM, Brent Davidson <
br...@texascountrytitle.com> wrote:

>  John A. Sullivan III wrote:
>
> Hello, all.  I am delightfully slogging my way through installing and
> configuring Asterisk 1.6.1.1 on CentOS 5.3.  I'm learning lots and
> admiring the product but I'm having a problem getting speex to install
> and I would very much like to use it.  It is not available in menuselect
> and the problem appears to be with speex_preprocess_ctl:
>
> [r...@pbx01 asterisk-1.6.1.1]# grep -i speex config.log
> configure:43813: checking for speex_encode in -lspeex
> configure:43848: gcc -o conftest -g -O2   conftest.c -lspeex  -lm  >&5
> configure:43906: checking speex/speex.h usability
> configure:43947: checking speex/speex.h presence
> configure:44015: checking for speex/speex.h
> configure:44076: checking for speex_preprocess_ctl in -lspeex
> configure:44111: gcc -o conftest -g -O2   conftest.c -lspeex  -lm  >&5
> /home/compuser/Asterisk/asterisk-1.6.1.1/conftest.c:306: undefined
> reference to `speex_preprocess_ctl'
> | #define HAVE_SPEEX 1
> | #define HAVE_SPEEX_VERSION
> | char speex_preprocess_ctl ();
> | return speex_preprocess_ctl ();
> configure:44341: checking for speex_preprocess_ctl in -lspeexdsp
> configure:44376: gcc -o conftest -g -O2   conftest.c -lspeexdsp  -lm
>
>
>  &5
>
>
>  /usr/bin/ld: cannot find -lspeexdsp
> | #define HAVE_SPEEX 1
> | #define HAVE_SPEEX_VERSION
> | char speex_preprocess_ctl ();
> | return speex_preprocess_ctl ();
>
> Internet searches have only further confused the issue for me.  It seems
> this is part of libspeex which in the RedHat world is provided by the
> speex-devel package (which I have installed):
>
> [r...@pbx01 ~]# rpm -qa | grep speex
> speex-devel-1.0.5-4.el5_1.1
> speex-1.0.5-4.el5_1.1
>
> What is the magic to make speex available to Asterisk on CentOS 5.3? Or
> am I stuck having to uninstall the speex packages and install speex from
> source?  Thanks - John
>
>
>
>
> I ended up having to install from source.  There are apparently bits of
> speex that are not included in the RPM's.  It's a farily simple install
> though.
>
> Good luck,
> -Brent
>
>
I am curious if a "yum -y install speex*" would have worked for you?  I will
give it a try on my next 5.3 box.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] Speex problem installing on CentOS 5.3

[John A. Sullivan III]

On Thu, 2009-06-18 at 19:58 +0300, Tzafrir Cohen wrote:
> On Thu, Jun 18, 2009 at 12:41:39PM -0400, John A. Sullivan III wrote:
> 
> > [r...@pbx01 ~]# rpm -qa | grep speex
> > speex-devel-1.0.5-4.el5_1.1
> > speex-1.0.5-4.el5_1.1
> 
> That is too old a version. speex 1.1.x will happen to work. 1.0.x will
> not have the newer "DSP" interface. It does have the basic Speex codec
> functionality.
> 
Ah, OK.  I'll see if I can find a newer RPM so I can maintain some kind
of package version control.  If not, I'll install from source as Brent
recommended.  Thanks - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] Speex problem installing on CentOS 5.3

[Tzafrir Cohen]

On Thu, Jun 18, 2009 at 12:41:39PM -0400, John A. Sullivan III wrote:

> [r...@pbx01 ~]# rpm -qa | grep speex
> speex-devel-1.0.5-4.el5_1.1
> speex-1.0.5-4.el5_1.1

That is too old a version. speex 1.1.x will happen to work. 1.0.x will
not have the newer "DSP" interface. It does have the basic Speex codec
functionality.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Asterisk + mySQL

[Tilghman Lesher]

On Thursday 18 June 2009 10:08:44 jonas kellens wrote:
> There are some things that are not that clear to me :
>
> When I want to write CDR-info to an external MySQL-DB
>
> - do I need to install the asterisk-addons prior to installing Asterisk
> or after having installed Asterisk ??

After.  Addons uses the headers from the Asterisk install to compile itself
correctly.

> - How do I tell Asterisk not to write CDR-info to the Master.csv file
> but into the remote MySQL-DB ? By editing cdr_mysql.conf and pointing to
> an external MySQL-server it seems to me that this is not enough to tell
> Asterisk to stop writing CDR-info also to the Master.csv-file.

In modules.conf:  noload => cdr_csv.so

-- 
Tilghman

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[asterisk-users] Speex problem installing on CentOS 5.3

[John A. Sullivan III]

Hello, all.  I am delightfully slogging my way through installing and
configuring Asterisk 1.6.1.1 on CentOS 5.3.  I'm learning lots and
admiring the product but I'm having a problem getting speex to install
and I would very much like to use it.  It is not available in menuselect
and the problem appears to be with speex_preprocess_ctl:

[r...@pbx01 asterisk-1.6.1.1]# grep -i speex config.log
configure:43813: checking for speex_encode in -lspeex
configure:43848: gcc -o conftest -g -O2   conftest.c -lspeex  -lm  >&5
configure:43906: checking speex/speex.h usability
configure:43947: checking speex/speex.h presence
configure:44015: checking for speex/speex.h
configure:44076: checking for speex_preprocess_ctl in -lspeex
configure:44111: gcc -o conftest -g -O2   conftest.c -lspeex  -lm  >&5
/home/compuser/Asterisk/asterisk-1.6.1.1/conftest.c:306: undefined
reference to `speex_preprocess_ctl'
| #define HAVE_SPEEX 1
| #define HAVE_SPEEX_VERSION
| char speex_preprocess_ctl ();
| return speex_preprocess_ctl ();
configure:44341: checking for speex_preprocess_ctl in -lspeexdsp
configure:44376: gcc -o conftest -g -O2   conftest.c -lspeexdsp  -lm
>&5
/usr/bin/ld: cannot find -lspeexdsp
| #define HAVE_SPEEX 1
| #define HAVE_SPEEX_VERSION
| char speex_preprocess_ctl ();
| return speex_preprocess_ctl ();

