Re: [asterisk-users] [Asterisk-users] asterisk-1.8.0 compilation error

[RAJNIKANT VANZA]

Hi Paul,

Thanks for reply.

I have some mistake send compilation logs. i have written cdr_webservice.c
module and its work on asterisk-1.6.2.6 version on production server. but i
want to upgrade asterisk version.

# make
[CC] cdr_webservice.c -> cdr_webservice.o
In file included from
/usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/cdr.h:31,
from cdr_webservice.c:29:
/usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/data.h:233: error:
field ‘AST_DATA_IPADDR’ has incomplete type
/usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/data.h:794: error:
return type is an incomplete type
/usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/data.h: In function
‘ast_data_retrieve_ipaddr’:
/usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/data.h:799: warning:
‘return’ with a value, in function returning void
make[1]: *** [cdr_webservice.o] Error 1
make: *** [cdr] Error 2

Help me for resolve this compilation errors.

-- 
Best Regards,
Rajnikant Vanza

On Wed, Nov 24, 2010 at 8:01 PM, Paul Belanger wrote:

> On 10-11-24 06:09 AM, RAJNIKANT VANZA wrote:
> > make[1]: *** [cdr_webservice.o] Error 1
> > make: *** [cdr] Error 2
> >
> What is cdr_webservice.o ?
>
> --
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org
>
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Re: [asterisk-users] [asterisk-ss7] Incoming calls through SS7 for data modem transmissions - possible??

[José Pablo Méndez Soto]

Thank you Horacio and Cary.

We will try receiving SS7, routing via SIP, answering on the AS5300, then
looping back to itself (out PRI, in PRI ports) in order to invoke the modem
termination. This way we may be able to spare the TDM cards in Asterisk and
reuse the E1 ports installed in the gateway.

Best regards,

*José Pablo Méndez
   *


2010/11/24 Horacio J. Peña 

> Hola!
>
> ZapRAS seems to work only with ISDN calls. "This command is not for use
> with
> analog lines; it does not provide a modem emulator."
> (http://www.voip-info.org/wiki/view/Asterisk+cmd+ZapRAS)
>
> You need something doing the modulation. It seems that iaxmodem is your
> best
> bet, and you'll have to make a good bunch of work on it to be able to use
> as you
> want to.
>
> If your client has the cisco gateways, I'd suggest you to keep them. They
> are
> very reliable and tested, and with MICA cards they have not a high resale
> value,
> so you'll probably end with them as paperweights unless you happen to have
> some
> stack of C549 cards to repurpose them.
>
> Saludos,
> H
>
> On Wed, Nov 24, 2010 at 07:58:37PM -0600, José Pablo Méndez Soto wrote:
> >Hello,
> >We are working on implementing a solution for a medium service
> >provider. They were previously using a Cisco AS5300 gateway with some
> >PRI trunks to receive modem calls, then route them out the Internet.
> >The Telco they were buying the trunks from, discovered this
> >configuration and restricted them due to legal conventions, and stated
> >that in order to continue doing this, they would have to talk SS7
> >directly.
> >We are planning on solving this by placing an Asterisk server with
> some
> >TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to
> >the AS5300 for the dial-up to complete after authenticating against a
> >RADIUS server.
> >My questions is: can we use only Asterisk to complete/terminate the
> >dial-up connection, removing the AS5300 out of the picture? We would
> >probably need a PPP channel configuration to link the modem connection
> >with the Internet.
> >Current topology to be set-up:
> >Telco --> SS7 --> TE410P-AsteriskServer --> ISDN --> AS5300 -->
> >Internet
> >Ideal topology:
> >Telco --> SS7 --> TE410P-AsteriskServer --> Internet
> >Some posts talk about zapRAS being able to accomplish this, not quite
> >sure though
> >Sounds like possible:
> >[1]
> http://lists.digium.com/pipermail/asterisk-users/2004-January/026956
> >.html
> >[2]
> http://lists.digium.com/pipermail/asterisk-users/2009-November/24021
> >8.html
> >Sounds like not possible:
> >[3]
> http://lists.digium.com/pipermail/asterisk-users/2009-November/24020
> >2.html
> >Thanks in advance,
> >José Pablo Méndez
> >
> > References
> >
> >1. mailto:asterisk-users@lists.digium.com
> >2.
> http://lists.digium.com/pipermail/asterisk-users/2009-November/240218.html
> >3.
> http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html
>
> > --
> > _
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> >
> > asterisk-ss7 mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-ss7
>
>
> --
> Horacio J. Peña
> hor...@compendium.com.ar
> hor...@uninet.edu
>
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Re: [asterisk-users] Spam

[Steven Stromer]

Same here. But, can the genie ever be put back in the bottle?

