Re: [asterisk-users] HP DL380 G5 with TE420

[Hans Konings]

Digium suspects these cards have a problem, so they are being rma'd.

http://www.digium.com/en/docs/tech_bulletins/20081113.php

Thanks for the input.



On Thu, Feb 26, 2009 at 2:34 PM, Miguel Molina wrote:

> Hans Konings escribió:
> > Hi
> >
> > I'm having problems getting the TE420 working in HP DL380G5 servers.
> >
> > The cards don't seem to be detected 100% by the BIOS. With two cards
> > in the server they are never detected.
> >
> Did you test the TE420 card on another server? It may be a defective
> card...
>
> --
> Ing. Miguel Molina
> Grupo de Tecnología
> Millenium Phone Center
> PBX: (+57 1)6500800 ext. 1201
> Fax: (+57 1)6500816
> Móvil: (+57)3138873587
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Question about Do Not Disturb

[Alexander Lopez]

In a nut shell the CHANNEL variable is just that variable. It has a call leg id 
attached to it so if that is what you are storing it will change everytime you 
create a new channel. 
For example if I place a call Thru SIP channel polycom1 the channel is:

SIP/polycom1-23a3bc, You could look at parsing the channel name out but that 
may not work depending on your peer names.

You may instead try using the EXTEN variable and testing for it when they dial. 

 Kindly consider the environment before printing this e-mail.


> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Haim Dimer
> Sent: Thursday, February 26, 2009 6:34 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Question about Do Not Disturb
> 
> Hello,
> 
> Some of my users have phones lacking a DND button. I need to provide
> an extension they can dial that will put them in DND, i.e. tell the
> server not to send them any calls until they get off the DND.
> 
> I've researched it for almost 3 days now and tried a range of
> configurations. I'm hoping somebody here has an answer. Currently, I
> have this in extensions.conf
> 
> [app-dnd-on]
> exten => *78,1,Answer
> exten => *78,n,NoOp(${CALLERID(num)} channel ${CHANNEL} is going on
> DND ACTIVE)
> exten => *78,n,Set(DB(DND/${CALLERID(num)})=On)
> exten => *78,n,UserEvent(ASTDB|Family: DND^Channel: ${CHANNEL} ^Value:
> On)
> exten => *78,n,Playback(do-not-disturb&activated)
> exten => *78,n,Hangup
> 
> [app-dnd-off]
> exten => *79,1,Answer
> exten => *79,n,NoOp(${CALLERID(num)} is going OFF DND)
> exten => *79,n,DBdel(DND/${CALLERID(num)})
> exten => *79,n,UserEvent(ASTDB|Family: DND^Channel: ${CHANNEL} ^Value:
> ^)
> exten => *79,n,Playback(do-not-disturb&de-activated)
> exten => *79,n,Hangup
> 
> Using the above config, if I dial *78 I hear Allison's voice telling
> me that do not disturb is activated but I can still be called (either
> directly or as part of a queue). BTW, there are many people on the
> wiki stuck with the same problem : http://www.voip-
> info.org/wiki/index.php?page_id=787&tk=c7f21c26a40ee72393d7&comments_page=
> 1
> 
> About the system:
> 
> bell*CLI> core show version
> Asterisk 1.4.23.1 built by root @ bell on a i686 running Linux on
> 2009-01-26 01:57:36 UTC
> 
> GUI-version : SVN-branch-2.0-r4489 (It's the Digium GUI)
> 
> Thanks!
> 
> Haim.
> 
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> --
> This message was scanned by Connected.Net and is believed to be clean.


--
This message was scanned by Connected.Net and is believed to be clean.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] call file concurrency

[Bill Michaelson]

Is there a convenient way to limit the number of call files (outgoing 
directory) that are processed concurrently?


smime.p7s
Description: S/MIME Cryptographic Signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Current state of Asterisk and Virtualization?

[Senad Jordanovic]

Gavin Henry wrote:
> Hi all,
> 
> In a pure VoIP env, what is the current state of do's and don't s of
> virtualizing * in order to provide multiple separate instances, say
> for hosting lots of Asterisk-gui/FreePBX/a-n-other gui?
> 
> I've read lots of threads going back to 2007 and I'm in the general
> option that kvm is the way to go now, if at all.
> 
> If dadhi_dummy/zt_dummy is still an issue for conferencing etc. a
> conference box could be put along side the vm hardware and have a card
> in it.
> 
> Thoughts, experiences and being told to shut up are all very much appreciated.
> 
> Thanks.
> 

http://www.bicomsystems.com/products/C/P/797/411/



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] using Cisco IP Communicator with SIP to Asterisk

[Hugo Garcia Gomez]

Hi,

 

I'm using a Cisco IP Communicator 7.0 enable with SIP, but I can't to
configure it to add the lines because  I can't to "Select" the options, for
example: the Name, Authentication name... 

Somebody help me??

 

 

 

Thank you!!!

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Problems with Outbound Calls

[David fire]

thanks

2009/2/26 Tzafrir Cohen 

> On Thu, Feb 26, 2009 at 05:06:56PM -0200, David fire wrote:
> > sorry but how do you know the warning is from an # ?
>
> 'Directive' is something that begins with a '#'.
>
> --
>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
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
(")_(")signature to help him gain world domination.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Question about Do Not Disturb

[Haim Dimer]

Hello,

Some of my users have phones lacking a DND button. I need to provide  
an extension they can dial that will put them in DND, i.e. tell the  
server not to send them any calls until they get off the DND.

I've researched it for almost 3 days now and tried a range of  
configurations. I'm hoping somebody here has an answer. Currently, I  
have this in extensions.conf

[app-dnd-on]
exten => *78,1,Answer
exten => *78,n,NoOp(${CALLERID(num)} channel ${CHANNEL} is going on  
DND ACTIVE)
exten => *78,n,Set(DB(DND/${CALLERID(num)})=On)
exten => *78,n,UserEvent(ASTDB|Family: DND^Channel: ${CHANNEL} ^Value:  
On)
exten => *78,n,Playback(do-not-disturb&activated)
exten => *78,n,Hangup

[app-dnd-off]
exten => *79,1,Answer
exten => *79,n,NoOp(${CALLERID(num)} is going OFF DND)
exten => *79,n,DBdel(DND/${CALLERID(num)})
exten => *79,n,UserEvent(ASTDB|Family: DND^Channel: ${CHANNEL} ^Value:  
^)
exten => *79,n,Playback(do-not-disturb&de-activated)
exten => *79,n,Hangup

Using the above config, if I dial *78 I hear Allison's voice telling  
me that do not disturb is activated but I can still be called (either  
directly or as part of a queue). BTW, there are many people on the  
wiki stuck with the same problem : 
http://www.voip-info.org/wiki/index.php?page_id=787&tk=c7f21c26a40ee72393d7&comments_page=1

About the system:

bell*CLI> core show version
Asterisk 1.4.23.1 built by root @ bell on a i686 running Linux on  
2009-01-26 01:57:36 UTC

GUI-version : SVN-branch-2.0-r4489 (It's the Digium GUI)

Thanks!

Haim.


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

[Steve Edwards]

On Thu, 26 Feb 2009, Ira wrote:

> I have a locked ATA here and I didn't pay for it. So while I'm not happy 
> it's locked and I can't use it, I can't complain that much as it was 
> free.

It may be more accurate to say "I have a locked ATA that was paid for by 
my provider charging me a little bit each month during my commitment 
period..."

Thanks in advance,

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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

[Ira]

At 01:25 PM 2/26/2009, you wrote:
I don't mind them
providing autoconfiguration, as many of their customers would be lost
otherwise, but blocking factory reset and other tricks they play is IMO
morally wrong, particularly since the customer paid for the
ATA.
I have a locked ATA here and I didn't pay for it. So while I'm not happy
it's locked and I can't use it, I can't complain that much as it was
free.
Ira



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SheevaPlug Development Kit

[Tzafrir Cohen]

On Thu, Feb 26, 2009 at 10:35:31PM +0100, Hans Witvliet wrote:

> Like to know if the power plug is only suitable for USA?

But can also take 220V, according to specs.

-- 
   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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SheevaPlug Development Kit

[Hans Witvliet]

On Wed, 2009-02-25 at 19:14 -0200, David fire wrote:
> please keep us informed about it.
> David
> 
> 2009/2/25 Kristian Kielhofner 
> Hello everyone,
> 
>  I just ordered one of these:
> 
> 
> http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp
> 
>  Just over $110 with shipping but they are expecting the price
> to
> come down quite a bit:
> 
> - 1.2Ghz ARM5
> - 512MB RAM
> - Multiple flash storage options
> - Gigabit ethernet
> - USB 2.0
> - 5 watt power usage
> 
>  They probably won't be shipping until late March but I
> thought I'd
> get my order in early.
> 
>  Of course one of my first tasks will be to get Asterisk
> running on 
> 
I noticed that at the website of marvel, they have a couple of other
tiny systems's (though unclear if they are already for sale, or just
future systems)...

Like to know if the power plug is only suitable for USA?

hw

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Dropping RTP packets

[Jim Dickenson]

The problem turned out to be a firewall issue. If one makes a call out the
PRI line * sends the ring audio to the sip phone. This opened a hole in the
firewall for the return traffic so things worked. If I make the call from my
office when the call was answered and the caller started talking and that
opened the hole.

As no traffic went out the firewall no dynamic hole was created so nothing
passed.

I had remembered a previous router's configuration that allowed any UDP
traffic and my current one is setup different.

One problem with working at this for 33 years and remembering things from
the past that are different now.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



> From: Brent Davidson 
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Thu, 26 Feb 2009 15:18:14 -0600
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [asterisk-users] Dropping RTP packets
> 
> You need "canreinvite=no" in the config for your sip phone and the
> veracity connection, otherwise Asterisk will just mediate the call setup
> then try to allow the sip phone and veracity to talk directly to one
> another.
> 
> Jim Dickenson wrote:
>> I have a SIP phone at home behind a NAT router registered with an * box at
>> my office with a routable static IP address running version
>> SVN-branch-1.6.0-r175638M.
>> 
>> If I make a call from my SIP phone out a PRI circuit to my cell phone
>> everything works as expected. I hear audio in both directions and all is
>> good.
>> 
>> If from the same SIP phone I make a call via our Veracity SIP account to my
>> cell phone I hear no audio in either direction.
>> 
>> In trying to find out what is wrong I used tcpdump to see if I could learn
>> anything. I can see the phone sending fixed length UDP packets on to my home
>> network heading to the IP address of the * box. If I run tcpdump on the *
>> box I do not see the packets being received. I do not see the * box sending
>> any packets to my home network either. I have not checked if the * box is
>> receiving packets from Veracity I only know that no audio packets are sent
>> to my home network.
>> 
>> If I use tcpdump to watch the SIP phone call via the PRI circuit I see
>> packets both on my home network and my * box.
>> 
>> If I use a SIP phone located in my office and make a call via Veracity
>> everything is okay. Also a co-worker has a vpn router on his home network
>> connected to the office vpn server and he can make calls from his SIP phone
>> via Veracity without problems.
>> 
>> I can also call his SIP phone from my SIP phone and packets pass as
>> expected.
>> 
>> It seems as if audio packets from my SIP phone disappear only if they are
>> involved with a call via Veracity.
>> 
>> Does anyone have some idea what I might look at to find what is causing this
>> problem?
>>   
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

[Jeff LaCoursiere]

On Thu, 26 Feb 2009, Wilton Helm wrote:

>> Actually you want to use the one with -N/A at the end.
>
> Thanks for the correction.  My memory failed me.  But definitely I want 
> people to know that not every PAP2T is useful to them so they don't get 
> burned.  IMO Vonage (and others) should not be locking them the way they 
> do (and Linksys shouldn't make it possible).  I don't mind them 
> providing autoconfiguration, as many of their customers would be lost 
> otherwise, but blocking factory reset and other tricks they play is IMO 
> morally wrong, particularly since the customer paid for the ATA.
>

I don't think they are "locking" the same device that you buy when you buy 
the "-NA" version.  I believe that Linksys is making pre-configured 
devices for these large buyers and selling them much cheaper to them in 
bulk than they sell the -NA version to the community at large.

I agree that it sucks, though ;)

j

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

[Wilton Helm]

>Actually you want to use the one with -N/A at the end.

Thanks for the correction.  My memory failed me.  But definitely I want people 
to know that not every PAP2T is useful to them so they don't get burned.  IMO 
Vonage (and others) should not be locking them the way they do (and Linksys 
shouldn't make it possible).  I don't mind them providing autoconfiguration, as 
many of their customers would be lost otherwise, but blocking factory reset and 
other tricks they play is IMO morally wrong, particularly since the customer 
paid for the ATA.

Wilton
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Problems with Outbound Calls

[Tzafrir Cohen]

On Thu, Feb 26, 2009 at 05:06:56PM -0200, David fire wrote:
> sorry but how do you know the warning is from an # ?

'Directive' is something that begins with a '#'.

-- 
   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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Dropping RTP packets

[Brent Davidson]

You need "canreinvite=no" in the config for your sip phone and the 
veracity connection, otherwise Asterisk will just mediate the call setup 
then try to allow the sip phone and veracity to talk directly to one 
another.

Jim Dickenson wrote:
> I have a SIP phone at home behind a NAT router registered with an * box at
> my office with a routable static IP address running version
> SVN-branch-1.6.0-r175638M.
>
> If I make a call from my SIP phone out a PRI circuit to my cell phone
> everything works as expected. I hear audio in both directions and all is
> good.
>
> If from the same SIP phone I make a call via our Veracity SIP account to my
> cell phone I hear no audio in either direction.
>
> In trying to find out what is wrong I used tcpdump to see if I could learn
> anything. I can see the phone sending fixed length UDP packets on to my home
> network heading to the IP address of the * box. If I run tcpdump on the *
> box I do not see the packets being received. I do not see the * box sending
> any packets to my home network either. I have not checked if the * box is
> receiving packets from Veracity I only know that no audio packets are sent
> to my home network.
>
> If I use tcpdump to watch the SIP phone call via the PRI circuit I see
> packets both on my home network and my * box.
>
> If I use a SIP phone located in my office and make a call via Veracity
> everything is okay. Also a co-worker has a vpn router on his home network
> connected to the office vpn server and he can make calls from his SIP phone
> via Veracity without problems.
>
> I can also call his SIP phone from my SIP phone and packets pass as
> expected.
>
> It seems as if audio packets from my SIP phone disappear only if they are
> involved with a call via Veracity.
>
> Does anyone have some idea what I might look at to find what is causing this
> problem?
>   

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] changing /etc/dahdi/system.conf

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Brandon B. wrote:
> At the top of my /etc/dahdi/system.conf file is this line:
> 
> # Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25 18:25:10
> 2009 -- do not hand edit
> 
> OK, so how do I adjust the timing source and LBO numbers, and echo
> cancellers if I'm not supposed to edit this file?

I had the same question when I ran dahdi_genconf.

My answer: ignore that message.  You'll only want to run dahdi_genconf
once, which creates that file.  Then edit it to your heart's content.