Internet searches have only further confused the issue for me.  It seems
this is part of libspeex which in the RedHat world is provided by the
speex-devel package (which I have installed):

[r...@pbx01 ~]# rpm -qa | grep speex
speex-devel-1.0.5-4.el5_1.1
speex-1.0.5-4.el5_1.1

What is the magic to make speex available to Asterisk on CentOS 5.3? Or
am I stuck having to uninstall the speex packages and install speex from
source?  Thanks - John

-- 
John A. Sullivan III
Open Source Development Corporation

Street Preacher: Are you SAVED?!!
Educated Skeptic: Saved from WHAT?!!
Educated Believer: From our selfishness that hurts the ones we love
   and condemns us to an eternity of hurting each other.
http://www.spiritualoutreach.com
Christianity that makes sense


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Re: [asterisk-users] Speex problem installing on CentOS 5.3

[Brent Davidson]

John A. Sullivan III wrote:

Hello, all.  I am delightfully slogging my way through installing and
configuring Asterisk 1.6.1.1 on CentOS 5.3.  I'm learning lots and
admiring the product but I'm having a problem getting speex to install
and I would very much like to use it.  It is not available in menuselect
and the problem appears to be with speex_preprocess_ctl:

[r...@pbx01 asterisk-1.6.1.1]# grep -i speex config.log
configure:43813: checking for speex_encode in -lspeex
configure:43848: gcc -o conftest -g -O2   conftest.c -lspeex  -lm  >&5
configure:43906: checking speex/speex.h usability
configure:43947: checking speex/speex.h presence
configure:44015: checking for speex/speex.h
configure:44076: checking for speex_preprocess_ctl in -lspeex
configure:44111: gcc -o conftest -g -O2   conftest.c -lspeex  -lm  >&5
/home/compuser/Asterisk/asterisk-1.6.1.1/conftest.c:306: undefined
reference to `speex_preprocess_ctl'
| #define HAVE_SPEEX 1
| #define HAVE_SPEEX_VERSION
| char speex_preprocess_ctl ();
| return speex_preprocess_ctl ();
configure:44341: checking for speex_preprocess_ctl in -lspeexdsp
configure:44376: gcc -o conftest -g -O2   conftest.c -lspeexdsp  -lm
  

&5


/usr/bin/ld: cannot find -lspeexdsp
| #define HAVE_SPEEX 1
| #define HAVE_SPEEX_VERSION
| char speex_preprocess_ctl ();
| return speex_preprocess_ctl ();

Internet searches have only further confused the issue for me.  It seems
this is part of libspeex which in the RedHat world is provided by the
speex-devel package (which I have installed):

[r...@pbx01 ~]# rpm -qa | grep speex
speex-devel-1.0.5-4.el5_1.1
speex-1.0.5-4.el5_1.1

What is the magic to make speex available to Asterisk on CentOS 5.3? Or
am I stuck having to uninstall the speex packages and install speex from
source?  Thanks - John

  


I ended up having to install from source.  There are apparently bits of 
speex that are not included in the RPM's.  It's a farily simple install 
though.


Good luck,
-Brent
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[asterisk-users] Asterisk + mySQL

[jonas kellens]

There are some things that are not that clear to me :

When I want to write CDR-info to an external MySQL-DB

- do I need to install the asterisk-addons prior to installing Asterisk
or after having installed Asterisk ??
- How do I tell Asterisk not to write CDR-info to the Master.csv file
but into the remote MySQL-DB ? By editing cdr_mysql.conf and pointing to
an external MySQL-server it seems to me that this is not enough to tell
Asterisk to stop writing CDR-info also to the Master.csv-file.

Thanks for the feedback.

Jonas.
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[asterisk-users] Configuring Asterisk behind a SIP Proxy

[Brad Johnson]

We are trying to configure Asterisk (version 1.6.1.0) with some SIP 
phones behind a SIP Proxy/NAT device. The phones register properly to 
Asterisk, and to get Asterisk to register properly to the external SIP 
registrar we added this to the general section of sip.conf (the address 
of the Asterisk system on the LAN is 192.168.30.5):

outboundproxy=192.168.30.10
register => myname:mysec...@my.provider.com/100

The problem we are facing is that it appears that the outboundproxy 
value is being treated globally by Asterisk so it sends all SIP traffic, 
including traffic to the phones, to the proxy. The behavior we want is 
that all outbound traffic is sent to the proxy, but inbound SIP traffic 
to the phones should be sent direct to the phones.
The result we see is that an inbound Invite is received by Asterisk and 
then the Invite for the phone is sent by Asterisk to the outbound proxy. 
This causes much confusion.
Can anyone please tell me how to configure Asterisk properly for working 
behind a SIP Proxy?
Below you will find our configuration.

Thanks,
Brad

Here is the channel for our SIP provider:

[my_provider]
type=peer
host=my.provider.com
username=100-phone
secret=mysecret
context=incoming
canreinvite=no
qualify=300
insecure=port,invite

Here is a sample phone entry in sip.conf:

[100_phone]
type=friend
username=100-phone
secret=100secret
host=dynamic
context=internal

Here is the relevant part of extensions.conf:

[incoming]
exten => 100,1,Dial(SIP/100_phone,30)
exten => 100,n,Hangup()

[internal]
exten => _X.,1,Dial(SIP/my_provider/${EXTEN})




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Re: [asterisk-users] asterisk and openvpn and sip

[Darrick Hartman (lists)]

Giorgio,

tcpdump and wireshark are your friends.  Instead of guessing, capture a 
call with tcpdump then look at it with wireshark.