> Cary Fitch wrote:
>> Has anyone else noticed "new spam" in the last 2-3 weeks?
>>
>
> No,
>
> But I run ASSP in front of my MTA.
>
> Doug

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Re: [asterisk-users] Spam

[Doug Lytle]

Cary Fitch wrote:
> Has anyone else noticed "new spam" in the last 2-3 weeks?
>

No,

But I run ASSP in front of my MTA.

Doug

-- 
Ben Franklin quote:

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


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Re: [asterisk-users] Incoming calls through SS7 for datamodemtransmissions - possible??

[José Pablo Méndez Soto]

Thanks Cary,

The first topology we are working on should be the best way then.

Asterisk will answer SS7 calls, route them to the ISDN channels to be
terminated by the AS5300 as they were doing before. I think TDM-2-TDM
shouldn't be that much of a problem eh? No further equipment needed?


*José Pablo Méndez
   *


2010/11/24 Cary Fitch 

>  I understand the problem.  You can’t resell PRI connections.
>
>
>
> I don’t think Asterisk can convert TDM to IP.  It does convert TDM to SIP
> which is then sent out over IP.What you want to do is have it do the
> TDM/Modem conversion without the PRIs and Cisco Gear.
>
>
>
> There used to be a way to do this, and maybe still is but not just with
> Asterisk perhaps.
>
>
>
> I know that Ascend/Lucent TNTs (and I am sure some other equipment)  could
> take TDM trunks, which could be SS7 trunks, and convert them to IP.
>
>
>
> The point in this is that they are SS7 based.  You can take SS7 trunks from
> either the Asterisk box or direct from the Telco and convert them to IP.
>
>
>
> NO PRIs involved.  Yes, more “telco grade carrier equipment” but no PRIs.
>
>
>
> A lot of this equipment was available by the pound a few years back.
>
>
>
> Cary
>
>
>  --
>
> *From:* José Pablo Méndez Soto [mailto:aux...@gmail.com]
> *Sent:* Wednesday, November 24, 2010 8:34 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Cc:* ca...@usawide.net
> *Subject:* Re: [asterisk-users] Incoming calls through SS7 for
> datamodemtransmissions - possible??
>
>
>
> Thanks Cary,
>
> What happens is, the Telco won't allow the small company to resell the ISDN
> connections, meaning, they bought the trunks and DIDs, then sold dialing
> plans to route incoming calls through the PRIs out the Internet. This is not
> the issue though. We definitely have to migrate to an SS7 capable platform,
> because that is the only way the Telco allows them to resell the dial-up
> connections (not ISDN), and Asterisk is the current bet.
>
> If we can get Asterisk to pick up those calls via SS7, then authenticate
> them, send them out to the Internet, we would be achieving a %100 usage on
> the Digium cards, because one of them wouldn't be used to talk to the AS.
>
> Can Asterisk do this?
>
>
> Thanks again,
>
> *José Pablo Méndez**
>*
>
>  On Wed, Nov 24, 2010 at 7:59 PM, Cary Fitch  wrote:
>
> I am not sure where you are and what legal conventions are involved.
>
>
>
> Are you saying the Telco (and legal restrictions) say you can’t send calls
> to the internet via the AS5300 but you can if Asterisk does it directly?
> What is the “logic” in that?
>
>
>
> Or are they saying your Telco to Asterisk trunks have to be SS7 controlled?
>
>
>
>
> Or are you concerned about Asterisk handling the TDM to IP conversion in an
> adequate manner?
>
>
>
> I am not sure/aware myself that Asterisk will do a modem to IP conversion.
> I think in your example the AS5300 is doing that.
>
>
>
> What is the Telco’s problem in doing what the customer was doing before?
>
>
>
> Feel free to correspond directly if you want to.
>
>
>
> Cary Fitch
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *José Pablo Méndez
> Soto
> *Sent:* Wednesday, November 24, 2010 7:31 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Incoming calls through SS7 for data
> modemtransmissions - possible??
>
>
>
> Hello,
>
> We are working on implementing a solution for a medium service provider.
> They were previously using a Cisco AS5300 gateway with some PRI trunks to
> receive modem calls, then route them out the Internet.
>
> The Telco they were buying the trunks to discovered this configuration and
> restricted them due to legal conventions, and stated that in order to
> continue doing this, they would have to talk SS7 directly.
>
> We are planning on solving this by placing an Asterisk server with some
> TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the
> AS5300 for the dial-up to complete after authenticating against a RADIUS
> server.
>
> My questions is: can we use only Asterisk to complete/terminate the dial-up
> connection, removing the AS5300 out of the picture?
>
> Current topology to be set-up:
> Telco --> SS7 --> TE410P-AsteriskServer --> ISDN --> AS5300 --> Internet
>
> Ideal topology:
> Telco --> SS7 --> TE410P-AsteriskServer --> Internet
>
>
> Some posts talk about zapRAS being able to accomplish this, not quite sure
> though
>
> Sounds like possible:
> http://lists.digium.com/pipermail/asterisk-users/2004-January/026956.html
>
> Sounds like not possible:
> http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html
>
>
> Thanks in advance,
>
>
> *José Pablo Méndez**
>   *
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://w