Barry
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)

iD8DBQFJpwQkCFu3bIiwtTARAgY6AKCEWrmsKFQdWbdRh9NgjCakJJk9qwCgioSB
5ufZXOMuUBalAxWPvhfauTE=
=r10C
-END PGP SIGNATURE-

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] changing /etc/dahdi/system.conf

[Tzafrir Cohen]

On Thu, Feb 26, 2009 at 01:44:19PM -0700, Brandon B. wrote:
> At the top of my /etc/dahdi/system.conf file is this line:
> 
> # Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25 18:25:10 2009
> -- do not hand edit
> 
> OK, so how do I adjust the timing source and LBO numbers, and echo
> cancellers if I'm not supposed to edit this file?

Hmm I guess you should n't take that too seriously :-(

Maybe this text should be changed.

-- 
   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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] changing /etc/dahdi/system.conf

[Joseph L. Casale]

>At the top of my /etc/dahdi/system.conf file is this line:
>
>    # Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25 18:25:10 2009 -- 
>do not hand edit
>
>OK, so how do I adjust the timing source and LBO numbers, and echo cancellers 
>if I'm not supposed to edit this file?

Well, if you hand edit it _and_ then rerun dahdi_genconf you then lose your 
edits.
So, if you need to re-run it, put the changes back in :)

heh,
jlc

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] dahdi wcb4xxp and fax

[Olivier]

2009/2/25 stoffell 

> Hi all,
>
> I wanted to switch from my current setup (mISDN) to the native dahdi with
> b410p support (wcb4xp). All works fine for normal phone calls but not for
> faxing. Faxes are distorted, if arriving at all, and hylafax logs the usual
> bad stuff (HDLC frame not byte-oriented.)


What about outgoing faxes ?

>
>
> Our setup uses a digium b410p card with asterisk 1.6, latest libpri and
> dahdi, hylafax with iaxmodem, and all this on 1 machine.
>
> chan_dahdi.conf contains:
> faxdetect=both
>
> When receiving a fax call, hylafax (iaxmodem) answers the call after the
> obligatory wait of 3 seconds (fax detection) but to me it seems that echo
> cancellation is still being done.


Theory is that any echo canceller hearing a 2100Hz fax signal would halt
itself, so I wouldn't search in that direction first.

Have you tried native 1.6 sendFax, receiveFax ?
Maybe it would improve fax performance.

>
>
> Any pointers on this or workarounds? We're back to our old misdn setup for
> now ;)
>
> Here's some output from "dahdi show channel 1" (the one that had the fax
> connection going), i cut out some non-related stuff :
> *CLI> dahdi show channel 4
> Signalling Type: ISDN BRI Point to Point
> Owner: DAHDI/4-1
> Real: DAHDI/4-1
> Callwait: 
> Threeway: 
> Confno: -1
> DSP: yes
> Busy Detection: no
> TDD: no
> Relax DTMF: no
> Dialing/CallwaitCAS: 0/0
> Default law: alaw
> Fax Handled: yes
> Pulse phone: no
> DND: no
> Echo Cancellation:
> 128 taps
> (unless TDM bridged) currently ON
> PRI Flags: Call
> PRI Logical Span: Implicit
> Actual Confinfo: Num/0, Mode/0x
> Actual Confmute: No
>
>
>
> Regards,
> stoffell
>
>
>
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] changing /etc/dahdi/system.conf

[Brandon B.]

At the top of my /etc/dahdi/system.conf file is this line:

# Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25 18:25:10 2009
-- do not hand edit

OK, so how do I adjust the timing source and LBO numbers, and echo
cancellers if I'm not supposed to edit this file?

Brandon.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] AGI problem using mono (.Net)

[Steve Edwards]

On Thu, 26 Feb 2009, Douglas Mortensen wrote:

> New problem:
> ==
> Although we see the following in our logs / asterisk console: [Feb 26 
> 11:11:05] VERBOSE[9824] logger.c:  -- Playing 'filename' 
> (escape_digits=) (sample_offset 0)
>
> All that the caller hears is a very brief click. And then the dial plan 
> continues. This is causing me to wonder whether asterisk halts playback 
> of the file, if the AGI script that send the STREAM FILE command 
> completes/returns.

If your AGI is exiting before the sound file has finished and without 
reading the "result" from STDIN, you have violated the protocol.

> Talking to my developer, I asked him to create a loop after sending the 
> STREAM FILE command and read from stdin until he gets a string that 
> starts with 200.

All it takes is a single "line oriented" read. This is pretty "elemental" 
to the protocol -- issue a request, read the result. Most (all?) AGI 
libraries "wrap" these two steps into a single function call so the 
library user never has the opportunity to do anything in between.

> I'm supposing that asterisk would send this AFTER the audio file has 
> successfully played out. But again, this is only a guess.

It makes sense that Asterisk sends the "result" when the request is 
finished, successful or not.

A few more suggestions:

) Try to STREAM FILE demo-congrats.

) Verify that your sound file is playable (in the correct format for the 
codecs you have loaded) by using playback() in the dialplan.

Apologies if it is too basic (but still I forget sometimes...) that the 
file name is specified without the file type and is relative to Asterisk's 
sound directory (usually /var/lib/asterisk/sounds/) if the file name does 
not start with a slash.

The error message you showed above looks like STREAM FILE is finding the 
file. I'm guessing that you are still violating the protocol or that the 
file is not in an acceptable format for the codecs you have loaded.

It appears that you did not use an existing AGI library. Why not?

Thanks in advance,

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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

[pe...@networkoblivion.com]

Nope, I gave up and haven't gone back.  Not worth the hassle.  We use 
the SPA2100/SPA2102 and they work great for providing analog/fax lines 
as they support T.38.

Olivier wrote:
> 
> 2009/2/26 pe...@networkoblivion.com  
> mailto:pe...@networkoblivion.com>>
> 
> I used the Patton M-ATA a year or so ago and it was a piece of junk.
> T.38 didn't work and there was no way to troubleshoot why it didn't
> work.
> 
> 
> Have you tried again recently ?
> I've just tried its T.38 capabilities and I'm not successful yet.
>  
> 
>  Also, the web interface was horrible.
> 
> Mike wrote:
>  > Hi,
>  >
>  >
>  >
>  > I am looking for a good ATA recommendation, ideally something:
>  >
>  >
>  >
>  > 1) with one FXS and one LAN port (so it's as inexpensive as possible)
>  >
>  >
>  >
>  > 2) That can be provisioning using _FTP_ (configuration and
> firmware upon
>  > reboot, ideally remote reboot from a sip notify)
>  >
>  >
>  >
>  > 3) Supports T.38
>  >
>  >
>  >
>  > Nice to have would be:
>  >
>  > a) PoE powered and AC powered (my choice)
>  >
>  > b) Small size-wise
>  >
>  >
>  >
>  > I have been recommended the PAP2T in the past, and although I
> have used
>  > it and sort of liked it, it wasn't possible to provision it using FTP
>  > (at least as far as I could tell)
>  >
>  >
>  >
>  > Any tip is welcomed.  I`m looking at the Patton ATA which is
> small, but
>  > it doesn't support FTP provisioning either as far as I can tell.
>  >
>  >
>  >
>  >
>  >
>  >
>  >
>  > Mike
>  >
>  >
>  >
> 
>  >
>  > ___
>  > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>  >
>  > asterisk-users mailing list
>  > To UNSUBSCRIBE or update options visit:
>  >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> 
> 
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Current state of Asterisk and Virtualization?

[Gavin Henry]

Hi all,

In a pure VoIP env, what is the current state of do's and don't s of
virtualizing * in order to provide multiple separate instances, say
for hosting lots of Asterisk-gui/FreePBX/a-n-other gui?

I've read lots of threads going back to 2007 and I'm in the general
option that kvm is the way to go now, if at all.

If dadhi_dummy/zt_dummy is still an issue for conferencing etc. a
conference box could be put along side the vm hardware and have a card
in it.

Thoughts, experiences and being told to shut up are all very much appreciated.

Thanks.

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Odd Read App Issues

[Brent Davidson]

Robert Broyles wrote:
I turned on DTMF debugging. It looks like the extra digits coming in 
are less than the minimum duration of 100ms


Anyone know how to force that minimum duration?

[Feb 26 12:15:07] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'1' received on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '1' on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '1' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:07] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '1' on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'1' received on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '1' on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '1' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:07] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '1' on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'1' received on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '1' on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '1' 
received on SIP/carrier-c4022740, duration 20 ms
[Feb 26 12:15:07] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '1' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'2' received on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '2' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '2' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:08] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '2' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'2' received on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '2' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '2' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:08] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '2' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'2' received on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '2' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '2' 
received on SIP/carrier-c4022740, duration 20 ms
[Feb 26 12:15:08] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '2' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'3' received on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '3' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '3' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:08] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '3' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'3' received on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '3' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '3' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:08] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '3' on SIP/carrier-c4022740
[Feb 26 12:15:09] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'4' received on SIP/carrier-c4022740
[Feb 26 12:15:09] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '4' on SIP/carrier-c4022740
[Feb 26 12:15:09] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '4' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:09] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '4' on SIP/carrier-c4022740
[Feb 26 12:15:09] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'4' received on SIP/carrier-c4022740
[Feb 26 12:15:09] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '4' on SIP/carrier-c4022740
[Feb 26 12:15:09] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '4' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:09] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '4' on SIP/carrier-c4022740
[Feb 26 12:15:10] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'5' received on SIP/carrier-c4022740
[Feb 26 12:15:10] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '5' on SIP/carrier-c4022740
[Feb 26 12:15:10] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '5' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:10] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '5' on SIP/c

[asterisk-users] Friday Feb 27th at 12 Noon EST: Polycom Applications

[randulo]

Hi,

We're moving ahead on this, there are some interesting files to check
out thanks to davevg's perl skills. If you find a way to write an
interesting application to run on a Polycom, you can win a Polycom IP
450. Even if you don't care about that, you may be interested in
knowing more about programming for the Polycom. These phones are way
underused IMO.

Join us for a discussion of programming for a Polycom Friday.

The Application Contest is here: http://winPolycom

Dial in info for PSTN and SIP is here on the front page: http://tr.im/voip

BYOB

/r

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Odd Read App Issues

[Robert Broyles]

I turned on DTMF debugging. It looks like the extra digits coming in are 
less than the minimum duration of 100ms


Anyone know how to force that minimum duration?

[Feb 26 12:15:07] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '1' 
received on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '1' on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '1' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:07] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '1' on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '1' 
received on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '1' on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '1' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:07] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '1' on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '1' 
received on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '1' on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '1' 
received on SIP/carrier-c4022740, duration 20 ms
[Feb 26 12:15:07] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '1' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '2' 
received on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '2' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '2' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:08] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '2' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '2' 
received on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '2' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '2' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:08] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '2' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '2' 
received on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '2' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '2' 
received on SIP/carrier-c4022740, duration 20 ms
[Feb 26 12:15:08] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '2' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '3' 
received on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '3' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '3' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:08] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '3' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '3' 
received on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '3' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '3' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:08] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '3' on SIP/carrier-c4022740
[Feb 26 12:15:09] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '4' 
received on SIP/carrier-c4022740
[Feb 26 12:15:09] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '4' on SIP/carrier-c4022740
[Feb 26 12:15:09] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '4' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:09] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '4' on SIP/carrier-c4022740
[Feb 26 12:15:09] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '4' 
received on SIP/carrier-c4022740
[Feb 26 12:15:09] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '4' on SIP/carrier-c4022740
[Feb 26 12:15:09] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '4' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:09] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '4' on SIP/carrier-c4022740
[Feb 26 12:15:10] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '5' 
received on SIP/carrier-c4022740
[Feb 26 12:15:10] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '5' on SIP/carrier-c4022740
[Feb 26 12:15:10] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '5' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:10] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '5' on SIP/carrier-c4022740
[Feb 26

[asterisk-users] asterisk 1.4.23.1 and mISDN 1.1.8 segfaults

[stoffell]

hi all,

I'm have a bit of a hard time with some segfaults on running 1.4.23.1 and
mISDN 1.1.8. I already enabled " DONT_OPTIMIZE" and " DEBUG_THREADS" in
asterisk so I can now generate a bt. I did that (following the instructions
on voip-info) but I'm not sure how to "read' the output now.

By looking  at the bt below, can one see if the problem is caused by
something special ? I'm running debian etc with kernel 2.6.18-6-686.

Core was generated by `asterisk -vg'.
Program terminated with signal 11, Segmentation fault.
#0  0xb7dfb5f7 in malloc_consolidate () from /lib/libc.so.6
(gdb) bt
#0  0xb7dfb5f7 in malloc_consolidate () from /lib/libc.so.6
#1  0xb7dfd236 in _int_malloc () from /lib/libc.so.6
#2  0xb7dffd56 in calloc () from /lib/libc.so.6
#3  0x08118697 in _ast_calloc (num=1, len=1420, file=0x8157ed4 "manager.c",
lineno=2432, func=0x81597b9 "accept_thread")
at
/srv/software/misdn-asterisk-20081107/asterisk-1.4.23.1/include/asterisk/utils.h:358
#4  0x080cb2ea in accept_thread (ignore=0x0) at manager.c:2432
#5  0x0811a025 in dummy_start (data=0x81acc70) at utils.c:856
#6  0xb7f31c51 in pthread_start_thread () from /lib/libpthread.so.0
#7  0xb7e537fa in clone () from /lib/libc.so.6
(gdb) q

Thanks in advance for any pointers.

cheers
stoffell
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] CDR - Asterisk-Stat and PHP5

[Tiago Durante]

On Thu, Feb 26, 2009 at 12:52 PM, Paulo Santos  wrote:
> Marco Signorini wrote:
>> Hi Tiago.
>>
>> I've it working on PHP 5.2.6 but only after having modified the php.ini
>> default configuration keys:
>>
>> zend.ze1_compatibility_mode = Off
>> short_open_tag = Off
>
> Though my zend.ze1_compatibility_mode is set to Off, short_open_tag is
> set to On and it is working.
>
> Those are my defaults, at least I never changed them. Installed with
> apt-get on Debian 4.0, PHP version 5.2.0-8+etch13.

Cool, I'm gonna test it and I let you guys know if worked or not.

Thanks a lot!

-- 
Tiago Durante

,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,
Perseverance is the hard work you do after you
get tired of doing the hard work you already did.
-- Newt Gingrich

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] DTMF tones mid conversation

[stoffell]

On Thu, Feb 26, 2009 at 6:08 PM, Simon Dixey wrote:

>  I wonder if anyone is able to offer any [polite ;-)] words of wisdom??
>
I can be polite, I'm not sure about the wisdom .. :-)

> DTMF threshold in misdn-init look high doesn't it...  I'm not entirely sure
> what it "should" be set to, to be honest..  (min-max values for tuning);
> have read max value is 100, but others suggest it'll go higher - but what
> exactly is it tuning the sensitivity value of specifically? (i.e. what is
> the threshold value).  DTMF detection works well for *genuine* DTMF digits
> dialled over the ISDN trunk, but mISDN/Asterisk still recognises them
> incorrectly at times during calls (to/from cell phones).
>
I'm using almost the same setup but I'm in Belgium. Also a b410p. The dtmf
seems to get triggered more by some calls then other. It also depends on the
voice. (higher tones trigger dtmf more easily) My dtmfthreshold is set to
100. Guess it's the (in)sane default ? :-)

Oh, are you having "random" crashes on your mISDN setup too ?