Darrick

On 06/18/2009 08:58 AM, Giorgio Incantalupo wrote:
> Hi Darrick,
>
> I always set canreinvite=no 'cause it gives a lot of problems if set to
> yes (and the default is).
> I made a call with rtp debug on and I noticed that normally, on the
> asterisk CLI, I see one packet sent corresponding to one packet  got
> (made a test with a local call on our production server). On the other
> server with the vpn, I get a bunch of sent followed by a group of
> got...there is something in the way the RTP packets are sent/received by
> Asterisk and maybe it can be correlated to the missing audio.
>
> Giorgio
>
> Darrick Hartman (lists) wrote:
>> Do you have 'canreinvite=no' in your sip.conf entry for this phone?  If
>> not, you should.
>>
>> On 06/18/2009 07:55 AM, Giorgio Incantalupo wrote:
>>
>>> Hi John,
>>>
>>> I already have the ccd dir with the iroute (mandatory for routing to
>>> pc/phone connected to vpn client). During the last test I could register
>>> and  make a call but voice disappears after 1, 2 seconds. I'm trying to
>>> understand if it is a bandwidth problem. At the moment I have my phone
>>> connected to the openvpn client (which is its gateway) but I have to use
>>> the vpn ip (10.0.0.1) to register the phone, the openvpn server local ip
>>> (192.168.1.12) is not working. I suppose it is a  sip protocol problem:
>>> I had to change the sip.conf setting nat=yes to make the phone dial and
>>> domain = 10.0.0.1 to make the voice pass (or at least the first 2 seconds).
>>> I keep on working on the vpn since it seems so little is missing to have
>>> a clear conversation. Let me know if your tests are successfull.

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Re: [asterisk-users] asterisk and openvpn and sip

[Giorgio Incantalupo]

Hi Darrick,

I always set canreinvite=no 'cause it gives a lot of problems if set to 
yes (and the default is).
I made a call with rtp debug on and I noticed that normally, on the 
asterisk CLI, I see one packet sent corresponding to one packet  got 
(made a test with a local call on our production server). On the other 
server with the vpn, I get a bunch of sent followed by a group of 
got...there is something in the way the RTP packets are sent/received by 
Asterisk and maybe it can be correlated to the missing audio.

Giorgio

Darrick Hartman (lists) wrote:
> Do you have 'canreinvite=no' in your sip.conf entry for this phone?  If 
> not, you should.
>
> On 06/18/2009 07:55 AM, Giorgio Incantalupo wrote:
>   
>> Hi John,
>>
>> I already have the ccd dir with the iroute (mandatory for routing to
>> pc/phone connected to vpn client). During the last test I could register
>> and  make a call but voice disappears after 1, 2 seconds. I'm trying to
>> understand if it is a bandwidth problem. At the moment I have my phone
>> connected to the openvpn client (which is its gateway) but I have to use
>> the vpn ip (10.0.0.1) to register the phone, the openvpn server local ip
>> (192.168.1.12) is not working. I suppose it is a  sip protocol problem:
>> I had to change the sip.conf setting nat=yes to make the phone dial and
>> domain = 10.0.0.1 to make the voice pass (or at least the first 2 seconds).
>> I keep on working on the vpn since it seems so little is missing to have
>> a clear conversation. Let me know if your tests are successfull.
>> 
>
>   


-- 
Giorgio Incantalupo, mailto:gincantal...@fgasoftware.com
vo...@work - The Agile PBX http://www.voiceatwork.eu
FG&A srl - http://www.fgasoftware.com
Tel: 02 997663.14, Fax: 02 91390172 


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Re: [asterisk-users] Noojeefax help

[wilfried bordoni]

Yes I already saw every page of the web containing "noojeefax" ...

I need a mail to fax, and I think Noojeefax is the only one to provide that.

!DSPAM:4a3a434263739759639719!



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Re: [asterisk-users] Noojeefax help

[Danny Nicholas]

Have you tried this page?  

Why don't you just freefaxforasterisk?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of wilfried
bordoni
Sent: Thursday, June 18, 2009 8:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Noojeefax help

Hi,
I really need some help, I can't find the way to install Noojeefax.
I have the files from sourceforge but there is no readme to explain what 
to do, and no help on the web...

thanks

Will

!DSPAM:4a3a3e6763731933410313!



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[asterisk-users] Noojeefax help

[wilfried bordoni]

Hi,
I really need some help, I can't find the way to install Noojeefax.
I have the files from sourceforge but there is no readme to explain what 
to do, and no help on the web...

thanks

Will

!DSPAM:4a3a3e6763731933410313!



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Re: [asterisk-users] Wideband (G722) MeetMe

[Michael Graves]

--Original Message Text---
From: Doken, Serhad
Date: Wed, 17 Jun 2009 16:07:12 -0700



Hi, 

I wanted to follow up on this thread about WB support on the MeetMe
bridge that is in 1.6.2. Does it only work for G722 or any WB codec ? 

I am working with another 16k WB codec that I can transcode to 722 and
vice versa. I was curious if the 1.6.2 MeetMe bridge can also mix 722
with any other WB codec natively(without downscaling). 

Thanks, 

Serhad Doken 

While not an expert in Asterisk internal, it seems unlikely that
Asterisk is mixing signals in encoded space. It's most likely
converting the stream to slin for mixing then encoding back into
whatever is most appropriate for each end-point.

Michael

--
Michael Graves
mgravesmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:mgra...@mstvp.onsip.com
skype mjgraves
fwd 54245


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Re: [asterisk-users] asterisk-gui: read/write in the conf files or db?

[Danny Nicholas]

It depends on how you are configured.  The gui interfaces using Asterisk
Manager, so you get the Same IO from the gui that you would get from a
native manager session.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Wednesday, June 17, 2009 5:36 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk-gui: read/write in the conf files or db?


Hi All;

asterisk-gui read/write from the conf files or database?

Any advise?
Regards
Bilal


  

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Re: [asterisk-users] asterisk and openvpn and sip

[Darrick Hartman (lists)]

Do you have 'canreinvite=no' in your sip.conf entry for this phone?  If 
not, you should.