[asterisk-users] Spam

[Cary Fitch]

I have been pounded with new, mostly text spam in the last few weeks.
Tonight I realized that the address that is being spammed is a personal one
I use for this list.

Has anyone else noticed "new spam" in the last 2-3 weeks?

Cary Fitch


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Re: [asterisk-users] DAHDI phantom pickup when ringing

[Lyle Giese]

Jonathan Hunter wrote:
> On 24 November 2010 01:20, Lyle Giese  > wrote:
>
> Post the revelent portions of your extension.conf.  Maybe you have
> a logic error somewhere.
>
> Thanks Lyle.
>
> My extensions.conf is fairly simple in this regard; I use macro-stdexten:
>
>  [macro-stdexten];
> exten => s,1,NoOp('${CALLERID(NAME)}' [${CALLERID(NUM)}] calling
> [${ARG1}])
> exten => s,n,Set(MBOXCONTEXT=)
> exten => s,n,Dial(${ARG1},30)   ; Ring the interface,
> 30 seconds maximum
> exten => s,n,MailboxExists(${macro_ext...@${mboxcontext})
> exten => s,n,NoOp(Got mailbox status of '${VMBOXEXISTSSTATUS}')
> exten =>
> s,n,GotoIf($["${VMBOXEXISTSSTATUS}"="SUCCESS"]?s-Voicemail,1:s-NOANSWER,1)
>
> and it is called with
> "SIP/&DAHDI/1r1&DAHDI/3r1&DAHDI/5r1&DAHDI/7r3&SIP/&SIP/&SIP/&SIP/&SIP/&DAHDI/2&DAHDI/4&DAHDI/6"
>
> Have you tried to move the set from channel 5 to 8 and 7 to 9? (to
> see if one or two of the fxs channels have gone bad in the chan bank?)
>
> Good idea, thank you - I will try this tonight.
>  
>
> It could also be a power supply issue inside the Zhone that tries
> to 'trip' the ringing.
>
>
> Hmm - not sure how I might determine whether this is the case or not..
> It only seems to occur on some channels, at the moment.
>
Thinking on this, if the power supply is going bad, reducing the number
of DAHDI channels in ringing state may help.  I am an old telco guy
having spent 23 years working for the biggest telco in the US in their
CO's.  I tend to think something funny with the channel bank or the
channel units.  Seen that happen many times working for them. 

I assume the wiring is good and not 'wet'.  If it was underground or in
a damp environment...

I would go back to thinking chan bank with FXS channel units, not DAHDI
channels.  It will focus the attention where you have power supplies(-24
or -48 volt talk battery and ringing current with trip battery
super-imposed) and all the electronic things that can go wrong with that
and the detecting of offhook state.  It would be easy for the
electronics to think the phone was offhook when ringing, but not when
supplying only talk battery when the channel units or power supply goes
flakey.

Lyle Giese
> Thanks,
>
> Jonathan

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Re: [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible??

[José Pablo Méndez Soto]

Thanks Cary,

What happens is, the Telco won't allow the small company to resell the ISDN
connections, meaning, they bought the trunks and DIDs, then sold dialing
plans to route incoming calls through the PRIs out the Internet. This is not
the issue though. We definitely have to migrate to an SS7 capable platform,
because that is the only way the Telco allows them to resell the dial-up
connections (not ISDN), and Asterisk is the current bet.