> I've also seen other posts refer to settings in Dahdi.conf (such as
> relaxdtmf) - surely Dahdi doesn't have anything to do with this if I'm using
> chan_misdn??
>
Correct. Only has anything to do with it if you're using chan_dahdi.
I can't reproduce the same behaviour on dahdi.

> Is anyone able to confirm exactly whether mISDN's hardware DSP and driver
> is responsible for detecting DTMF, or whether it's Asterisk analysing the
> inbound audio?  Scanning the README.misdn (sourced separately) the
> chan_misdn driver readme comments a feature as "DTMF Detection in
> HW+mISDNdsp (much better than asterisks internal!)" - so surely DTMF is
> recognised and passed on by mISDN to Asterisk.  The fact that the log
> messages prefixed by P[ 1] are mISDN - I think I've answered my own question
> there...
>
Yes, it's mISDN that detects the dtmf.

> Prior to going down the mISDN route, I looked at Dahdi as the Dahdi configs
> mention native Dahdi B410P support.  But, the conclusion I came to (although
> what I read didn't make it clear me) is that the readme was referring to
> Dahdi B410P support in Ast 1.6, not 1.4.  That sound right?  Dahdi readme:
>
Right again. i've been experimenting with Dahdi's b410p for a while now,
it's only available in asterisk 1.6, with the latest libpri and dahdi
releases.
It's much cleaner, imho, but I'm having issues with receiving faxes when
using dahdi, so I'm stuck with mISDN for the moment :-)

> Enough reading.. if you're still awake!  Any help would be very much
> appreciated.
>
Nice to see someone else is using the same setup. I was beginning to think
that people with BRI stopped using asterisk in Europe :-)

Keep in touch or post to the mailing list if you have any further
news/experiences..


cheers,
stoffell
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Problems with Outbound Calls

[David fire]

sorry but how do you know the warning is from an # ?

he only post this from zaptel.conf

span = 1,1,0,esf,b8zs
bchan = 1-23
dchan = 24
loadzone = jp
defaultzone = jp

David

2009/2/26 Tzafrir Cohen 

> On Thu, Feb 26, 2009 at 11:50:57AM -0200, David fire wrote:
> > hi
> > you should first solve this
> >
> > Warning [2630]: config.c:768 process_text_line: Unknown Directive at
> > line 231 of /etc/asterisk/../zaptel.conf
>
> In zaptel.conf it is perfectly legal to have lines beginning with a '#'.
> With Asterisk '#' is reserved for special directives (currently only
> #include and #exec exist).
>
> The asterisk-gui has a script that creates
>
>  /etc/asterisk/dahdi_guiread.conf
>
> With the following content (if the system is Zaptel and not DAHDI)
>
>  [general]
>  zaptel
>  #include "../zaptel.conf"
>
> This makes it simple to read zaptel.conf through the manager interface
> of reading Asterisk configuration files. It also has several other
> atvantages:
>
> * The user is well aware of its existance if there are any comments in
>  zaptel.conf (the message pointed out above)
> * Its format is not of a real valid configuration file ('zaptel' is a
>  key without a value), which serves to provides interesting chanlanges
>  to configuration parser writiers.
> * It assumes the Asterisk configuration file sits at /etc/asterisk .
>  Indeed why even bother with a situation where Asterisk can only read
>  Zaptel's configuration but not write it?
>
> --
>   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
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
(")_(")signature to help him gain world domination.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] AGI problem using mono (.Net)

[Douglas Mortensen]

An update here. Yesterday's problem has been solved (partly). After
looking closer at the results of my perl script, as well as looking at
the documentation, I realized that we were leaving the escape character
argument off of the STREAM FILE command in the mono application. After
appending "" to the STREAM FILE filename, we are now seeing asterisk
"attempt" to play the file (both in the console and
/var/log/asterisk/full log).

New problem:
==
Although we see the following in our logs / asterisk console:
[Feb 26 11:11:05] VERBOSE[9824] logger.c: -- Playing 'filename'
(escape_digits=) (sample_offset 0)

All that the caller hears is a very brief click. And then the dial plan
continues. This is causing me to wonder whether asterisk halts playback
of the file, if the AGI script that send the STREAM FILE command
completes/returns. Talking to my developer, I asked him to create a loop
after sending the STREAM FILE command and read from stdin until he gets
a string that starts with 200. I'm supposing that asterisk would send
this AFTER the audio file has successfully played out. But again, this
is only a guess.

Thanks so much for the responses you provided yesterday. I look forward
to further information you can assist us with.

Sincerely,
-
Doug Mortensen
Network Consultant
Impala Networks Inc
CCNA, MCP, Security+
Linux+, Network+, A+
.
www.impalanetworks.com
P: (505) 327-7300
F: (505) 327-7545



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] incoming call problem

[michel freiha]

Dear All,
I have created an inbound context in SIP .conf that forward incoming call to
opensips server...The problem appears as soon as I enable t38pt_udptl =
yes...The Asterisk negotiate the SIP session with OpenSIPS without adding
voice codec to INVITE packet...It just contains T.38 protocol...When
t38pt_udptl is disabled everything looks OK and Ulaw is negotiated with
OpenSIPS and cal success..Any suggestion here?

Thanks
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] AGI problem using mono (.Net)

[Douglas Mortensen]

Steve:
=
Thanks for the info on the agi debug command. We'll see what information
we can garner with that. Thanks also for the advanced logging info.

Unfortunately, we are pretty aware of how AGI works (at least at the
level that you explained it). Thanks for the illustrations though.

Also thanks for the info regarding the 1 active request at a time. We
appreciate the information.



Luis:
=
I'm not sure that we can use curl in our situation. We are querying an
Microsoft SQL 2005 database server directly from the asterisk box. It
doesn't look like curl support SQL. Let me know if I am not
understanding your suggestion.

-
Doug Mortensen
Network Consultant
Impala Networks Inc
CCNA, MCP, Security+
Linux+, Network+, A+
.
www.impalanetworks.com
P: (505) 327-7300
F: (505) 327-7545



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

[Olivier]

2009/2/26 Mike 

>  a) PoE powered
>

Unfortunately, I don't think any PoE powered exists, AFAIK.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

[Olivier]

2009/2/26 pe...@networkoblivion.com 

> I used the Patton M-ATA a year or so ago and it was a piece of junk.
> T.38 didn't work and there was no way to troubleshoot why it didn't
> work.


Have you tried again recently ?
I've just tried its T.38 capabilities and I'm not successful yet.


>  Also, the web interface was horrible.
>
> Mike wrote:
> > Hi,
> >
> >
> >
> > I am looking for a good ATA recommendation, ideally something:
> >
> >
> >
> > 1) with one FXS and one LAN port (so it's as inexpensive as possible)
> >
> >
> >
> > 2) That can be provisioning using _FTP_ (configuration and firmware upon
> > reboot, ideally remote reboot from a sip notify)
> >
> >
> >
> > 3) Supports T.38
> >
> >
> >
> > Nice to have would be:
> >
> > a) PoE powered and AC powered (my choice)
> >
> > b) Small size-wise
> >
> >
> >
> > I have been recommended the PAP2T in the past, and although I have used
> > it and sort of liked it, it wasn't possible to provision it using FTP
> > (at least as far as I could tell)
> >
> >
> >
> > Any tip is welcomed.  I`m looking at the Patton ATA which is small, but
> > it doesn't support FTP provisioning either as far as I can tell.
> >
> >
> >
> >
> >
> >
> >
> > Mike
> >
> >
> > 
> >
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Can't build today's AGX Asterisk Addon with spandsp0.0.6pre3 or 4

[Olivier]

I must add I tried spandsp0.0.6xxx as a warning message advised me to do so
(using 0.0.4 would be ok for me but current trunk doesn't allow this
anymore, it seems).


2009/2/26 Olivier 

> Hi,
>
> With 0.0.6pre3:
> # ./build.sh
> CMake Warning (dev) in CMakeLists.txt:
>   No cmake_minimum_required command is present.  A line of code such as
>
> cmake_minimum_required(VERSION 2.6)
>
>   should be added at the top of the file.  The version specified may be
> lower
>   if you wish to support older CMake versions for this project.  For more
>   information run "cmake --help-policy CMP".
> This warning is for project developers.  Use -Wno-dev to suppress it.
>
> -- Configuring done
> -- Generating done
> -- Build files have been written to: /usr/src/agx-ast-addons/trunk
> [ 12%] Built target app_devstate
> [ 25%] Built target app_nv_backgrounddetect
> [ 37%] Built target app_nv_faxdetect
> [ 50%] Built target app_pickup2
> [ 62%] Built target app_valetparking
> [ 75%] Built target func_devstate
> Scanning dependencies of target test_spandsp
> [ 87%] Building C object CMakeFiles/test_spandsp.dir/test_spandsp.o
> Linking C executable test_spandsp
> [ 87%] Built target test_spandsp
> [100%] Building C object spandsp-0.0.6/CMakeFiles/app_fax.dir/app_fax.o
> Linking C shared module ../dist/app_fax.so
> [100%] Built target app_fax
> ./test_spandsp: error while loading shared libraries: libspandsp.so.1:
> cannot open shared object file: No such file or directory
> addons not installed, please upgrade your spandsp library first
>
>
> With 0.0.6pre4 :
> # ./build.sh
> CMake Warning (dev) in CMakeLists.txt:
>   No cmake_minimum_required command is present.  A line of code such as
>
> cmake_minimum_required(VERSION 2.6)
>
>   should be added at the top of the file.  The version specified may be
> lower
>   if you wish to support older CMake versions for this project.  For more
>   information run "cmake --help-policy CMP".
> This warning is for project developers.  Use -Wno-dev to suppress it.
>
> -- Configuring done
> -- Generating done
> -- Build files have been written to: /usr/src/agx-ast-addons/trunk
> [ 12%] Building C object CMakeFiles/app_devstate.dir/app_devstate.o
> Linking C shared module dist/app_devstate.so
> [ 12%] Built target app_devstate
> [ 25%] Building C object
> CMakeFiles/app_nv_backgrounddetect.dir/app_nv_backgrounddetect.o
> Linking C shared module dist/app_nv_backgrounddetect.so
> [ 25%] Built target app_nv_backgrounddetect
> [ 37%] Building C object CMakeFiles/app_nv_faxdetect.dir/app_nv_faxdetect.o
> Linking C shared module dist/app_nv_faxdetect.so
> [ 37%] Built target app_nv_faxdetect
> [ 50%] Building C object CMakeFiles/app_pickup2.dir/app_pickup2.o
> Linking C shared module dist/app_pickup2.so
> [ 50%] Built target app_pickup2
> [ 62%] Building C object CMakeFiles/app_valetparking.dir/app_valetparking.o
> Linking C shared module dist/app_valetparking.so
> [ 62%] Built target app_valetparking
> [ 75%] Building C object CMakeFiles/func_devstate.dir/func_devstate.o
> Linking C shared module dist/func_devstate.so
> [ 75%] Built target func_devstate
> [ 87%] Building C object CMakeFiles/test_spandsp.dir/test_spandsp.o
> Linking C executable test_spandsp
> [ 87%] Built target test_spandsp
> [100%] Building C object spandsp-0.0.6/CMakeFiles/app_fax.dir/app_fax.o
> /usr/src/agx-ast-addons/trunk/spandsp-0.0.6/app_fax.c: In function
> ‘phase_e_handler’:
> /usr/src/agx-ast-addons/trunk/spandsp-0.0.6/app_fax.c:155: error:
> ‘t30_stats_t’ has no member named ‘pages_transferred’
> /usr/src/agx-ast-addons/trunk/spandsp-0.0.6/app_fax.c:184: error:
> ‘t30_stats_t’ has no member named ‘pages_transferred’
> /usr/src/agx-ast-addons/trunk/spandsp-0.0.6/app_fax.c:188: error:
> ‘t30_stats_t’ has no member named ‘pages_transferred’
> /usr/src/agx-ast-addons/trunk/spandsp-0.0.6/app_fax.c:189: error:
> ‘t30_stats_t’ has no member named ‘pages_transferred’
> /usr/src/agx-ast-addons/trunk/spandsp-0.0.6/app_fax.c: In function
> ‘phase_d_handler’:
> /usr/src/agx-ast-addons/trunk/spandsp-0.0.6/app_fax.c:208: error:
> ‘t30_stats_t’ has no member named ‘pages_transferred’
> /usr/src/agx-ast-addons/trunk/spandsp-0.0.6/app_fax.c:211: error:
> ‘t30_stats_t’ has no member named ‘pages_transferred’
> make[2]: *** [spandsp-0.0.6/CMakeFiles/app_fax.dir/app_fax.o] Erreur 1
> make[1]: *** [spandsp-0.0.6/CMakeFiles/app_fax.dir/all] Erreur 2
> make: *** [all] Erreur 2
>
>
> Can anyone help ?
>
> Cheers
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Can't build today's AGX Asterisk Addon with spandsp0.0.6pre3 or 4

[Olivier]

Hi,

With 0.0.6pre3:
# ./build.sh
CMake Warning (dev) in CMakeLists.txt:
  No cmake_minimum_required command is present.  A line of code such as

cmake_minimum_required(VERSION 2.6)

  should be added at the top of the file.  The version specified may be
lower
  if you wish to support older CMake versions for this project.  For more
  information run "cmake --help-policy CMP".
This warning is for project developers.  Use -Wno-dev to suppress it.

-- Configuring done
-- Generating done
-- Build files have been written to: /usr/src/agx-ast-addons/trunk
[ 12%] Built target app_devstate
[ 25%] Built target app_nv_backgrounddetect
[ 37%] Built target app_nv_faxdetect
[ 50%] Built target app_pickup2
[ 62%] Built target app_valetparking
[ 75%] Built target func_devstate
Scanning dependencies of target test_spandsp
[ 87%] Building C object CMakeFiles/test_spandsp.dir/test_spandsp.o
Linking C executable test_spandsp
[ 87%] Built target test_spandsp
[100%] Building C object spandsp-0.0.6/CMakeFiles/app_fax.dir/app_fax.o
Linking C shared module ../dist/app_fax.so
[100%] Built target app_fax
./test_spandsp: error while loading shared libraries: libspandsp.so.1:
cannot open shared object file: No such file or directory
addons not installed, please upgrade your spandsp library first


With 0.0.6pre4 :
# ./build.sh
CMake Warning (dev) in CMakeLists.txt:
  No cmake_minimum_required command is present.  A line of code such as

cmake_minimum_required(VERSION 2.6)

  should be added at the top of the file.  The version specified may be
lower
  if you wish to support older CMake versions for this project.  For more
  information run "cmake --help-policy CMP".
This warning is for project developers.  Use -Wno-dev to suppress it.