On 06/18/2009 07:55 AM, Giorgio Incantalupo wrote:
> Hi John,
>
> I already have the ccd dir with the iroute (mandatory for routing to
> pc/phone connected to vpn client). During the last test I could register
> and  make a call but voice disappears after 1, 2 seconds. I'm trying to
> understand if it is a bandwidth problem. At the moment I have my phone
> connected to the openvpn client (which is its gateway) but I have to use
> the vpn ip (10.0.0.1) to register the phone, the openvpn server local ip
> (192.168.1.12) is not working. I suppose it is a  sip protocol problem:
> I had to change the sip.conf setting nat=yes to make the phone dial and
> domain = 10.0.0.1 to make the voice pass (or at least the first 2 seconds).
> I keep on working on the vpn since it seems so little is missing to have
> a clear conversation. Let me know if your tests are successfull.

-- 
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com

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Re: [asterisk-users] how can I get Better natural Voice in Festival

[Danny Nicholas]

If you like the voice, but it is just too low, you can amplify the Festival
output with sox (sox -V 3 softer.wav louder.wav)

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Thursday, June 18, 2009 1:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] how can I get Better natural Voice in Festival

 

hello All

I am using festival as an application

but it default voice is not good to hear 

anybody have solution about better voice in Festival 

regards
Dhaval

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Re: [asterisk-users] help setting up transfering

[Danny Nicholas]

Have you tried #1103 or *2103?  The # would do a blind transfer, the * would
initiate an attended transfer.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Samad
Sent: Wednesday, June 17, 2009 9:41 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] help setting up transfering

Hi

I am trying to get transferring of calls working, I place a call from
ext 101 => 103 and then from 101 I try and transfer the call to 102
(such that it will be 102=>103), I have tried flash and *2 and nothing
seems to work.

I have allowed transfers in sip.conf, I am expecting a dial tone when i
hit flash

101 -> dahdi/1 a uniden pots phone


Thanks
Alex




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Re: [asterisk-users] Asterisk 1.6.1 and dahdichanname = no

[Remco Barendse]

On Thu, 18 Jun 2009, Kevin P. Fleming wrote:

> Remco Barendse wrote:
>> I am using FreePBX with Asterisk 1.4 and i wanted to upgrade to Asterisk
>> 1.6.1.
>>
>> As FreePBX only supports ZAP naming i set dahdichanname = no in my
>> asterisk.conf.
>>
>> However, after installation the console was still merrily chattering about
>> incoming calls on DAHDI channels and nothing happened because all the
>> ZAP stuff was ignored.
>>
>> Are all Asterisk versions (1.6.0, 1.6.1 as well as the soon to be released
>> 1.6.2) able to deal with dahdichanname = no ?
>>
>> If yes, where could i be going wrong?
>
> No. Only Asterisk 1.4.x releases support 'dahdichanname'; 1.6.x releases
> require DAHDI, and will only support DAHDI channel names. If you saw any
> documentation to the contrary please point it out so we can get it fixed.

Thanks for your reply! Strictly speaking, i didn't. I found some posts on 
a forum that FreePBX works with Asterisk 1.6.x so i just assumed that it 
would work.

My bad, thanks for clearing it up!

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Re: [asterisk-users] asterisk and openvpn and sip

[Giorgio Incantalupo]

Hi John,

I already have the ccd dir with the iroute (mandatory for routing to 
pc/phone connected to vpn client). During the last test I could register 
and  make a call but voice disappears after 1, 2 seconds. I'm trying to 
understand if it is a bandwidth problem. At the moment I have my phone 
connected to the openvpn client (which is its gateway) but I have to use 
the vpn ip (10.0.0.1) to register the phone, the openvpn server local ip 
(192.168.1.12) is not working. I suppose it is a  sip protocol problem: 
I had to change the sip.conf setting nat=yes to make the phone dial and 
domain = 10.0.0.1 to make the voice pass (or at least the first 2 seconds).
I keep on working on the vpn since it seems so little is missing to have 
a clear conversation. Let me know if your tests are successfull.

Thank you. 

Giorgio

John A. Sullivan III wrote:
> On Thu, 2009-06-18 at 10:31 +0200, Giorgio Incantalupo wrote:
>   
>> Hi all,
>>
>> I'm trying to connect one phone to a remote asterisk server via openvpn. 
>> First of all, I put the vpn server on the box hosting asterisk and the 
>> vpn client on another box, both with public ips.
>> Then I set the client ip as my phone IP gateway and the remote pbx ip as 
>> the registrar and outbound proxy.
>>
>> I see in the phone log register packets are sent but nothing in return. 
>> Asterisk console shows it tries to give back the packets but they seem 
>> to be lost somewhere.
>>
>> I made some tests with my pc setting its gateway with the vpn client IP 
>> and I can reach the pbx machine (ping, ssh,...) but sipsak gets no response.
>> It seems ping and ssh response packets are correctly routed but sip 
>> packets aren't.
>>
>> I tried to set nat=yes in sip.conf but without result.
>> Is there any asterisk parameter to set to make it work with openvpn?
>>
>> Any help really appreciated.
>> 
> 
> Hi, Giorgio.  I am a complete noob to Asterisk (well ... an eight year
> noob but only now learning to do more than recipe approaches) but I
> wonder if this is more of a routing than Asterisk issue.
>
> I am also doing my initial testing with OpenVPN and it is working.  My
> setup is slightly different.  OpenVPN is running on the firewall in the
> data center to support remote access; * is on a separate system.  Given
> that you are running * on the OpenVPN gateway, you might want to ensure
> that * is listening on the address of the tun interface.
>
> I found the routing somewhat complicated to set up.  If the clients are
> routed through the VPN client, I found I had to do two things to my data
> center router/firewall:
>   * I had to add a route on the firewall to the network behind the
> client - ip route add 192.168.5.0/24 via 192.168.7.18 (virtual
> openvpn address of my openvpn client)
>   * I had to use a ccd file to add an iroute command telling OpenVPN
> to use my OpenVPN client as a route to the client's network
> (iroute 192.168.5.0 255.255.255.0)
> That worked to allow me to fake a public IP address inside my test lab
> so I could configure some additional gateways; the OpenVPN also worked
> with a softphone running on my OpenVPN client.  Today I will test
> putting these together using hardphones behind my OpenVPN client.  Hope
> this helps - John
>   

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Re: [asterisk-users] snom mass deploy help

[Philipp Kempgen]

> On Jun 18, 2009, at 7:25 AM, Alex Samad  wrote:
>> I am trying to setup asterisk to do a mass deploy of some snom  
>> phones. I
>> can't find where i configure asteriks to listen to the multicast
>> address, nor where to set the notify reply.
>>
>> I was hoping to not have to use dhcp options

Alex Balashov schrieb:
> I thought TFTP (and therefore, DHCP option 66) is the only
> autoprovisioning method Asterisk supports?