If we can get Asterisk to pick up those calls via SS7, then authenticate
them, send them out to the Internet, we would be achieving a %100 usage on
the Digium cards, because one of them wouldn't be used to talk to the AS.

Can Asterisk do this?


Thanks again,

*José Pablo Méndez
   *


On Wed, Nov 24, 2010 at 7:59 PM, Cary Fitch  wrote:

>  I am not sure where you are and what legal conventions are involved.
>
>
>
> Are you saying the Telco (and legal restrictions) say you can’t send calls
> to the internet via the AS5300 but you can if Asterisk does it directly?
> What is the “logic” in that?
>
>
>
> Or are they saying your Telco to Asterisk trunks have to be SS7 controlled?
>
>
>
>
> Or are you concerned about Asterisk handling the TDM to IP conversion in an
> adequate manner?
>
>
>
> I am not sure/aware myself that Asterisk will do a modem to IP conversion.
> I think in your example the AS5300 is doing that.
>
>
>
> What is the Telco’s problem in doing what the customer was doing before?
>
>
>
> Feel free to correspond directly if you want to.
>
>
>
> Cary Fitch
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *José Pablo Méndez
> Soto
> *Sent:* Wednesday, November 24, 2010 7:31 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Incoming calls through SS7 for data
> modemtransmissions - possible??
>
>
>
> Hello,
>
> We are working on implementing a solution for a medium service provider.
> They were previously using a Cisco AS5300 gateway with some PRI trunks to
> receive modem calls, then route them out the Internet.
>
> The Telco they were buying the trunks to discovered this configuration and
> restricted them due to legal conventions, and stated that in order to
> continue doing this, they would have to talk SS7 directly.
>
> We are planning on solving this by placing an Asterisk server with some
> TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the
> AS5300 for the dial-up to complete after authenticating against a RADIUS
> server.
>
> My questions is: can we use only Asterisk to complete/terminate the dial-up
> connection, removing the AS5300 out of the picture?
>
> Current topology to be set-up:
> Telco --> SS7 --> TE410P-AsteriskServer --> ISDN --> AS5300 --> Internet
>
> Ideal topology:
> Telco --> SS7 --> TE410P-AsteriskServer --> Internet
>
>
> Some posts talk about zapRAS being able to accomplish this, not quite sure
> though
>
> Sounds like possible:
> http://lists.digium.com/pipermail/asterisk-users/2004-January/026956.html
>
> Sounds like not possible:
> http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html
>
>
> Thanks in advance,
>
>
> *José Pablo Méndez**
>   *
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Why doesn't Asterisk project document certain important features of Asterisk officially?

[Sherwood McGowan]

On Wed, Nov 24, 2010 at 8:06 PM, Paul Belanger  wrote:
> On 10-11-24 08:34 PM, Sherwood McGowan wrote:
>> True, but then some of us registered on that site and still don't have
>> the ability to edit...I thought it was a community effort? Maybe I was
>> wrong
>>
> Once registered you will be able to post comments, not edit.  If you
> would like to become part of the documentation process, I would
> recommend talking to some on #asterisk-dev.
>
> It is my understanding there will be workflows around the wiki content
> and a reviews process for new submissions.
>
> --
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

Hey thanks for the reply mate! I'll pop by the -dev channel soon and
see about getting in on the project. INSERTDEITYNAMEHERE knows I've
been in the community long enough that I should warrant at least a
probational add to the process...

Slainte,
Sherwood McGowan
That guy who bugged Murf like CRAZY about the Macro iteration bug, and
several AEL items ;-)

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Re: [asterisk-users] Why doesn't Asterisk project document certain important features of Asterisk officially?

[Paul Belanger]

On 10-11-24 08:34 PM, Sherwood McGowan wrote:
> True, but then some of us registered on that site and still don't have
> the ability to edit...I thought it was a community effort? Maybe I was
> wrong
>
Once registered you will be able to post comments, not edit.  If you 
would like to become part of the documentation process, I would 
recommend talking to some on #asterisk-dev.

It is my understanding there will be workflows around the wiki content 
and a reviews process for new submissions.

-- 
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible??

[Cary Fitch]

I am not sure where you are and what legal conventions are involved.

 

Are you saying the Telco (and legal restrictions) say you can’t send calls
to the internet via the AS5300 but you can if Asterisk does it directly?
What is the “logic” in that?

 

Or are they saying your Telco to Asterisk trunks have to be SS7 controlled? 