-- Configuring done
-- Generating done
-- Build files have been written to: /usr/src/agx-ast-addons/trunk
[ 12%] Building C object CMakeFiles/app_devstate.dir/app_devstate.o
Linking C shared module dist/app_devstate.so
[ 12%] Built target app_devstate
[ 25%] Building C object
CMakeFiles/app_nv_backgrounddetect.dir/app_nv_backgrounddetect.o
Linking C shared module dist/app_nv_backgrounddetect.so
[ 25%] Built target app_nv_backgrounddetect
[ 37%] Building C object CMakeFiles/app_nv_faxdetect.dir/app_nv_faxdetect.o
Linking C shared module dist/app_nv_faxdetect.so
[ 37%] Built target app_nv_faxdetect
[ 50%] Building C object CMakeFiles/app_pickup2.dir/app_pickup2.o
Linking C shared module dist/app_pickup2.so
[ 50%] Built target app_pickup2
[ 62%] Building C object CMakeFiles/app_valetparking.dir/app_valetparking.o
Linking C shared module dist/app_valetparking.so
[ 62%] Built target app_valetparking
[ 75%] Building C object CMakeFiles/func_devstate.dir/func_devstate.o
Linking C shared module dist/func_devstate.so
[ 75%] Built target func_devstate
[ 87%] Building C object CMakeFiles/test_spandsp.dir/test_spandsp.o
Linking C executable test_spandsp
[ 87%] Built target test_spandsp
[100%] Building C object spandsp-0.0.6/CMakeFiles/app_fax.dir/app_fax.o
/usr/src/agx-ast-addons/trunk/spandsp-0.0.6/app_fax.c: In function
‘phase_e_handler’:
/usr/src/agx-ast-addons/trunk/spandsp-0.0.6/app_fax.c:155: error:
‘t30_stats_t’ has no member named ‘pages_transferred’
/usr/src/agx-ast-addons/trunk/spandsp-0.0.6/app_fax.c:184: error:
‘t30_stats_t’ has no member named ‘pages_transferred’
/usr/src/agx-ast-addons/trunk/spandsp-0.0.6/app_fax.c:188: error:
‘t30_stats_t’ has no member named ‘pages_transferred’
/usr/src/agx-ast-addons/trunk/spandsp-0.0.6/app_fax.c:189: error:
‘t30_stats_t’ has no member named ‘pages_transferred’
/usr/src/agx-ast-addons/trunk/spandsp-0.0.6/app_fax.c: In function
‘phase_d_handler’:
/usr/src/agx-ast-addons/trunk/spandsp-0.0.6/app_fax.c:208: error:
‘t30_stats_t’ has no member named ‘pages_transferred’
/usr/src/agx-ast-addons/trunk/spandsp-0.0.6/app_fax.c:211: error:
‘t30_stats_t’ has no member named ‘pages_transferred’
make[2]: *** [spandsp-0.0.6/CMakeFiles/app_fax.dir/app_fax.o] Erreur 1
make[1]: *** [spandsp-0.0.6/CMakeFiles/app_fax.dir/all] Erreur 2
make: *** [all] Erreur 2


Can anyone help ?

Cheers
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] CDR - Asterisk-Stat and PHP5

[Paulo Santos]

Marco Signorini wrote:
> Hi Tiago.
> 
> I've it working on PHP 5.2.6 but only after having modified the php.ini
> default configuration keys:
> 
> zend.ze1_compatibility_mode = Off
> short_open_tag = Off

Though my zend.ze1_compatibility_mode is set to Off, short_open_tag is 
set to On and it is working.

Those are my defaults, at least I never changed them. Installed with 
apt-get on Debian 4.0, PHP version 5.2.0-8+etch13.


-- 
HTML e-mail is evil. Go plain text.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Residential portals and real world scalability

[J. Oquendo]

Hey all, I have a couple of questions.

1) What is the maximum amount of registrations and ongoing
calls you've been able to achieve on your Asterisk systems.
Please do not respond with marketing hyperbole. I'm looking
for real world implementation in the thousands range. For
instance, max I recall having on one souped up server was
a couple hundred registrations with no more than 70 ongoing
calls simultaneously. The farm of Asterisk servers we had
were split to accomodate a couple thousand users.

So again, what's the maximum "effective" amount of users
you've been able to support without coming to a crawling
halt because of a memory leak or other cluster that
brought your system to a reload.

2) What - if any - portal have you come across that gives
a "Vonage" like offering to your end user. Please do not
respond with "Trixbox!" as I'm looking to find something
which can potentially accomodate over 5,000 users to begin
with.

Off-list answers appreciated if need be. Trying to find
something in a turn-key (not turkey) solution to deploy
to scrap something built in house. Would like to be able
to offer residential users the ability to destroy their
dialplans (remote call forwarding, find me follow me,
see/print bill) etc.

I'd appreciate real world experiences and have already
been down the VoIP-Info road looking at the majority of
platforms offered (A2Billing, Supertec, etc.) with no
success.


=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J. Oquendo
SGFA, SGFE, C|EH, CNDA, CHFI, OSCP

"Enough research will tend to support your
conclusions." - Arthur Bloch

"A conclusion is the place where you got
tired of thinking" - Arthur Bloch

227C 5D35 7DCB 0893 95AA  4771 1DCE 1FD1 5CCD 6B5E
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x5CCD6B5E


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

[Jeff LaCoursiere]

On Thu, 26 Feb 2009, Wilton Helm wrote:

> Just beware that most PAP2s on places like E-Bay are from Vonage.  They 
> lock the things up quite seriously.  There are procedures out there to 
> unlock them but it requires stuff like setting up an isolated LAN with a 
> DNS server and FTP server and a special file.  If someone would make a 
> live boot Linux disk that had it all set up it might be a useful 
> service, but in the mean time, it wasn't worth my time, so I just wasted 
> some money.  The one with the letter T on the end is supposed to be 
> general purpose and open.  A lot of people don't know the difference 
> until after they've wasted their money.  Also they make the mistake of 
> putting the thing on line immediately and it goes to Vonage and get the 
> latest upgrade, making it even harder to unlock.
>
> Wilton
>

Actually you want the one with "-NA" at the end.  That means it is 
unlocked and open.  So look specifically for "Linksys PAP2T-NA".

If it has "-R" on the end it is made for a specific ITSP.

j

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

[Wilton Helm]

Just beware that most PAP2s on places like E-Bay are from Vonage.  They lock 
the things up quite seriously.  There are procedures out there to unlock them 
but it requires stuff like setting up an isolated LAN with a DNS server and FTP 
server and a special file.  If someone would make a live boot Linux disk that 
had it all set up it might be a useful service, but in the mean time, it wasn't 
worth my time, so I just wasted some money.  The one with the letter T on the 
end is supposed to be general purpose and open.  A lot of people don't know the 
difference until after they've wasted their money.  Also they make the mistake 
of putting the thing on line immediately and it goes to Vonage and get the 
latest upgrade, making it even harder to unlock.

Wilton
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Dictate

[amit mehta]

Hi Brent,
Thanks for the help.

I am working to achieve the middle solution.

The scenario that i am trying to achieve :

Customer calls in the tollfree number:
Enter the ID and press"#"
Pwd: #
Press * to begin dictate,4 to pause and 2 to start again.
PRess * to end and start new

Upto this has been achieved using Dictate Application.

Now the file is saved in .raw format and when converted to .wav format
doesnt give quality output and just makes loud noise.

Moreover i want to get a jobid  at the end of the dictation.Also want to
provide pause,rewind and fastforward to it.I want to give some tags to file
when it is stored like authid,name,time.

Thanks for all the suggestions you are going to give.

Regards,
Amit

On Thu, Feb 26, 2009 at 9:53 PM, Brent Davidson  wrote:

>
>
> amit mehta wrote:
> > Hello Members,
> >
> > Sorry for hijacking the earlier thread and asking the question last time.
> >
> > Is anyone aware about a solution to call incoming number and dictate
> > the  files by using Dictate feature of Asterisk used for Medical
> > Transcription industry.
> >
> > Thanks & Regards,
> > Amit Mehta
> I'm not quite sure exactly what you're asking, so I'll cover what I see
> as answers to three possible scenarios.
>
> If all you want to do is read out the contents of text files, then look
> at the Cepstral text to speech engine.  It would be fairly simple to
> build a script that parsed a list of files, read some form of numeric
> identifier to the user that allowed them to select a file to be read,
> then the file is sent to the Cepstral (app_swift) module and the
> contents of the file are read back to the user.  You would probably need
> to implement some sort of pause, go back 1 sentence, go back one
> paragraph, etc controls as well.
>
> If you're looking for a way for a user to call in and record a voice
> file that will be later transcribed, that is probably easier than the
> reading a text file back.  Just set up a macro that prompts for the
> callers ID, Patient ID, or whatever info you need using the "Read"
> function, then use the "Record" function to record the fileand save it
> with the previously gathered info in the filename (easiest solution) or
> store the recording and all the other info in a database (A bit more
> complicated).
>
> If you're looking for a way to allow a caller to read some information
> and have the system save that as a text file, then you'll need to talk
> to someone with more knowledge than me.  Speech recognition isn't very
> easy right now.
>
> Best of luck,
> -Brent
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Odd Read App Issues

[Robert Broyles]

Yea, I tried that too. I have it: dtmfmode=rfc2833

--
Regards,
Robert Broyles


Brent Davidson wrote:

Robert Broyles wrote:

Okay. I'm using this all over SIP Trunking with Vitelity.
Any other suggestions?
--
Regards,
Robert Broyles

  



Eric Wieling, Asteria Solutions Group wrote:

Robert Broyles wrote:
  
So I'm using the READ() application within an IVR, and having a strange 
issue, and wondering if anyone else has had this problem.


When calling from an outside line, and entering the digits during the 
read() part of my dialplan, it's accepting some of the digits twice, 
though it's only keyed in once.


When testing the dialplan internally, it accepts only the digits that I 
key in.


Anyone else experienced this?



Yes.  Most of the time it is either because I put relaxdtmf=yes in 
zapata.conf or because my rxgain is too low on that port.



I've seen an issue similar to this when the sip peer was providing 
DTMF over multiple encodings at the same time.  Usually, it's when 
Asterisk is expecting DTMF via inband, but the peer is sending inband 
and either INFO or rfc2833.  What do you have the dtmfmode= line set 
to in your sip.conf?



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] DTMF tones mid conversation

[Simon Dixey]

Hi list,



I wonder if anyone is able to offer any [polite ;-)] words of wisdom??



I'm having the same problem too, with random DTMF heard during ISDN calls.  Am
running Asterisk 1.4.22 and mISDN 1.1.8. (kernel 2.6.18 CentOS)



Symptoms are on incoming and out outgoing ISDN calls, particularly to mobile
(cell) phones or those of a low quality/compressed audio.  I'm using a
Digium B410P (only using 1 port; ISDN 2e in TE mode) connected to the British 
Telecom PSTN at the NT.



I've changed my misdn-init.conf about many times (dtmfthreshold and recently
added dtmf to card= & restart) without any noticeable difference. 



misdn-init.conf:



card=1,0x4,dtmf

te_ptp=1

nt_ptp=2,3,4

option=1,master_clock

dsp_options=0

dtmfthreshold=2500

timer=1/etc/asterisk/misdn.conf:[general]
misdn_init=/etc/misdn-init.conf
debug=4
ntdebugflags=0
ntdebugfile=/var/log/misdn-nt.log
ntkeepcalls=no
tracefile=/var/log/asterisk/misdn.log
bridging=yes

stop_tone_after_first_digit=yes
append_digits2exten=yes

dynamic_crypt=no
crypt_prefix=**
crypt_keys=test,muh

[default]
context=default
language=en
musicclass=default
senddtmf=yes
far_alerting=yes
allowed_bearers=all
nationalprefix=0
internationalprefix=00
rxgain=0
txgain=0
te_choose_channel=no
pmp_l1_check=no
reject_cause=16
need_more_infos=yes
nttimeout=no
method=standard
overlapdial=no
dialplan=0
localdialplan=2
cpndialplan=0
early_bconnect=yes
incoming_early_audio=no
nodialtone=no
presentation=0
screen=0
echocancel=yes
jitterbuffer=4000
jitterbuffer_upper_threshold=0
hdlc=no
max_incoming=-1
max_outgoing=-1

[isdn]
ports=1
context=from-pstn

DTMF threshold in misdn-init look high doesn't it...  I'm not entirely sure 
what it
"should" be set to, to be honest..  (min-max values for tuning); have read max
value is 100, but others suggest it'll go higher - but what exactly is it
tuning the sensitivity value of specifically? (i.e. what is the threshold
value).  DTMF detection works well for *genuine* DTMF digits dialled over
the ISDN trunk, but mISDN/Asterisk still recognises them incorrectly at times
during calls (to/from cell phones).



I've also seen other posts refer to settings in Dahdi.conf (such as relaxdtmf)
- surely Dahdi doesn't have anything to do with this if I'm using chan_misdn??



As a side note, interestingly, some withheld calls present their incoming
caller ID as '0', others 'unknown'... Not got to the bottom of that one
either.  Anyway.. back on topic..



Here's an excerpt from my asterisk 'full' log showing a random DTMF
digit.  This spurious digit was incorrectly recognised from voice: on a
mobile phone call (outgoing this time - yes could turn DTMF detection off on
the dial statement but doesn't help on incoming):



[Feb 26 13:47:38] VERBOSE[20051] logger.c: -- Executing
[...@macro-dialout-trunk:26] Dial("SIP/101-09ff5c68",
"misdn/1/07")

[Feb 26 13:47:38] VERBOSE[20051] logger.c: -- Called
1/07

[Feb 26 13:47:39] VERBOSE[20051] logger.c: --
mISDN/1-u364 is proceeding passing it to SIP/101-09ff5c68

[Feb 26 13:47:43] VERBOSE[20051] logger.c: --
mISDN/1-u365 is ringing

[Feb 26 13:47:46] VERBOSE[20051] logger.c: --
mISDN/1-u365 answered SIP/101-09ff5c68

[Feb 26 13:49:14] DTMF[20051] channel.c: DTMF end '2' received on mISDN/1-u365,
duration 0 ms

[Feb 26 13:49:14] DTMF[20051] channel.c: DTMF begin emulation of '2' with
duration 100 queued on mISDN/1-u365

[Feb 26 13:49:14] DTMF[20051] channel.c: DTMF end emulation of '2' queued on
mISDN/1-u365

[Feb 26 13:50:54] DEBUG[20051] chan_misdn.c: misdn_hangup(mISDN/1-u365)

[Feb 26 13:50:54] VERBOSE[20051] logger.c: -- Executing
[...@macro-hangupcall:11] Hangup("SIP/101-09ff5c68", "") in
new stack



Apologies for the trail of logs; just noticed this happening on my Asterisk
console on an incoming call from a mobile:



P[ 1]  --> * IND :  -1! (stop indication)
pid:168

P[ 1] * ANSWER:

P[ 1]  --> empty cad using dad

P[ 1] I SEND:CONNECT oad:07x dad:781890 pid:168

P[ 1]  --> channel:2 mode:TE cause:16 ocause:16 rad: cad:781890

P[ 1]  --> info_dad: onumplan:2 dnumplan:0 rnumplan:  cpnnumplan:0

P[ 1]  --> * Unknown Indication:20 pid:168

P[ 1] I IND :CONNECT_ACKNOWLEDGE  oad:07 dad:78xxx pid:168
state:CONNECTED

P[ 1]  --> channel:2 mode:TE cause:16 ocause:16 rad: cad:781890

P[ 1]  --> info_dad: onumplan:2 dnumplan:0 rnumplan:  cpnnumplan:0

P[ 1] I IND :DTMF_TONE oad:07 dad:78xxx pid:168 state:CONNECTED

P[ 1]  --> channel:2 mode:TE cause:16 ocause:16 rad: cad:781890

P[ 1]  --> info_dad: onumplan:2 dnumplan:0 rnumplan:  cpnnumplan:0

P[ 1]  --> DTMF:1

[Feb 26 16:41:47] DTMF[21144]: channel.c:2148 __ast_read: DTMF end '1' received
on mISDN/2-u397, duration 0 ms

[Feb 26 16:41:47] DTMF[21144]: channel.c:2184 __ast_read: DTMF begin emulation
of '1' with duration 100 queued on mISDN/2-u397

P[ 1]  --> * Unknown Indication:20 pid:168

P[ 1]  --> * Unknown Indication:20 pid:168

[Feb 26 16:41:47] DTMF[21144]: channel.c:2296 __ast_read: DTMF end emulation of

Re: [asterisk-users] Odd Read App Issues

[Brent Davidson]

Robert Broyles wrote:

Okay. I'm using this all over SIP Trunking with Vitelity.
Any other suggestions?
--
Regards,
Robert Broyles

  



Eric Wieling, Asteria Solutions Group wrote:

Robert Broyles wrote:
  
So I'm using the READ() application within an IVR, and having a strange 
issue, and wondering if anyone else has had this problem.