Asterisk is not involved here at all.
Snom supports what they call "PnP config".
Technically:
---cut---
# SIP Event Notification:
#   http://tools.ietf.org/html/rfc3265
# SIP UA Profile Event Package:
#   http://tools.ietf.org/html/draft-ietf-sipping-config-framework-15
#   http://tools.ietf.org/html/draft-channabasappa-sipping-app-profile-type-03
#
# Snom 3xx:
#   http://wiki.snom.com/SIP_Traces#PnP_Config

# other drafts:
#   http://tools.ietf.org/html/draft-petrie-sip-config-framework-01
#   
http://www.cs.columbia.edu/sip/drafts/sip/draft-schulzrinne-sip-config-events-00.txt
---cut---

Gemeinschaft (Asterisk-based open-source PBX) comes with a SIP UA
config responder.

http://www.amooma.de/gemeinschaft/
https://svn.amooma.com/gemeinschaft/trunk/opt/gemeinschaft/sbin/gs-sip-ua-config-responder/gs-sip-ua-config-responder

GNU GPL.


Philipp Kempgen
-- 
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 ->  http://www.amoocon.de
-- 

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Re: [asterisk-users] Multiple Outgoing Lines: extensions.conf

[Roger Casaponsa]

hello,

you can define a variable in sip.conf in each extension like:

[201]
...
setvar=LINE=89859716
...

then in extensions when user 201 calls you have a the var defined and you can
use it with ${LINE}.

On Thu, Jun 18, 2009 at 08:19:27PM +1000, Clara Chan wrote:
> Dear all,
> 
>  
> 
> I am currently trying to configure a PBX make use of a multiple of outgoing
> lines, currently my extensions.conf looks something like below
> 
>  
> 
> >> 
> 
>  
> 
> ; extensions.conf
> 
> ; 20th October 2008
> 
>  
> 
>  
> 
> [globals]
> 
> sip1=201
> 
> sip2=202
> 
> sip3=203
> 
> sip4=204
> 
>  
> 
> [general]
> 
> autofallthrough=yes
> 
>  
> 
> [default]
> 
>  
> 
> [incoming_calls]
> 
>  
> 
> exten => _89859715,1,Dial(SIP/201)
> 
> exten => _89859716,1,Dial(SIP/202)
> 
>  
> 
> [macro-sipmail]
> 
> exten => s,1,Verbose(1,Extension ${ARG1})  ;line req to pick up ext if it's 
> not
> reg.
> 
> exten => s,n,Dial(SIP/${ARG1},30)
> 
> exten => s,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail)
> 
> exten => s,n(unavail),Voicemail(${ar...@default,u)
> 
> exten => s,n,Hangup()
> 
> exten => s,n(busy),VoiceMail(${ar...@default,b)
> 
> exten => s,n,Hangup()
> 
>  
> 
> [macro-conference]
> 
> exten => s,1,Playback(conf-theatre)
> 
> exten => s,n,MeetMe(${ARG1},i)
> 
>  
> 
> [internal]
> 
> include => outbound
> 
>  
> 
> ;Voicemail
> 
> exten => 8,1,VoiceMailMain()
> 
>  
> 
> ;Conference Rooms
> 
> exten => 600,1,Macro(conference,600)
> 
> exten => 601,1,Macro(conference,601)
> 
> exten => 602,1,Macro(conference,602)
> 
> exten => 603,1,Macro(conference,603)
> 
> exten => 604,1,Macro(conference,604)
> 
> exten => 605,1,Macro(conference,605)
> 
>  
> 
> ;Extensions
> 
> exten => 201,1,Macro(sipmail,201)
> 
> exten => 202,1,Macro(sipmail,202)
> 
> exten => 203,1,Macro(sipmail,203)
> 
> exten => 204,1,Macro(sipmail,204)
> 
> exten => 205,1,Macro(sipmail,205)
> 
> exten => 206,1,Macro(sipmail,206)
> 
> exten => 207,1,Macro(sipmail,207)
> 
> exten => 208,1,Macro(sipmail,208)
> 
>  
> 
> ;Digium card Channels
> 
> exten => 301,1,Dial(Zap/1-1)
> 
> exten => 302,1,Dial(Zap/1-2)
> 
>  
> 
> [outbound]
> 
> exten => _9.,1,Dial(SIP/${EXTEN:1...@61289859715,30,tr)
> 
> exten => _9.,n,Hangup()
> 
> exten => 000,1,Dial(SIP/0...@61289859715)
> 
>  
> 
> exten => _7.,1,Dial(SIP/${EXTEN:1...@61289859716,30,tr)
> 
> exten => _7.,n,Hangup()
> 
> exten => 000,1,Dial(SIP/0...@61289859716)
> 
>  
> 
> [phones]
> 
> include => internal
> 
> include => incoming_calls
> 
> include => outbound
> 
>  
> 
> >> 
> 
>  
> 
> Each extension has its own incoming and outgoing account, I know how to route
> the incoming number to each particular extension, but how does one route
> outgoing calls from a particular phone to use a specific line, ie, from phone
> no. 89859715 an outgoing call will use caller id 89859715 and line 89859715? 
> Or
> for phone no. 89859716 to use the 89859716 line? 
> 
>  
> 
> I have sixteen outgoing lines I need to configure, so that each individual
> phone can send its own caller id; any suggestions?
> 
>  
> 
> Thanks for your thoughts.
> 
> 
> Rgds,
> 
> Clara
> 

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-- 
Roger Casaponsa - Adam Telefonía IP
email: roger.casapo...@adamvozip.es  
www: http://www.adamvozip.es 
tlf: 902546800

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Re: [asterisk-users] asterisk and openvpn and sip

[John A. Sullivan III]

On Thu, 2009-06-18 at 10:31 +0200, Giorgio Incantalupo wrote:
> Hi all,
> 
> I'm trying to connect one phone to a remote asterisk server via openvpn. 
> First of all, I put the vpn server on the box hosting asterisk and the 
> vpn client on another box, both with public ips.
> Then I set the client ip as my phone IP gateway and the remote pbx ip as 
> the registrar and outbound proxy.
> 
> I see in the phone log register packets are sent but nothing in return. 
> Asterisk console shows it tries to give back the packets but they seem 
> to be lost somewhere.
> 
> I made some tests with my pc setting its gateway with the vpn client IP 
> and I can reach the pbx machine (ping, ssh,...) but sipsak gets no response.
> It seems ping and ssh response packets are correctly routed but sip 
> packets aren't.
> 
> I tried to set nat=yes in sip.conf but without result.
> Is there any asterisk parameter to set to make it work with openvpn?
> 
> Any help really appreciated.

Hi, Giorgio.  I am a complete noob to Asterisk (well ... an eight year
noob but only now learning to do more than recipe approaches) but I
wonder if this is more of a routing than Asterisk issue.

I am also doing my initial testing with OpenVPN and it is working.  My
setup is slightly different.  OpenVPN is running on the firewall in the
data center to support remote access; * is on a separate system.  Given
that you are running * on the OpenVPN gateway, you might want to ensure
that * is listening on the address of the tun interface.

I found the routing somewhat complicated to set up.  If the clients are
routed through the VPN client, I found I had to do two things to my data
center router/firewall:
  * I had to add a route on the firewall to the network behind the
client - ip route add 192.168.5.0/24 via 192.168.7.18 (virtual
openvpn address of my openvpn client)
  * I had to use a ccd file to add an iroute command telling OpenVPN
to use my OpenVPN client as a route to the client's network
(iroute 192.168.5.0 255.255.255.0)
That worked to allow me to fake a public IP address inside my test lab
so I could configure some additional gateways; the OpenVPN also worked
with a softphone running on my OpenVPN client.  Today I will test
putting these together using hardphones behind my OpenVPN client.  Hope
this helps - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] Asterisk 1.6.1 and dahdichanname = no

[Kevin P. Fleming]

Remco Barendse wrote:
> I am using FreePBX with Asterisk 1.4 and i wanted to upgrade to Asterisk 
> 1.6.1.
> 
> As FreePBX only supports ZAP naming i set dahdichanname = no in my 
> asterisk.conf.
> 
> However, after installation the console was still merrily chattering about 
> incoming calls on DAHDI channels and nothing happened because all the 
> ZAP stuff was ignored.
> 
> Are all Asterisk versions (1.6.0, 1.6.1 as well as the soon to be released 
> 1.6.2) able to deal with dahdichanname = no ?
> 
> If yes, where could i be going wrong?

No. Only Asterisk 1.4.x releases support 'dahdichanname'; 1.6.x releases
require DAHDI, and will only support DAHDI channel names. If you saw any
documentation to the contrary please point it out so we can get it fixed.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] snom mass deploy help

[Alex Balashov]

I thought TFTP (and therefore, DHCP option 66) is the only  
autoprovisioning method Asterisk supports?

--
Sent from mobile device

On Jun 18, 2009, at 7:25 AM, Alex Samad  wrote:

> Hi
>
> I am trying to setup asterisk to do a mass deploy of some snom  
> phones. I
> can't find where i configure asteriks to listen to the multicast
> address, nor where to set the notify reply.
>
> I was hoping to not have to use dhcp options
>
> alex
>
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[asterisk-users] snom mass deploy help

[Alex Samad]

Hi

I am trying to setup asterisk to do a mass deploy of some snom phones. I
can't find where i configure asteriks to listen to the multicast
address, nor where to set the notify reply.

I was hoping to not have to use dhcp options

alex



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Re: [asterisk-users] Incoming SIP and the 's' extension

[John A. Sullivan III]

On Thu, 2009-06-18 at 03:50 +, Joseph L. Casale wrote:
> >I then fire up twinkle on my desktop and dial sip:3...@pbx.mycompany.com.
> >The Asterisk console shows:
> >[Jun 17 22:58:55] NOTICE[32060]: chan_sip.c:18160 handle_request_invite:
> >Call from '' to extension '36' rejected because extension not found.
> >
> >If I use the same extensions.conf but change "s" to 36", it works.  I
> >would have expected the SIP channel to see that it had nothing which
> >matched my name or IP address and sent processing to the [incoming]
> >context where it would encounter "s" and process accordingly.
> 
> http://www.voip-info.org/wiki/view/Asterisk+s+extension
> http://voip-info.ehd.com.br/wiki/view/Asterisk+standard+extensions.html
> 
> >What concept am I missing? Does SIP always have a FROM and TO and thus
> >never uses "s"? I'm obviously misunderstanding a fundamental concept.
> >Thanks - John
> 
> You have a known #, your explicitly calling 36 from your soft phone.
> 
> What you want is a pattern match for your sip phones, and the "s" for
> a dahdi line for example...

Ah, ok.  Thanks very much. That's what I thought might be happening but
didn't trust my instincts over my ignorance and over the tutorials I was
following which did not point that out when describing a minimal
dialplan. It makes perfect sense.  Thanks again - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] Multiple Outgoing Lines: extensions.conf

[Ioan Indreias]

Hi Clara,
You could put some data into astdb and query for the outgoing line and
callerid based on internal callerid (extension).

something like

user/201/outline 89859715
user/201/outcallerid 89859715

and so on...

By the way: "_89859715" without the dot (".") is same like 89859715  - maybe
you renounce to the underline ...

HTH,
Ioan.