When calling from an outside line, and entering the digits during the 
read() part of my dialplan, it's accepting some of the digits twice, 
though it's only keyed in once.


When testing the dialplan internally, it accepts only the digits that I 
key in.


Anyone else experienced this?



Yes.  Most of the time it is either because I put relaxdtmf=yes in 
zapata.conf or because my rxgain is too low on that port.



I've seen an issue similar to this when the sip peer was providing DTMF 
over multiple encodings at the same time.  Usually, it's when Asterisk 
is expecting DTMF via inband, but the peer is sending inband and either 
INFO or rfc2833.  What do you have the dtmfmode= line set to in your 
sip.conf?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] CDR - Asterisk-Stat and PHP5

[Marco Signorini]

Hi Tiago.

I've it working on PHP 5.2.6 but only after having modified the php.ini
default configuration keys:

zend.ze1_compatibility_mode = Off
short_open_tag = Off

setting together to On and restarting apache forces PHP5 to behave like
PHP 4.x version.

regards,
Marco Signorini

===
INGEGNI Tech S.r.l.
http://www.ingegnitech.com



Paulo Santos wrote:
> Tiago Durante wrote:
>   
>> Hi all,
>>
>> I don't know if its the right place to ask, but... Does any one have
>> the asterisk-stat-v2 running with PHP5?
>>
>>
>> Tks!
>>
>>
>> 
>
> # php --version
>
> PHP 5.2.0-8+etch13 (cli) (built: Oct  2 2008 08:26:18)
> Copyright (c) 1997-2006 The PHP Group
> Zend Engine v2.2.0, Copyright (c) 1998-2006 Zend Technologies
>
> Working for me. Don't forget you need php5-gd for the graphics to show.
>
>   


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Getting SIP field P-Asserted-Identity from EAGI

[equis software]

Thanks to all, I'll try it!


On Thu, Feb 26, 2009 at 12:25 PM, Danny Nicholas  wrote:

>  Based on this page -
> http://blog.herbertm.ca/2007/09/03/extracting-dids-from-the-sip-header
>
>
>
> You could put this in your dialplan
>
>
>
> [ext-did-custom]
>
> exten => s,1,Set(ASSERT="${SIP_HEADER(Call-ID)}")
>
> exten => s,2,AGI(xxx.pl,${ASSERT})
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *equis software
> *Sent:* Thursday, February 26, 2009 7:37 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Getting SIP field P-Asserted-Identity from
> EAGI
>
>
>
> Hi, using EAGI variables like
>
> agi_request
>
> agi_channel
>
> agi_language
>
> agi_type
>
> agi_uniqueid
>
> agi_callerid
>
> agi_dnid
>
> agi_rdnis
>
> agi_context
>
> agi_extension
>
> agi_priority
>
> agi_enhanced
>
> agi_accountcode
>
>I get a lot of data about a call, but I need to obtain
> P-Asserted-Identity value from a SIP call. Are tehe any eagi variable to get
> that? Or have you any solution??
>
>
> Thanks!!!
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Odd Read App Issues

[Robert Broyles]

Not at all.
In fact, I found that relaxdtmf=yes is now available for sip.conf as of 
1.4 as well.

However, that didn't resolve the problem.

--
Regards,
Robert Broyles




Eric Wieling, Asteria Solutions Group wrote:

Robert Broyles wrote:
  

Okay. I'm using this all over SIP Trunking with Vitelity.
Any other suggestions?



Sorry for wasting your time.


  




DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] DID's in a specific rate center

[Wilton Helm]

>> What is illegal is to set caller-id to a fraudulent value such that the 
>> person on the other end will not be able to correctly identify the 
>> originator of the call.


>I don't know if there is anything that falls under the FCC rules.  In any 
>event it 
>would be unethical and evidence of fraudulent intent if one was trying to 
>defraud someone in the process of doing so.

Another case is in telemarketing.  FCC rules require a caller-ID be present and 
identify a phone number where a person can request to be added to a do not call 
list.  I am filing a complaint against a firm at present that provides a 
caller-ID of a non-working number!

Wilton
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Dictate

[Brent Davidson]

amit mehta wrote:
> Hello Members,
>
> Sorry for hijacking the earlier thread and asking the question last time.
>
> Is anyone aware about a solution to call incoming number and dictate 
> the  files by using Dictate feature of Asterisk used for Medical
> Transcription industry.
>
> Thanks & Regards,
> Amit Mehta
I'm not quite sure exactly what you're asking, so I'll cover what I see 
as answers to three possible scenarios.

If all you want to do is read out the contents of text files, then look 
at the Cepstral text to speech engine.  It would be fairly simple to 
build a script that parsed a list of files, read some form of numeric 
identifier to the user that allowed them to select a file to be read, 
then the file is sent to the Cepstral (app_swift) module and the 
contents of the file are read back to the user.  You would probably need 
to implement some sort of pause, go back 1 sentence, go back one 
paragraph, etc controls as well.

If you're looking for a way for a user to call in and record a voice 
file that will be later transcribed, that is probably easier than the 
reading a text file back.  Just set up a macro that prompts for the 
callers ID, Patient ID, or whatever info you need using the "Read" 
function, then use the "Record" function to record the fileand save it 
with the previously gathered info in the filename (easiest solution) or 
store the recording and all the other info in a database (A bit more 
complicated).

If you're looking for a way to allow a caller to read some information 
and have the system save that as a text file, then you'll need to talk 
to someone with more knowledge than me.  Speech recognition isn't very 
easy right now.

Best of luck,
-Brent


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Getting SIP field P-Asserted-Identity from EAGI

[Tilghman Lesher]

On Thursday 26 February 2009 08:39:29 Benny Amorsen wrote:
> Can you access channel functions from EAGI?
>
> SIP_HEADER(P-Asserted-Identity) contains what you need.
>
> It would be nice if Asterisk supported it natively and forgot about
> the deprecated Remote-Party-ID header.

Of course you can.

GET VARIABLE SIP_HEADER(P-Asserted-Identity)

-- 
Tilghman

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Odd Read App Issues

[Eric Wieling, Asteria Solutions Group]

Robert Broyles wrote:
> Okay. I'm using this all over SIP Trunking with Vitelity.
> Any other suggestions?

Sorry for wasting your time.


-- 
Eric Wieling * Asteria Solutions Group * Huntsville, AL
Call centers * IVRs * Enterprise PBXs * Conferencing applications
256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

[Jeff LaCoursiere]

On Thu, 26 Feb 2009, Mike wrote:

> Hi,
>
>
>
> I am looking for a good ATA recommendation, ideally something:
>
>
>
> 1) with one FXS and one LAN port (so it's as inexpensive as possible)
>
>
>
> 2) That can be provisioning using FTP (configuration and firmware upon
> reboot, ideally remote reboot from a sip notify)
>
>
>
> 3) Supports T.38
>
>
>
> Nice to have would be:
>
> a) PoE powered and AC powered (my choice)
>
> b) Small size-wise
>
>
>
> I have been recommended the PAP2T in the past, and although I have used it
> and sort of liked it, it wasn't possible to provision it using FTP (at least
> as far as I could tell)

I second the PAP2T.  No PoE though.  I am almost certain it can be 
provisioned by FTP.  I use TFTP personally...  on't think you can get one 
much cheaper either.

j

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] [cdr_odbc] error: Cannot insert the value NULL into column 'calldate'

[Rajkumar S]

Hi,

I am trying to get * log to mssql server. I have odbc and freetds
configured, but my insert query is missing calldate which is a NOT
NULL field in database schema.

cdr_adaptive_odbc: Insert failed on 'sqlserver:cdr'.  CDR failed:
INSERT INTO cdr
(clid,src,dst,dcontext,channel,lastapp,lastdata,duration,billsec,disposition,amaflags,uniqueid)
VALUES 
('1000','1000','100','sip','SIP/1000-09388800','AddQueueMember','test,Local/1...@sip,5,,Agent/1000,Local/1...@sip',9,9,'ANSWERED',3,'1235681438.1')

Schema is:

My table structure is:

CREATE TABLE cdr (
[calldate]  [datetime]  NOT NULL ,
[clid]  [varchar] (80)  NOT NULL ,
[src]   [varchar] (80)  NOT NULL ,
[dst]   [varchar] (80)  NOT NULL ,
[dcontext]  [varchar] (80)  NOT NULL ,
[channel]   [varchar] (80)  NOT NULL ,
[dstchannel][varchar] (80)  NOT NULL ,
[lastapp]   [varchar] (80)  NOT NULL ,
[lastdata]  [varchar] (80)  NOT NULL ,
[duration]  [int]   NOT NULL ,
[billsec]   [int]   NOT NULL ,
[disposition]   [varchar] (45)  NOT NULL ,
[amaflags]  [int]   NOT NULL ,
[accountcode]   [varchar] (20)  NOT NULL ,
[uniqueid]  [varchar] (32)  NOT NULL ,
[userfield] [varchar] (255) NOT NULL
)

How can I make sure that my insert commands reflects my database schema?

I have attached details of all my config files below.

I can connect using isql.

isql -v DSN_NAME sa password
+---+
| Connected!|
|   |
| sql-statement |
| help [tablename]  |
| quit  |
|   |
+---+

When I make a call I get the following error in console:

Unable to retrieve database handle.  CDR failed.
SQL Execute returned an error -1: 23000: [FreeTDS][SQL Server]Cannot
insert the value NULL into column 'calldate', table
'production.dbo.cdr'; column does not allow nulls. INSERT fails. (153)
SQL Execute returned an error -1: 01000: [FreeTDS][SQL Server]The
statement has been terminated. (55)
SQL Execute error -1! Attempting a reconnect...
Connection is down attempting to reconnect...
Disconnected 0 from sqlserver [DSN_NAME]
Database handle deallocated
Connecting sqlserver
res_odbc: Connected to sqlserver [DSN_NAME]
SQL Execute returned an error -1: 23000: [FreeTDS][SQL Server]Cannot
insert the value NULL into column 'calldate', table
'production.dbo.cdr'; column does not allow nulls. INSERT fails. (153)
SQL Execute returned an error -1: 01000: [FreeTDS][SQL Server]The
statement has been terminated. (55)
SQL Execute error -1! Attempting a reconnect...
Connection is down attempting to reconnect...
Disconnected 0 from sqlserver [DSN_NAME]
Database handle deallocated
Connecting sqlserver
res_odbc: Connected to sqlserver [DSN_NAME]
cdr_adaptive_odbc: Insert failed on 'sqlserver:cdr'.  CDR failed:
INSERT INTO cdr
(clid,src,dst,dcontext,channel,lastapp,lastdata,duration,billsec,disposition,amaflags,uniqueid)
VALUES 
('1000','1000','100','sip','SIP/1000-09388800','AddQueueMember','test,Local/1...@sip,5,,Agent/1000,Local/1...@sip',9,9,'ANSWERED',3,'1235681438.1')


a16-q1:/etc/asterisk# cat cdr.conf
[general]
enable=yes
;unanswered = no
;batch=no
;size=100
;time=300
;scheduleronly=no
;safeshutdown=yes
;endbeforehexten=no

[csv]
usegmtime=yes; log date/time in GMT.  Default is "no"
loguniqueid=yes  ; log uniqueid.  Default is "no"
loguserfield=yes ; log user field.  Default is "no"

[odbc]
usegmtime=yes; log date/time in GMT.  Default is "no"
loguniqueid=yes  ; log uniqueid.  Default is "no"
loguserfield=yes ; log user field.  Default is "no"

;
; cdr_odbc.conf
;

[global]
username => sa
password => password
dsn => DSN_NAME
loguniqueid=yes
dispositionstring=yes
table=cdr
usegmtime=yes

a16-q1:/etc/asterisk# cat cdr_adaptive_odbc.conf
[first]
connection=sqlserver
table=cdr
alias calldate => start

a16-q1:/etc/asterisk# cat res_odbc.conf
[ENV]

[sqlserver]
enabled => yes
dsn => DSN_NAME
share_connections => no
limit => 5
username => sa
password => password
pre-connect => yes
sanitysql => select count(*) from systables
backslash_is_escape => no

a16-q1:/etc/asterisk# ls
asterisk.conf   cdr.conf   chan_dahdi.conf
extensions.conf  iax.conf  modules.conf  queuerules.conf
res_odbc.conf
cdr_adaptive_odbc.conf  cdr_odbc.c

Re: [asterisk-users] Odd Read App Issues

[Robert Broyles]

Okay. I'm using this all over SIP Trunking with Vitelity.
Any other suggestions?

--
Regards,
Robert Broyles




Eric Wieling, Asteria Solutions Group wrote:

Robert Broyles wrote:
  
So I'm using the READ() application within an IVR, and having a strange 
issue, and wondering if anyone else has had this problem.


When calling from an outside line, and entering the digits during the 
read() part of my dialplan, it's accepting some of the digits twice, 
though it's only keyed in once.


When testing the dialplan internally, it accepts only the digits that I 
key in.


Anyone else experienced this?



Yes.  Most of the time it is either because I put relaxdtmf=yes in 
zapata.conf or because my rxgain is too low on that port.


  




DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Problems with Outbound Calls

[Tzafrir Cohen]

On Thu, Feb 26, 2009 at 11:50:57AM -0200, David fire wrote:
> hi
> you should first solve this
> 
> Warning [2630]: config.c:768 process_text_line: Unknown Directive at
> line 231 of /etc/asterisk/../zaptel.conf

In zaptel.conf it is perfectly legal to have lines beginning with a '#'. 
With Asterisk '#' is reserved for special directives (currently only
#include and #exec exist).

The asterisk-gui has a script that creates

  /etc/asterisk/dahdi_guiread.conf

With the following content (if the system is Zaptel and not DAHDI)

  [general]
  zaptel
  #include "../zaptel.conf"

This makes it simple to read zaptel.conf through the manager interface
of reading Asterisk configuration files. It also has several other
atvantages:

* The user is well aware of its existance if there are any comments in
  zaptel.conf (the message pointed out above)
* Its format is not of a real valid configuration file ('zaptel' is a
  key without a value), which serves to provides interesting chanlanges
  to configuration parser writiers.
* It assumes the Asterisk configuration file sits at /etc/asterisk .
  Indeed why even bother with a situation where Asterisk can only read
  Zaptel's configuration but not write it?

-- 
   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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Odd Read App Issues

[Eric Wieling, Asteria Solutions Group]

Robert Broyles wrote:
> So I'm using the READ() application within an IVR, and having a strange 
> issue, and wondering if anyone else has had this problem.
> 
> When calling from an outside line, and entering the digits during the 
> read() part of my dialplan, it's accepting some of the digits twice, 
> though it's only keyed in once.
> 
> When testing the dialplan internally, it accepts only the digits that I 
> key in.
> 
> Anyone else experienced this?

Yes.  Most of the time it is either because I put relaxdtmf=yes in 
zapata.conf or because my rxgain is too low on that port.

-- 
Eric Wieling * Asteria Solutions Group * Huntsville, AL
Call centers * IVRs * Enterprise PBXs * Conferencing applications
256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

[Benny Amorsen]

Mike  writes:

> 2) That can be provisioning using FTP (configuration and firmware upon
> reboot, ideally remote reboot from a sip notify)

What's wrong with HTTP?

> I have been recommended the PAP2T in the past, and although I have used it
> and sort of liked it, it wasn't possible to provision it using FTP (at
> least as far as I could tell)

I haven't tried FTP (why does anyone use this protocol anymore? TFTP I
can understand, but FTP?!). HTTP works fine.


/Benny


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Getting SIP field P-Asserted-Identity from EAGI

[Benny Amorsen]

Can you access channel functions from EAGI?

SIP_HEADER(P-Asserted-Identity) contains what you need.

It would be nice if Asterisk supported it natively and forgot about
the deprecated Remote-Party-ID header.


/Benny


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Odd Read App Issues

[Robert Broyles]

Btw, I'm using Asterisk SVN-branch-1.4-r178640


Robert Broyles wrote:
> So I'm using the READ() application within an IVR, and having a strange 
> issue, and wondering if anyone else has had this problem.
>
> When calling from an outside line, and entering the digits during the 
> read() part of my dialplan, it's accepting some of the digits twice, 
> though it's only keyed in once.
>
> When testing the dialplan internally, it accepts only the digits that I 
> key in.
>
> Anyone else experienced this?
>
>   


DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6

[Benny Amorsen]

Klaus Darilion  writes:

> What about a config option
>gototriggersinvalid=yes (default=no)
> in extensions.conf for users which are using this feature?

Please, no more options. There are way too many options already.

Since I personally believe the use of the special extensions should be
limited as much as possible, I believe Goto and Gosub should not end
up in the i extension. I can live with the opposite as well, even if
it isn't what I prefer. Just not another option.


/Benny


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Odd Read App Issues

[Robert Broyles]

So I'm using the READ() application within an IVR, and having a strange 
issue, and wondering if anyone else has had this problem.

When calling from an outside line, and entering the digits during the 
read() part of my dialplan, it's accepting some of the digits twice, 
though it's only keyed in once.

When testing the dialplan internally, it accepts only the digits that I 
key in.

Anyone else experienced this?

-- 
Regards,
Robert Broyles




DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

[Mike]

Thanks for the comment, it's good to know.

Mike

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of pe...@networkoblivion.com
> Sent: Thursday, February 26, 2009 9:16
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] ATA recommendation (wih FTP provisioning)
> 
> I used the Patton M-ATA a year or so ago and it was a piece of junk.
> T.38 didn't work and there was no way to troubleshoot why it didn't
> work.  Also, the web interface was horrible.
> 
> Mike wrote:
> > Hi,
> >
> >
> >
> > I am looking for a good ATA recommendation, ideally something:
> >
> >
> >
> > 1) with one FXS and one LAN port (so it's as inexpensive as possible)
> >
> >
> >
> > 2) That can be provisioning using _FTP_ (configuration and firmware upon
> > reboot, ideally remote reboot from a sip notify)
> >
> >
> >
> > 3) Supports T.38
> >
> >
> >
> > Nice to have would be:
> >
> > a) PoE powered and AC powered (my choice)
> >
> > b) Small size-wise
> >
> >
> >
> > I have been recommended the PAP2T in the past, and although I have used
> > it and sort of liked it, it wasn't possible to provision it using FTP
> > (at least as far as I could tell)
> >
> >
> >
> > Any tip is welcomed.  I`m looking at the Patton ATA which is small, but
> > it doesn't support FTP provisioning either as far as I can tell.
> >
> >
> >
> >
> >
> >
> >
> > Mike
> >
> >
> > 
> >
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Getting SIP field P-Asserted-Identity from EAGI

[Danny Nicholas]

Based on this page -
http://blog.herbertm.ca/2007/09/03/extracting-dids-from-the-sip-header

 

You could put this in your dialplan 

 

[ext-did-custom]
exten => s,1,Set(ASSERT="${SIP_HEADER(Call-ID)}")

exten => s,2,AGI(xxx.pl,${ASSERT})

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of equis software
Sent: Thursday, February 26, 2009 7:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Getting SIP field P-Asserted-Identity from EAGI

 

Hi, using EAGI variables like 

agi_request


agi_channel


agi_language


agi_type


agi_uniqueid


agi_callerid


agi_dnid


agi_rdnis


agi_context


agi_extension


agi_priority


agi_enhanced


agi_accountcode







  I get a lot of data about a call, but I need to obtain P-Asserted-Identity
value from a SIP call. Are tehe any eagi variable to get that? Or have you
any solution??


Thanks!!!

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

[pe...@networkoblivion.com]

I used the Patton M-ATA a year or so ago and it was a piece of junk. 
T.38 didn't work and there was no way to troubleshoot why it didn't 
work.  Also, the web interface was horrible.

Mike wrote:
> Hi,
> 
>  
> 
> I am looking for a good ATA recommendation, ideally something:
> 
>  
> 
> 1) with one FXS and one LAN port (so it's as inexpensive as possible)
> 
>  
> 
> 2) That can be provisioning using _FTP_ (configuration and firmware upon 
> reboot, ideally remote reboot from a sip notify)
> 
>  
> 
> 3) Supports T.38
> 
>  
> 
> Nice to have would be:
> 
> a) PoE powered and AC powered (my choice)
> 
> b) Small size-wise
> 
>  
> 
> I have been recommended the PAP2T in the past, and although I have used 
> it and sort of liked it, it wasn't possible to provision it using FTP 
> (at least as far as I could tell)
> 
>  
> 
> Any tip is welcomed.  I`m looking at the Patton ATA which is small, but 
> it doesn't support FTP provisioning either as far as I can tell.
> 
>  
> 
>  
> 
>  
> 
> Mike
> 
> 
> 
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Problems with Outbound Calls

[David fire]

hi
you should first solve this

Warning [2630]: config.c:768 process_text_line: Unknown Directive at
line 231 of /etc/asterisk/../zaptel.conf

check what do you have in the line 231 of your zaptel.conf file.

David

2009/2/26 Wye-khe Kwok 

> Hi everyone!
>
> I'm quite a newbie at this Asterisk stuff so please bear with me.
>
> We've recently decided to start training in Asterisk via AsteriskNow!
>
> Asterisk version is 1.4.18.1 through AsteriskNow! 1.02
>
> The box we have is paired with a Digium TE110P and we've managed to get
> it to the point where incoming calls via a single DID (from NTT Japan)
> can be received and answered (INS1500 here in Japan). We're using SIP
> phones here.
>
> However, on attempting outbound calls, I've noticed the following
> message on the Live Console.
>
> Warning [2630]: config.c:768 process_text_line: Unknown Directive at
> line 231 of /etc/asterisk/../zaptel.conf
>
> The phones have no dial tones and we get nothing but silence when
> dialing no.s and hitting the 'send' button.
>
> Following are some excerpts from the Conf Files (Sorry about the spam -
> I'm not sure what's redundant)
>
> ---
>
> Sip.Conf: (Only included info that didn't start with semicolons)
>
> [general]
> context=default ; Default context for incoming calls
> allowoverlap=no ; Disable overlap dialing support.
> (Default is yes)
> bindport=5060   ; UDP Port to bind to (SIP standard port
> is 5060)
> bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds
> to all)
> srvlookup=yes   ; Enable DNS SRV lookups on outbound
> calls
>
> language=jp ; Default language setting for all
> users/peers
>; This may also be set for individual
> users/peers
>
>
> Extensions.Conf:
>
> [general]
> static = yes
> writeprotect = no
> autofallthrough = yes
> clearglobalvars = no
> priorityjumping = no
>
> [globals]
> span_1 = Zap/g1
>
> [dundi-e164-canonical]
>
> [dundi-e164-customers]
>
> [dundi-e164-via-pstn]
>
> [dundi-e164-local]
> include => dundi-e164-canonical
> include => dundi-e164-customers
> include => dundi-e164-via-pstn
>
> [dundi-e164-switch]
> switch => DUNDi/e164
>
> [dundi-e164-lookup]
> include => dundi-e164-local
> include => dundi-e164-switch
>
> [macro-dundi-e164]
> exten => s,1,Goto(${ARG1},1)
> include => dundi-e164-lookup
>
> [iaxtel700]
> exten =>
> _91700XXX,1,Dial(IAX2/${iaxin...@iaxtel.com/${EXTEN:1...@iaxtel
> )
>
> [iaxprovider]
>
> [trunkint]
> exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
> exten => _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
>
> [trunkld]
> exten => _91NXXNXX,1,Macro(dundi-e164,${EXTEN:1})
> exten => _91NXXNXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
>
> [trunklocal]
> exten => _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
>
> [trunktollfree]
> exten => _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
>
> [international]
> ignorepat => 9
> include => longdistance
> include => trunkint
>
> [longdistance]
> ignorepat => 9
> include => local
> include => trunkld
>
> [local]
> ignorepat => 9
> include => default
> include => parkedcalls
> include => trunklocal
> include => iaxtel700
> include => trunktollfree
> include => iaxprovider
>
> [macro-stdexten]
> exten => s,1,Dial(${ARG2},20)
> exten => s,2,Goto(s-${DIALSTATUS},1)
> exten => s-NOANSWER,1,Voicemail(${ARG1},u)
> exten => s-NOANSWER,2,Goto(default,s,1)
> exten => s-BUSY,1,Voicemail(${ARG1},b)
> exten => s-BUSY,2,Goto(default,s,1)
> exten => _s-.,1,Goto(s-NOANSWER,1)
> exten => a,1,VoicemailMain(${ARG1})
>
> [macro-stdPrivacyexten]
> exten => s,1,Dial(${ARG2},20|p)
> exten => s,2,Goto(s-${DIALSTATUS},1)
> exten => s-NOANSWER,1,Voicemail(u${ARG1})
> exten => s-NOANSWER,2,Goto(default,s,1)
> exten => s-BUSY,1,Voicemail(b${ARG1})
> exten => s-BUSY,2,Goto(default,s,1)
> exten => s-DONTCALL,1,Goto(${ARG3},s,1)
> exten => s-TORTURE,1,Goto(${ARG4},s,1)
> exten => _s-.,1,Goto(s-NOANSWER,1)
> exten => a,1,VoicemailMain(${ARG1})
>
> [macro-page]
> exten => s,1,ChanIsAvail(${ARG1}|js)
> exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail)
> exten => s,n(autoanswer),Set(_ALERT_INFO="RA")
> exten => s,n,SIPAddHeader(Call-Info: Answer-After=0)
> exten => s,n,NoOp()
> exten => s,n,Dial(${ARG1}||)
> exten => s,n(fail),Hangup
>
>  [page]
> exten => _X.,1,Macro(page,SIP/${EXTEN})
>
> [default]
> exten => 6050,1,VoiceMailMain
> exten = 7000,1,Goto(voicemenu-custom-1|s|1)
> exten => 6000,1,MeetMe(${EXTEN}|MI)
> exten = 3010,1,Goto(ringroups-custom-1|s|1)
> exten = 3020,1,Goto(ringroups-custom-2|s|1)
> exten = 6005,1,Queue(${EXTEN})
>
> [voicemenu-custom-1]
> include = default
> comment = Welcome
> alias_exten = 7000
> exten = s,1,Answer
> exten = s,2,Wai

[asterisk-users] ATA recommendation (wih FTP provisioning)

[Mike]

Hi,

 

I am looking for a good ATA recommendation, ideally something:

 

1) with one FXS and one LAN port (so it's as inexpensive as possible)

 

2) That can be provisioning using FTP (configuration and firmware upon
reboot, ideally remote reboot from a sip notify)

 

3) Supports T.38

 

Nice to have would be:

a) PoE powered and AC powered (my choice)

b) Small size-wise

 

I have been recommended the PAP2T in the past, and although I have used it
and sort of liked it, it wasn't possible to provision it using FTP (at least
as far as I could tell)

 

Any tip is welcomed.  I`m looking at the Patton ATA which is small, but it
doesn't support FTP provisioning either as far as I can tell.

 

 

 

Mike

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Congestion Tone

[Gustavo A Gonzalez]

Hello! I?ve connected an avaya PABX with an asterisk box through h323, all

calls from Avaya are sended to the asterisk. What I need is send to the

AVAYA PABX a congestion tone when Zap channels are full. How I do it?Thanks

for any idea!

 

 

Gustavo A. González
Dto. de Infraestructura
Despegar.com, Inc.
ggonza...@despegar.com 

 

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Getting SIP field P-Asserted-Identity from EAGI

[equis software]

Hi, using EAGI variables like

agi_request
agi_channel
agi_language
agi_type
agi_uniqueid
agi_callerid
agi_dnid
agi_rdnis
agi_context
agi_extension
agi_priority
agi_enhanced
agi_accountcode

  I get a lot of data about a call, but I need to obtain P-Asserted-Identity
value from a SIP call. Are tehe any eagi variable to get that? Or have you
any solution??


Thanks!!!
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] HP DL380 G5 with TE420

[Miguel Molina]

Hans Konings escribió:
> Hi
>
> I'm having problems getting the TE420 working in HP DL380G5 servers.
>
> The cards don't seem to be detected 100% by the BIOS. With two cards 
> in the server they are never detected.
>
Did you test the TE420 card on another server? It may be a defective card...