On Thu, Jun 18, 2009 at 1:19 PM, Clara Chan  wrote:

>  Dear all,
>
>
>
> I am currently trying to configure a PBX make use of a multiple of outgoing
> lines, currently my extensions.conf looks something like below
>
>
>
> >>
>
>
>
> ; extensions.conf
>
> ; 20th October 2008
>
>
>
>
>
> [globals]
>
> sip1=201
>
> sip2=202
>
> sip3=203
>
> sip4=204
>
>
>
> [general]
>
> autofallthrough=yes
>
>
>
> [default]
>
>
>
> [incoming_calls]
>
>
>
> exten => _89859715,1,Dial(SIP/201)
>
> exten => _89859716,1,Dial(SIP/202)
>
>
>
> [macro-sipmail]
>
> exten => s,1,Verbose(1,Extension ${ARG1})  ;line req to pick up ext if it's
> not reg.
>
> exten => s,n,Dial(SIP/${ARG1},30)
>
> exten => s,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail)
>
> exten => s,n(unavail),Voicemail(${ar...@default,u)
>
> exten => s,n,Hangup()
>
> exten => s,n(busy),VoiceMail(${ar...@default,b)
>
> exten => s,n,Hangup()
>
>
>
> [macro-conference]
>
> exten => s,1,Playback(conf-theatre)
>
> exten => s,n,MeetMe(${ARG1},i)
>
>
>
> [internal]
>
> include => outbound
>
>
>
> ;Voicemail
>
> exten => 8,1,VoiceMailMain()
>
>
>
> ;Conference Rooms
>
> exten => 600,1,Macro(conference,600)
>
> exten => 601,1,Macro(conference,601)
>
> exten => 602,1,Macro(conference,602)
>
> exten => 603,1,Macro(conference,603)
>
> exten => 604,1,Macro(conference,604)
>
> exten => 605,1,Macro(conference,605)
>
>
>
> ;Extensions
>
> exten => 201,1,Macro(sipmail,201)
>
> exten => 202,1,Macro(sipmail,202)
>
> exten => 203,1,Macro(sipmail,203)
>
> exten => 204,1,Macro(sipmail,204)
>
> exten => 205,1,Macro(sipmail,205)
>
> exten => 206,1,Macro(sipmail,206)
>
> exten => 207,1,Macro(sipmail,207)
>
> exten => 208,1,Macro(sipmail,208)
>
>
>
> ;Digium card Channels
>
> exten => 301,1,Dial(Zap/1-1)
>
> exten => 302,1,Dial(Zap/1-2)
>
>
>
> [outbound]
>
> exten => _9.,1,Dial(SIP/${EXTEN:1...@61289859715,30,tr)
>
> exten => _9.,n,Hangup()
>
> exten => 000,1,Dial(SIP/0...@61289859715)
>
>
>
> exten => _7.,1,Dial(SIP/${EXTEN:1...@61289859716,30,tr)
>
> exten => _7.,n,Hangup()
>
> exten => 000,1,Dial(SIP/0...@61289859716)
>
>
>
> [phones]
>
> include => internal
>
> include => incoming_calls
>
> include => outbound
>
>
>
> >>
>
>
>
> Each extension has its own incoming and outgoing account, I know how to
> route the incoming number to each particular extension, but how does one
> route outgoing calls from a particular phone to use a specific line, ie,
> from phone no. 89859715 an outgoing call will use caller id 89859715 and
> line 89859715? Or for phone no. 89859716 to use the 89859716 line?
>
>
>
> I have sixteen outgoing lines I need to configure, so that each individual
> phone can send its own caller id; any suggestions?
>
>
>
> Thanks for your thoughts.
>
>
> Rgds,
>
> Clara
>
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-- 
Best regards,
Ioan (Nini) Indreias - indre...@gmail.com
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[asterisk-users] Multiple Outgoing Lines: extensions.conf

[Clara Chan]

Dear all, 

 

I am currently trying to configure a PBX make use of a multiple of
outgoing lines, currently my extensions.conf looks something like below

 

>> 

 

; extensions.conf

; 20th October 2008

 

 

[globals]

sip1=201

sip2=202

sip3=203

sip4=204

 

[general]

autofallthrough=yes

 

[default]

 

[incoming_calls]

 

exten => _89859715,1,Dial(SIP/201)

exten => _89859716,1,Dial(SIP/202)

 

[macro-sipmail]

exten => s,1,Verbose(1,Extension ${ARG1})  ;line req to pick up ext if
it's not reg.

exten => s,n,Dial(SIP/${ARG1},30)

exten => s,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail)

exten => s,n(unavail),Voicemail(${ar...@default,u)

exten => s,n,Hangup()

exten => s,n(busy),VoiceMail(${ar...@default,b)

exten => s,n,Hangup()

 

[macro-conference]

exten => s,1,Playback(conf-theatre)

exten => s,n,MeetMe(${ARG1},i)

 

[internal]

include => outbound

 

;Voicemail

exten => 8,1,VoiceMailMain()

 

;Conference Rooms

exten => 600,1,Macro(conference,600)

exten => 601,1,Macro(conference,601)

exten => 602,1,Macro(conference,602)

exten => 603,1,Macro(conference,603)

exten => 604,1,Macro(conference,604)

exten => 605,1,Macro(conference,605)

 

;Extensions

exten => 201,1,Macro(sipmail,201)

exten => 202,1,Macro(sipmail,202)

exten => 203,1,Macro(sipmail,203)

exten => 204,1,Macro(sipmail,204)

exten => 205,1,Macro(sipmail,205)

exten => 206,1,Macro(sipmail,206)

exten => 207,1,Macro(sipmail,207)

exten => 208,1,Macro(sipmail,208)

 

;Digium card Channels

exten => 301,1,Dial(Zap/1-1)

exten => 302,1,Dial(Zap/1-2)

 

[outbound]

exten => _9.,1,Dial(SIP/${EXTEN:1...@61289859715,30,tr)

exten => _9.,n,Hangup()

exten => 000,1,Dial(SIP/0...@61289859715)

 

exten => _7.,1,Dial(SIP/${EXTEN:1...@61289859716,30,tr)

exten => _7.,n,Hangup()

exten => 000,1,Dial(SIP/0...@61289859716)

 

[phones]

include => internal

include => incoming_calls

include => outbound

 

>> 

 

Each extension has its own incoming and outgoing account, I know how to
route the incoming number to each particular extension, but how does one
route outgoing calls from a particular phone to use a specific line, ie,
from phone no. 89859715 an outgoing call will use caller id 89859715 and
line 89859715? Or for phone no. 89859716 to use the 89859716 line?  