-- 
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
PBX: (+57 1)6500800 ext. 1201
Fax: (+57 1)6500816
Móvil: (+57)3138873587 


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Patton 5.3. How to get incoming calls ? [SOLVED]

[Olivier]

Hi,

Changing the line bellow helped to get incoming calls but I add to remove
secret= option in sip.conf (otherwise, Patton wouldn't respond to 407 Auth
required challenges).
If someone could enable secret and still get incoming calls (in any
SmartWare 5.X), please, do not hesitate to share here ...

  interface sip IF-ASTERISK
bind context sip-gateway ASTERISK
route call dest-table calls_from_SIP
remote 192.168.100.254 5060


Cheers

2009/2/25 Olivier 

> Hi,
>
> I'm trying to configure a 4638 to pass inbound and outbound to and from
> ISDN and SIP interfaces.
> I'm using web interface at the moment.
>
> Setup is:
>
> ISDN --  -- Patton 4638 --  Asterisk --  -- 
>
> I can call from IP phone but can't receive any incoming call : I can't see
> any SIP message coming in when a call comes in.
>
> Previously, with 4.2 firmware, you just have to edit routing table binding
> ISDN ports to SIP interface to get calls coming in but now with 5.3,
> configuration process changed.
> Here is an extract from my running config.
> Any idea ?
>
> Regards
>
> context cs switch
>
>   routing-table called-e164 appels_provenance_ISDN
> route [0-9]+ dest-service ASTERISK_SRV
> route default dest-service ASTERISK_SRV
>
>   routing-table called-uri appels_vers_ISDN
> route default dest-service isdnports
>
>   mapping-table called-e164 to called-ip transfo
> map [0-9]+ to 192.168.100.254
>
>   mapping-table called-e164 to called-uri transfo2
>
>   interface isdn IF-PBX
> route call dest-table appels_provenance_ISDN
>
>   interface isdn IF-PBX2
> route call dest-table appels_provenance_ISDN
>
>   interface isdn IF-PBX3
> route call dest-table appels_provenance_ISDN
>
>   interface isdn IF-PBX4
> route call dest-table appels_provenance_ISDN
>
>   interface sip IF-ASTERISK
> bind context sip-gateway ASTERISK
> route call dest-table appels_vers_ISDN
>
>   service sip-location-service ASTERISK_SRV
>
> bind location-service ASTERISK_SRV
> mode hunt
> hunt-timeout 20
>
>   service hunt-group isdnports
> drop-cause normal-unspecified
> drop-cause no-circuit-channel-available
> drop-cause network-out-of-order
>
> drop-cause temporary-failure
> drop-cause switching-equipment-congestion
> drop-cause access-info-discarded
> drop-cause circuit-channel-not-available
> drop-cause resources-unavailable
> route call 1 dest-interface IF-PBX
>
> route call 2 dest-interface IF-PBX2
> route call 3 dest-interface IF-PBX3
>
> context cs switch
>   no shutdown
>
> authentication-service patton
>   realm 1 asterisk
>   username patton password Otx2vJCEWP+8Bb6tqoGkwA== encrypted
>
> location-service ASTERISK_SRV
>   domain 1 192.168.100.254 5060
>   domain 2 asterisk 5060
>
>   identity-group default
>   identity patton
> alias name patton
>
> authentication outbound
>   authenticate 1 authentication-service patton username patton
>
> registration outbound
>   registrar 192.168.100.254 5060
>   proxy none
>   lifetime 3600
>   register auto
>   retry-timeout on-system-error 10
>   retry-timeout on-client-error 10
>
>   retry-timeout on-server-error 10
>
> call outbound
>   use profile tone-set default
>   use profile voip default
>   use profile sip default
>   preferred-transport-protocol udp
>   invite-transaction-timeout 32
>
>   non-invite-transaction-timeout 32
>
> call inbound
>   use profile tone-set default
>   use profile voip default
>   use profile sip default
>
>
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] DID's in a specific rate center

[M Hulber]

According to Wikipedia no law was actually ever passed.  I don't know if 
there is anything that falls under the FCC rules.  In any event it would 
be unethical and evidence of fraudulent intent if one was trying to 
defraud someone in the process of doing so.

Jason Aarons (US) wrote:
> Any idea what legal statues setting caller-id fraudulently falls under?
> Is there a federal law you can reference?
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of M Hulber
> Sent: Wednesday, February 25, 2009 4:13 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] DID's in a specific rate center
>
> Since it's not clear from this thread of conversation, do you need 100 
> unique DIDs?  If you do:
>
> That NPA is owned by Pacbell with the central office:  SCRMCA12
>
> I don't know if anyone but Pacbell will have numbers in that NPA.
>
> Since I use them and am happy with the service, you can try
> contacting http://www.jnctn.com and ask if they can get numbers
> there.  I do see they have others in the Sacramento area, in fact I
> have a Sacramento number with them already.
>
>
> If you don't and you just need outbound channels you can buy one (or 
> more) DIDs and then use that as the caller-id setting for all the 
> outbound calls.  This is perfectly legal since you own the DID that you 
> are using as the caller-id.  The channels you are using for outbound 
> calling don't have a DID associated with them so you need to associate 
> it with one by setting the caller-id to an owned/valid DID.  They don't 
> have to be unique.
>
> What is illegal is to set caller-id to a fraudulent value such that the 
> person on the other end will not be able to correctly identify the 
> originator of the call.
>
>
> Vikas wrote:
>   
>> I need 100 DID's in a specific rate center (916-854-). How do I go
>> about finding who owns the rate center ? If the DID's are available in
>> this rate center ?
>>
>> Thanks
>>
>> Vikas
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>   
>> 
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> -
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
>   

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] DID's in a specific rate center

[M Hulber]

Ok, then you can contact one of the myriad of providers to see if you 
can get a block of numbers.  Keep in mind, there will be differences in 
quality and reliability depending on which provider you use and on 
whether or not you use PRI or pure VoIP.  Since you already have the 
outbound capacity you probably have experience with providers in your 
area and how you want to implement it.  You might be able to ask for a 
fairly contiguous block if you desire it.

Vikas wrote:
>> Since it's not clear from this thread of conversation, do you need 100
>> unique DIDs?
>> 
>
> I apologize for not being more clear. I need 100 DID's. I already have
> channels which allow me to set the outgoing caller id. Depending on
> which extension is making the call I will be sending out the unique
> caller id. So that the person receiving the call can call back
> directly to the caller id that they received on their phone instead of
> going through the IVR hell.
>
> Vikas
>
> On Wed, Feb 25, 2009 at 3:13 PM, M Hulber  wrote:
>   
>> Since it's not clear from this thread of conversation, do you need 100
>> unique DIDs?  If you do:
>>
>>That NPA is owned by Pacbell with the central office:  SCRMCA12
>>
>>I don't know if anyone but Pacbell will have numbers in that NPA.
>>
>>Since I use them and am happy with the service, you can try
>>contacting http://www.jnctn.com and ask if they can get numbers
>>there.  I do see they have others in the Sacramento area, in fact I
>>have a Sacramento number with them already.
>>
>>
>> If you don't and you just need outbound channels you can buy one (or
>> more) DIDs and then use that as the caller-id setting for all the
>> outbound calls.  This is perfectly legal since you own the DID that you
>> are using as the caller-id.  The channels you are using for outbound
>> calling don't have a DID associated with them so you need to associate
>> it with one by setting the caller-id to an owned/valid DID.  They don't
>> have to be unique.
>>
>> What is illegal is to set caller-id to a fraudulent value such that the
>> person on the other end will not be able to correctly identify the
>> originator of the call.
>>
>>
>> Vikas wrote:
>> 
>>> I need 100 DID's in a specific rate center (916-854-). How do I go
>>> about finding who owns the rate center ? If the DID's are available in
>>> this rate center ?
>>>
>>> Thanks
>>>
>>> Vikas
>>>
>>> ___
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>   
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> 

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Multiple SIPGate accounts.

[Gordon Henderson]

On Wed, 25 Feb 2009, Klaus Darilion wrote:

> I supsect that the incoming request is tried to match against a peer -
> based on IP:port. Thus, it will always match the same peer, regardsless
> if the call is incoming from account1 or account2.
>
> Try using the same context in both peer definitions and put the
> 1212121 and 1313131 extensions in this context.

Is this considered a bug in asterisk, or "just the way it is?"

I've just had exactly the same issue with another ITSP who want to create 
a separate SIP Trunk for each incoming DDI to the same PBX and had exactly 
the same issue.

(And ironically this ITSP is also using asterisk, even if they won't admit 
it - those voicemail messages sound surprisingly familiar!)

Gordon

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] HP DL380 G5 with TE420

[Hans Konings]

On Thu, Feb 26, 2009 at 11:49 AM, stoffell  wrote:

> On Thu, Feb 26, 2009 at 11:25 AM, Hans Konings  wrote:
>
>>
>>
> We have it working on a dl320, several ML350's, ml310, but never tried on a
> dl380 yet.
>
> We had serious issues in the past when iLO was enabled on a 350. disabling
> iLO on that machine helped us. (we had irq errors)
>
> You could try disabling iLO, just to make sure.
>

Disabling the ILO didn't work.

Thanks anyway
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] DID's in a specific rate center

[Alex Balashov]

As always, heed should be taken that number pooling and LNP have made  
the idea that someone "owns" a rate center increasingly meaningless.

See the first or second question on our VoIP FAQ for more information  
@ www.evaristesys.com. It is related. I would paste direct URL but am  
not at computer right now.

Thanks,

-- Alex

--
Sent from mobile device

On Feb 26, 2009, at 6:57 AM, "Frank Bulk"  wrote:

> Look it up at www.localcallingguide.com.  The name on record there  
> may be
> the LECs name -- then you need to find a SIP provider or DID handler  
> that
> has a business relationship with that LEC.  For example, in the  
> state of
> Iowa Vonage obtained (at least some of) it's numbering resources from
> McCleod.
>
> Frank
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vikas
> Sent: Wednesday, February 25, 2009 12:46 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] DID's in a specific rate center
>
> I need 100 DID's in a specific rate center (916-854-). How do I go
> about finding who owns the rate center ? If the DID's are available in
> this rate center ?
>
> Thanks
>
> Vikas
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] DID's in a specific rate center

[Frank Bulk]

Look it up at www.localcallingguide.com.  The name on record there may be
the LECs name -- then you need to find a SIP provider or DID handler that
has a business relationship with that LEC.  For example, in the state of
Iowa Vonage obtained (at least some of) it's numbering resources from
McCleod.

Frank

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vikas
Sent: Wednesday, February 25, 2009 12:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] DID's in a specific rate center

I need 100 DID's in a specific rate center (916-854-). How do I go
about finding who owns the rate center ? If the DID's are available in
this rate center ?

Thanks

Vikas

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Swichted digits on received number from fax on an fxs port

[Loic Didelot]

Yes,
its the dialed number.

I have asterisk 1.4.21.1 and zaptel-1.4.11. Reading your question I am
pretty sure you advise me to update.

Loic

On Thu, 2009-02-26 at 12:44 +0200, Tzafrir Cohen wrote:
> On Tue, Feb 24, 2009 at 10:15:27AM +0100, Loic Didelot wrote:
> > Hello,
> > I have connected a fax machine to a xorcom fxs port (signalling=fxo_ks).
> > But when I try to dial (send a fax) the number that I receive in
> > asterisk is wrong. Quite often a few digits are wrong but sometime is
> > correct. It looks like it works 2 times out of 10.
> > 
> > Examples: 
> > 2090 becomes 2999 or 2000
> > 1234567890 becomes 1234566790
> 
> Is this the number Asterisk detects as dialed?
> 
> What version of Zaptel (DAHDI?) do you have there exactly?
> 
-- 
Loïc DIDELOT
MIXvoip S.a.
Tel: +352 20  20
Fax: +352 20  90
ldide...@mixvoip.com
http://www.mixvoip.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] call-limit on a per destination basis

[Klaus Darilion]

I have no clue about IAX, but if IAX does not support it you can program 
it yourself using the GROUP and GROUPCOUNT functions.

regards
klaus

Jean-Michel Hiver wrote:
> Hello,
> 
> I use asterisk to to IAX2 trunking between London POP & Reunion Island 
> pop. I would like to know if it's possible to do a kind of call-limit 
> (i.e. restrict to XX) channels but on a per dialcode and / or 
> destination basis.
> 
> 
> For example:
> 
> [trunk]
> ; reunion proper, i want to send no more than 24 channels
> exten => _0262XX,1,Dial(IAX2/mytrunk/${EXTEN})
> 
> ; reunion mobile, i want to send no more than 12 channels
> exten => _0692XX,1,Dial(IAX2/mytrunk/${EXTEN})
> exten => _0693XX,1,Dial(IAX2/mytrunk/${EXTEN})
> 
> 
> How would you go about it? Currently my IAX2 peer definition looks like 
> this:
> 
> # machine in london
> [mytrunk]
> type=friend
> host=$reunion_ip
> trunk=yes
> qualify=yes
> context=route
> 
> # machine in reunion island
> [mytrunk]
> type=friend
> host=$london_ip
> trunk=yes
> qualify=yes
> context=route
> 
> I use version Asterisk 1.4.11, production environment currently doing 
> 25,000 minutes / day (that means if i want to upgrade i need to do it on 
> separate servers just in case something goes wrong).
> 
> 
> Cheers,
> Jean-Michel.
> 
> 
> 
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] HP DL380 G5 with TE420

[Michel Verbraak]

Hans Konings schreef:

Hi

I'm having problems getting the TE420 working in HP DL380G5 servers.

The cards don't seem to be detected 100% by the BIOS. With two cards 
in the server they are never detected.


things I've tried:

1 Update firmware to latest (P56) for the server
2 change irq settings
3 disable all onboard devices on server and remove raid controller
4 different cards in different slots

What I mean by not detected is that in the HP utility SMBIOS you can 
read out the status of the pci-express slot and it says slot available 
also lspci does not list the card.

Every so often the card is detected and works properly.


I've tried with a different server (HPdl320g5p) and the card is 
detected in this but the cards generate NMI errors on many bootups.



Does anybody have this combination of hardware working? Or can anybody 
think of something I've missed?



Rgds
Hans



I have a HP DL380G5 (dual quadcore) with a TE121 (PCI-E) card which 
works like a charm.


Regards,

Michel.






___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] call-limit on a per destination basis

[Jean-Michel Hiver]

Hello,

I use asterisk to to IAX2 trunking between London POP & Reunion Island pop.
I would like to know if it's possible to do a kind of call-limit (i.e.
restrict to XX) channels but on a per dialcode and / or destination basis.


For example:

[trunk]
; reunion proper, i want to send no more than 24 channels
exten => _0262XX,1,Dial(IAX2/mytrunk/${EXTEN})

; reunion mobile, i want to send no more than 12 channels
exten => _0692XX,1,Dial(IAX2/mytrunk/${EXTEN})
exten => _0693XX,1,Dial(IAX2/mytrunk/${EXTEN})


How would you go about it? Currently my IAX2 peer definition looks like
this:

# machine in london
[mytrunk]
type=friend
host=$reunion_ip
trunk=yes
qualify=yes
context=route

# machine in reunion island
[mytrunk]
type=friend
host=$london_ip
trunk=yes
qualify=yes
context=route

I use version Asterisk 1.4.11, production environment currently doing 25,000
minutes / day (that means if i want to upgrade i need to do it on separate
servers just in case something goes wrong).


Cheers,
Jean-Michel.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] HP DL380 G5 with TE420

[stoffell]

On Thu, Feb 26, 2009 at 11:25 AM, Hans Konings  wrote:

> I've tried with a different server (HPdl320g5p) and the card is detected in
> this but the cards generate NMI errors on many bootups.
> Does anybody have this combination of hardware working? Or can anybody
> think of something I've missed?
>

We have it working on a dl320, several ML350's, ml310, but never tried on a
dl380 yet.

We had serious issues in the past when iLO was enabled on a 350. disabling
iLO on that machine helped us. (we had irq errors)

You could try disabling iLO, just to make sure.

regards,
stoffell
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Swichted digits on received number from fax on an fxs port

[Tzafrir Cohen]

On Tue, Feb 24, 2009 at 10:15:27AM +0100, Loic Didelot wrote:
> Hello,
> I have connected a fax machine to a xorcom fxs port (signalling=fxo_ks).
> But when I try to dial (send a fax) the number that I receive in
> asterisk is wrong. Quite often a few digits are wrong but sometime is
> correct. It looks like it works 2 times out of 10.
> 
> Examples: 
> 2090 becomes 2999 or 2000
> 1234567890 becomes 1234566790

Is this the number Asterisk detects as dialed?

What version of Zaptel (DAHDI?) do you have there exactly?

-- 
   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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Swichted digits on received number from fax on an fxs port

[Loic Didelot]

Does anyone have ideas on how to debug this?

I connected a different and old fax machine and everything works fine.
But for sure the other fax machine works fine too if I connect it to an
analog line from the telco.

Si it some incompatibility between my setup (asterisk, xorcom) and
the fax machine from Ricoh.

Best regards,
Loic. 


On Tue, 2009-02-24 at 20:12 +0100, Loic Didelot wrote:
> Hello,
> I do not mean the caller ID, I am talking about the dialed extension.
> 
> Best regards,
> Loïc Didelot.
> 
> On Tue, 2009-02-24 at 13:34 +0200, Tzafrir Cohen wrote:
> > On Tue, Feb 24, 2009 at 10:15:27AM +0100, Loic Didelot wrote:
> > > Hello,
> > > I have connected a fax machine to a xorcom fxs port (signalling=fxo_ks).
> > > But when I try to dial (send a fax) the number that I receive in
> > > asterisk is wrong. Quite often a few digits are wrong but sometime is
> > > correct. It looks like it works 2 times out of 10.
> > > 
> > > Examples: 
> > > 2090 becomes 2999 or 2000
> > > 1234567890 becomes 1234566790
> > 
> > On fxo_ks the phone does not get to set the caller ID. Where do you set
> > the caller ID? Where do you read it?
> > 
> -- 
> Loïc DIDELOT
> MIXvoip S.a.
> Tel: +352 20  20
> Fax: +352 20  90
> ldide...@mixvoip.com
> http://www.mixvoip.com
> 
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Loïc DIDELOT
MIXvoip S.a.
Tel: +352 20  20
Fax: +352 20  90
ldide...@mixvoip.com
http://www.mixvoip.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] HP DL380 G5 with TE420

[Hans Konings]

Hi

I'm having problems getting the TE420 working in HP DL380G5 servers.

The cards don't seem to be detected 100% by the BIOS. With two cards in the
server they are never detected.

things I've tried:

1 Update firmware to latest (P56) for the server
2 change irq settings
3 disable all onboard devices on server and remove raid controller
4 different cards in different slots

What I mean by not detected is that in the HP utility SMBIOS you can read
out the status of the pci-express slot and it says slot available also lspci
does not list the card.
Every so often the card is detected and works properly.


I've tried with a different server (HPdl320g5p) and the card is detected in
this but the cards generate NMI errors on many bootups.


Does anybody have this combination of hardware working? Or can anybody think
of something I've missed?


Rgds
Hans
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Dictate

[amit mehta]

Hello Members,
Sorry for hijacking the earlier thread and asking the question last time.

Is anyone aware about a solution to call incoming number and dictate
the  files by using Dictate feature of Asterisk used for Medical
Transcription industry.

Thanks & Regards,
Amit Mehta
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6

[Klaus Darilion]

amit mehta wrote:
> Hello Users,
> 
> Is anyone aware about a solution to call incoming number and dictate the 
> files by using Dictate feature of Asterisk used for Medical 
> Transcription industry.

I guess nobody will read your email as you:
1. hijacked a thread 
(http://www.internet-description.com/t/thread-hijacking.html)
2. did not even changed the subject

So, most user do not even read your email as they think the subject is 
about Gosub, not dictate.

So I would suggest to try again, writing a "new email" instead of 
clicking on "reply".

regards
klaus



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] CDR - Asterisk-Stat and PHP5

[Paulo Santos]

Tiago Durante wrote:
> Hi all,
> 
> I don't know if its the right place to ask, but... Does any one have
> the asterisk-stat-v2 running with PHP5?
> 
> 
> Tks!
> 
> 

# php --version

PHP 5.2.0-8+etch13 (cli) (built: Oct  2 2008 08:26:18)
Copyright (c) 1997-2006 The PHP Group
Zend Engine v2.2.0, Copyright (c) 1998-2006 Zend Technologies

Working for me. Don't forget you need php5-gd for the graphics to show.

-- 
HTML e-mail is evil. Go plain text.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] TE121 on Asterisk

[Oguzhan Kayhan]

> On Thu, Feb 26, 2009 at 08:41:25AM +0200, Oguzhan Kayhan wrote:
>> >> Oguzhan Kayhan wrote:
>> >>> I want to change it to E1 instead of T1.
>> >>> here comes the problem.
>> >>>
>> >>
>> >> If it's anything like the older cards, there is a jumper on the card
>> >> that sets it to T1/E1
>> >>
>> >> Doug
>> >>
>> > Yes,
>> > I just noticed the jumper on the card.
>> > Thanks a lot.
>> >
>> >
>> Yes i changed the jumper to enable  E1.
>> dahdi_scan shows the following
>> [1]
>> active=yes
>> alarms=UNCONFIGURED
>> description=Wildcard TE121 Card 0
>> name=WCT1/0
>> manufacturer=Digium
>> devicetype=Wildcard TE121 with VPMADT032
>> location=PCI Bus 04 Slot 09
>> basechan=1
>> totchans=31
>> irq=17
>> type=digital-E1
>> syncsrc=0
>> lbo=0 db (CSU)/0-133 feet (DSX-1)
>> coding_opts=HDB3
>> framing_opts=CCS,CRC4
>> coding=
>> framing=
>>
>> But when i try to run dahdi_genconf i got the following error.
>>
>> 31 channels in a T1 span at /usr/local/share/perl/5.10.0/Dahdi/Span.pm
>> line 244.
>>
>> dahdi_hardware
>> 31 channels in a T1 span at /usr/local/share/perl/5.10.0/Dahdi/Span.pm
>> line 244.
>>
>>
>> What should i do about it?
>
> I believe it was resolved with:
>
>   http://svn.digium.com/view/dahdi?view=revision&revision=5626
>   
> http://svn.digium.com/view/dahdi/tools/trunk/xpp/perl_modules/Dahdi/Span.pm?r1=5452&r2=5626
>
> What version of the tools do you use?
>

Dahdi 2.1.0.4 not svn versions..




> --
>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
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Need US Dialing Account with Asterisk

[Kashif Naeem]

Hello,

We need a US Dialing Plan to use it with Asterisk. We need 3 - 4 Channels
with it. Please suggest some good voice quality service.

Regards,

Kashif Naeem
Business Development Manager
Hadi Telecom
www.haditelecom.com

Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766

Email: kas...@haditelecom.com
MSN: kashif__na...@hotmail.com
Gmail: meet.kas...@gmail.com
Skype: kashif.naeem

302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] TE121 on Asterisk

[Tzafrir Cohen]

On Thu, Feb 26, 2009 at 08:41:25AM +0200, Oguzhan Kayhan wrote:
> >> Oguzhan Kayhan wrote:
> >>> I want to change it to E1 instead of T1.
> >>> here comes the problem.
> >>>
> >>
> >> If it's anything like the older cards, there is a jumper on the card
> >> that sets it to T1/E1
> >>
> >> Doug
> >>
> > Yes,
> > I just noticed the jumper on the card.
> > Thanks a lot.
> >
> >
> Yes i changed the jumper to enable  E1.
> dahdi_scan shows the following
> [1]
> active=yes
> alarms=UNCONFIGURED
> description=Wildcard TE121 Card 0
> name=WCT1/0
> manufacturer=Digium
> devicetype=Wildcard TE121 with VPMADT032
> location=PCI Bus 04 Slot 09
> basechan=1
> totchans=31
> irq=17
> type=digital-E1
> syncsrc=0
> lbo=0 db (CSU)/0-133 feet (DSX-1)
> coding_opts=HDB3
> framing_opts=CCS,CRC4
> coding=
> framing=
> 
> But when i try to run dahdi_genconf i got the following error.
> 
> 31 channels in a T1 span at /usr/local/share/perl/5.10.0/Dahdi/Span.pm
> line 244.
> 
> dahdi_hardware
> 31 channels in a T1 span at /usr/local/share/perl/5.10.0/Dahdi/Span.pm
> line 244.
> 
> 
> What should i do about it?

I believe it was resolved with:

  http://svn.digium.com/view/dahdi?view=revision&revision=5626
  
http://svn.digium.com/view/dahdi/tools/trunk/xpp/perl_modules/Dahdi/Span.pm?r1=5452&r2=5626

What version of the tools do you use?

-- 
   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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] codec_dahdi and Asterisk 1.6.0.6

[Olivier]

2009/2/26 Brandon B. 

> I've got a question about codec_dahdi witrh a system running Asterisk
> 1.6.0.6 and DAHDI 2.1.0.4 with a TE410P card. The system is used primary to
> route calls between different PRI connections, so no transcoding between
> codecs is happening as far as I am aware.
>
> 1) How can I use codec_dahdi? Would it be useful when passing a call from
> one dahdi channel to another dahdi channel?
>
> 2) I'm getting the following error on the Asterisk CLI unless I disable the
> codec_dahdi module:
>
> ERROR[18854]: codec_dahdi.c:399 find_transcoders: Failed to open
> /dev/dahdi/transcode: No such file or directory
>
> Is this because I do not have a hardware trancoding device? Can I safely
> ignore this error or is it a bug?


I'm far from being certain (20% sure) but I think codec_dahdi relates to
Digium transcoding cards (G729 to A/U LAW).

My understanding is that current default and implicit installation policy is
to have those cards enabled and running without any user action which make
those error messages appear on systems not including those cards.

I would be very happy if someone could correct this explanation and shed
some light about what this is all about ...

>
>
> Brandon.
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] TE121 on Asterisk

[Oguzhan Kayhan]

Yeah thanks a lot,
It worked that way...


> Oguzhan Kayhan schreef:
 Oguzhan Kayhan wrote:

> I want to change it to E1 instead of T1.
> here comes the problem.
>
>
 If it's anything like the older cards, there is a jumper on the card
 that sets it to T1/E1

 Doug


>>> Yes,
>>> I just noticed the jumper on the card.
>>> Thanks a lot.
>>>
>>>
>>>
>> Yes i changed the jumper to enable  E1.
>> dahdi_scan shows the following
>> [1]
>> active=yes
>> alarms=UNCONFIGURED
>> description=Wildcard TE121 Card 0
>> name=WCT1/0
>> manufacturer=Digium
>> devicetype=Wildcard TE121 with VPMADT032
>> location=PCI Bus 04 Slot 09
>> basechan=1
>> totchans=31
>> irq=17
>> type=digital-E1
>> syncsrc=0
>> lbo=0 db (CSU)/0-133 feet (DSX-1)
>> coding_opts=HDB3
>> framing_opts=CCS,CRC4
>> coding=
>> framing=
>>
>> But when i try to run dahdi_genconf i got the following error.
>>
>> 31 channels in a T1 span at /usr/local/share/perl/5.10.0/Dahdi/Span.pm
>> line 244.
>>
>>
>
>> dahdi_hardware
>> 31 channels in a T1 span at /usr/local/share/perl/5.10.0/Dahdi/Span.pm
>> line 244.
>>
>>
> There is a bug in the perl module. I had the same problem. It trips over
> the following part "*WCT1/0*" in "*WCT1/0* "Wildcard TE121 Card 0"
> (MASTER) HDB3/CCS". It finds the text T1 in there and expects it to be a
> T1 jumpered card in stead of an E1 jumpered card and it tries to create
> a T1 system.conf file. I still need to make a bug report about this.
>
> Your card is probably working allright. Create the right system.conf
> file in /etc/dahdi/
> Mine has (E1 for Dutch KPN ISDN15/20/30) and the following lines:
> /# Span 1: WCT1/0 "Wildcard TE121 Card 0" (MASTER) HDB3/CCS/CRC4
> RECOVERING
> span=1,1,0,ccs,hdb3
> # termtype: te
> bchan=1-15,17-31
> dchan=16
> echocanceller=mg2,1-15,17-31
> /
> Probably you have to enter ",CRC4" at the end of the span line.
> (span=1,1,0,ccs,hdb3,crc4)
> When you edited the file do a:
> # dahdi_cfg -
> DAHDI Tools Version - 2.1.0.2
>
> DAHDI Version: 2.1.0.4
> Echo Canceller(s): MG2
> Configuration
> ==
>
> SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
>
> Channel map:
>
> Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01)
> Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02)
> 
> Channel 15: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 15)
> Channel 16: D-channel (Default) (Slaves: 16)
> Channel 17: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 17)
> 
> Channel 31: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 31)
>
> 31 channels to configure.
>
> Setting echocan for channel 1 to mg2
> 
>
> And followed by:
> #cat /proc/dahdi/1
> Span 1: WCT1/0 "Wildcard TE121 Card 0" (MASTER) HDB3/CCS
> IRQ misses: 1
>
>1 WCT1/0/1 Clear (In use)  (EC: MG2)
>2 WCT1/0/2 Clear (In use)  (EC: MG2)
>3 WCT1/0/3 Clear (In use)  (EC: MG2)
>4 WCT1/0/4 Clear (In use)  (EC: MG2)
>5 WCT1/0/5 Clear (In use)  (EC: MG2)
>6 WCT1/0/6 Clear (In use)  (EC: MG2)
>7 WCT1/0/7 Clear (In use)  (EC: MG2)
>8 WCT1/0/8 Clear (In use)  (EC: MG2)
>9 WCT1/0/9 Clear (In use)  (EC: MG2)
>   10 WCT1/0/10 Clear (In use)  (EC: MG2)
>   11 WCT1/0/11 Clear (In use)  (EC: MG2)
>   12 WCT1/0/12 Clear (In use)  (EC: MG2)
>   13 WCT1/0/13 Clear (In use)  (EC: MG2)
>   14 WCT1/0/14 Clear (In use)  (EC: MG2)
>   15 WCT1/0/15 Clear (In use)  (EC: MG2)
>   16 WCT1/0/16 HDLCFCS (In use)
>   17 WCT1/0/17 Clear  (EC: MG2)
>   18 WCT1/0/18 Clear  (EC: MG2)
>   19 WCT1/0/19 Clear  (EC: MG2)
>   20 WCT1/0/20 Clear  (EC: MG2)
>   21 WCT1/0/21 Clear  (EC: MG2)
>   22 WCT1/0/22 Clear  (EC: MG2)
>   23 WCT1/0/23 Clear  (EC: MG2)
>   24 WCT1/0/24 Clear  (EC: MG2)
>   25 WT1/0/25 Clear  (EC: MG2)
>   26 WCT1/0/26 Clear  (EC: MG2)
>   27 WCT1/0/27 Clear  (EC: MG2)
>   28 WCT1/0/28 Clear  (EC: MG2)
>   29 WCT1/0/29 Clear  (EC: MG2)
>   30 WCT1/0/30 Clear  (EC: MG2)
>   31 WCT1/0/31 Clear  (EC: MG2)
>
> I have a ISDN15 connected to it so only 15 lines are in use. If you see
> something like YELLOW or RED or BLUE in the previous something is wrong
> with your line. I had this first but this was because I had the CRC4
> option added to my system.conf file. Your syslog log file
> /var/log/messages wil tell also if you have an alarm.
>
> Regards, Michel.
>>
>> What should i do about it?
>>
>>
>>
>>>
 --

 Ben Franklin quote:

 "Those who would give up Essential Liberty to purchase a little
 Temporary
 Safety, deserve neither Liberty nor Safety."


 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


>>>
>>> __