 

I have sixteen outgoing lines I need to configure, so that each
individual phone can send its own caller id; any suggestions?

 

Thanks for your thoughts.


Rgds,

Clara

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[asterisk-users] Asterisk on AVR32

[Paulo Santos]

Greetings everyone,

I'm trying to compile asterisk for an AVR32 (Atmel NGW100).
Buildroot for AVR32 already has the asterisk package, though it has 
bugs. Firstly it tries to apply a patch for 1.2 on a 1.6, but deleting 
the contents of the patch file did the trick.

Now, the problem is making asterisk. The first error is because asterisk 
needed to be ./configure:ed.

Trying to just do ./configure, make gives an error [1].

Trying to do ./configure with the same args as make plus --host it can't 
even configure [2]

I don't know much about cross-compiling, or even regular compiling for 
that matter. Does any one have any idea on how to do this?

Thanks in advance,
Best regards,
Paulo Santos


[1]
menuselect/menuselect --check-deps   menuselect.makeopts
/bin/bash: menuselect/menuselect: cannot execute binary file
make[1]: *** [menuselect.makeopts] Error 126
make[1]: Leaving directory 
`/home/psantos/br/buildroot-avr32-v2.3.0/build_avr32/asterisk-1.6.0-beta6'
make: *** 
[/home/psantos/br/buildroot-avr32-v2.3.0/build_avr32/asterisk-1.6.0-beta6/asterisk]
 
Error 2

[2]
configure: WARNING: If you wanted to set the --build type, don't use --host.
 If a cross compiler is detected then cross compile mode will be used.
checking build system type... i686-pc-linux-gnu
checking host system type... Invalid configuration `CROSS_ARCH=Linux': 
machine `CROSS_ARCH=Linux' not recognized
configure: error: /bin/bash ./config.sub CROSS_ARCH=Linux failed

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Re: [asterisk-users] asterisk and openvpn and sip

[Duncan Turnbull]

Usually this is a routing error with openvpn setup and asterisk thinking 
it needs to route someway other than the vpn. If the originating packets 
have an external ip address asterisk might send them back out another route

Have a look using tcpdump on the server to see where the returned 
packets are destined

Cheers Duncan

Giorgio Incantalupo wrote:
> Hi all,
>
> I'm trying to connect one phone to a remote asterisk server via openvpn. 
> First of all, I put the vpn server on the box hosting asterisk and the 
> vpn client on another box, both with public ips.
> Then I set the client ip as my phone IP gateway and the remote pbx ip as 
> the registrar and outbound proxy.
>
> I see in the phone log register packets are sent but nothing in return. 
> Asterisk console shows it tries to give back the packets but they seem 
> to be lost somewhere.
>
> I made some tests with my pc setting its gateway with the vpn client IP 
> and I can reach the pbx machine (ping, ssh,...) but sipsak gets no response.
> It seems ping and ssh response packets are correctly routed but sip 
> packets aren't.
>
> I tried to set nat=yes in sip.conf but without result.
> Is there any asterisk parameter to set to make it work with openvpn?
>
> Any help really appreciated.
>
> Thank you.
>
> Giorgio
>
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[asterisk-users] Asterisk 1.6.1 and dahdichanname = no

[Remco Barendse]

I am using FreePBX with Asterisk 1.4 and i wanted to upgrade to Asterisk 
1.6.1.

As FreePBX only supports ZAP naming i set dahdichanname = no in my 
asterisk.conf.

However, after installation the console was still merrily chattering about 
incoming calls on DAHDI channels and nothing happened because all the 
ZAP stuff was ignored.

Are all Asterisk versions (1.6.0, 1.6.1 as well as the soon to be released 
1.6.2) able to deal with dahdichanname = no ?

If yes, where could i be going wrong?

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[asterisk-users] asterisk and openvpn and sip

[Giorgio Incantalupo]

Hi all,

I'm trying to connect one phone to a remote asterisk server via openvpn. 
First of all, I put the vpn server on the box hosting asterisk and the 
vpn client on another box, both with public ips.
Then I set the client ip as my phone IP gateway and the remote pbx ip as 
the registrar and outbound proxy.

I see in the phone log register packets are sent but nothing in return. 
Asterisk console shows it tries to give back the packets but they seem 
to be lost somewhere.

I made some tests with my pc setting its gateway with the vpn client IP 
and I can reach the pbx machine (ping, ssh,...) but sipsak gets no response.
It seems ping and ssh response packets are correctly routed but sip 
packets aren't.

I tried to set nat=yes in sip.conf but without result.
Is there any asterisk parameter to set to make it work with openvpn?

Any help really appreciated.

Thank you.

Giorgio

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Re: [asterisk-users] gap between Playback and Queue

[Louis-David Mitterrand]

On Wed, Jun 17, 2009 at 01:08:33PM -0500, Danny Nicholas wrote:
> If this is a recorded sound, you might want to truncate it with lame or
> audacity.  It is quite common in my shop as we record using the phones.

Thanks for this suggestion.

The problem was indeed a silence at the beginning of my musiconhold
tracks. Audacity did a fine job and fixed my problem.

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Re: [asterisk-users] ODP: Re: Polycom Stop Downloading Config

[randulo]

>> What happens if the http server is down?  My point is that I don't
>> want it
>> to try and pull any config from a server.  I just want it to use
>> its local
>> config.

I don't recall this looping probelm. The value of tries is supposed to
prevent this from happening.

r

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