 

Or are you concerned about Asterisk handling the TDM to IP conversion in an
adequate manner?

 

I am not sure/aware myself that Asterisk will do a modem to IP conversion.
I think in your example the AS5300 is doing that.

 

What is the Telco’s problem in doing what the customer was doing before?

 

Feel free to correspond directly if you want to.

 

Cary Fitch

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of José Pablo
Méndez Soto
Sent: Wednesday, November 24, 2010 7:31 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Incoming calls through SS7 for data
modemtransmissions - possible??

 

Hello,

We are working on implementing a solution for a medium service provider.
They were previously using a Cisco AS5300 gateway with some PRI trunks to
receive modem calls, then route them out the Internet.

The Telco they were buying the trunks to discovered this configuration and
restricted them due to legal conventions, and stated that in order to
continue doing this, they would have to talk SS7 directly.

We are planning on solving this by placing an Asterisk server with some
TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the
AS5300 for the dial-up to complete after authenticating against a RADIUS
server.

My questions is: can we use only Asterisk to complete/terminate the dial-up
connection, removing the AS5300 out of the picture?

Current topology to be set-up:
Telco --> SS7 --> TE410P-AsteriskServer --> ISDN --> AS5300 --> Internet

Ideal topology:
Telco --> SS7 --> TE410P-AsteriskServer --> Internet


Some posts talk about zapRAS being able to accomplish this, not quite sure
though

Sounds like possible:
http://lists.digium.com/pipermail/asterisk-users/2004-January/026956.html
 

Sounds like not possible:
http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html


Thanks in advance,


José Pablo Méndez
  

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Re: [asterisk-users] Why doesn't Asterisk project document certain important features of Asterisk officially?

[Sherwood McGowan]

On Wed, Nov 24, 2010 at 7:20 PM, Warren Selby  wrote:
> On Wed, Nov 24, 2010 at 11:30 AM, Tilghman Lesher 
> wrote:
>>
>> On Wednesday 24 November 2010 11:07:40 Bruce B wrote:
>> > This is not to bash the Asterisk project or Digium. Don't respond if you
>> > have a difference of opinion as I am not looking for personal opinions
>> > but rather JUST WONDERING THE TECHNICAL/TACTICAL CAUSE of not
>> > documenting thing that are really hidden from the community and one has
>> > to go through a much manual process to find out about.
>>
>> It's not a tactical or even a conscious decision.  The documentation arose
>> at voip-info organically, from the days when there was only a single
>> committer to Asterisk.  There's never been much of a coordinated attempt
>> to replace that resource, mostly due to the large amount of labor that we
>> would need.  There are certainly efforts to document Asterisk and keep the
>> documentation up to date (see the Asterisk Documentation Project).
>> Anything that you would like to contribute in this regard would be
>> welcome.
>
> I thought that's what http://wiki.asterisk.org was supposed to be for?
>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.selbytech.com
>
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>


True, but then some of us registered on that site and still don't have
the ability to edit...I thought it was a community effort? Maybe I was
wrong

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[asterisk-users] Incoming calls through SS7 for data modem transmissions - possible??

[José Pablo Méndez Soto]

Hello,

We are working on implementing a solution for a medium service provider.
They were previously using a Cisco AS5300 gateway with some PRI trunks to
receive modem calls, then route them out the Internet.

The Telco they were buying the trunks to discovered this configuration and
restricted them due to legal conventions, and stated that in order to
continue doing this, they would have to talk SS7 directly.

We are planning on solving this by placing an Asterisk server with some
TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the
AS5300 for the dial-up to complete after authenticating against a RADIUS
server.

My questions is: can we use only Asterisk to complete/terminate the dial-up
connection, removing the AS5300 out of the picture?

Current topology to be set-up:
Telco --> SS7 --> TE410P-AsteriskServer --> ISDN --> AS5300 --> Internet

Ideal topology:
Telco --> SS7 --> TE410P-AsteriskServer --> Internet


Some posts talk about zapRAS being able to accomplish this, not quite sure
though

Sounds like possible:
http://lists.digium.com/pipermail/asterisk-users/2004-January/026956.html

Sounds like not possible:
http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html


Thanks in advance,


*José Pablo Méndez
  *
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Re: [asterisk-users] Why doesn't Asterisk project document certain important features of Asterisk officially?

[Warren Selby]

On Wed, Nov 24, 2010 at 11:30 AM, Tilghman Lesher wrote:

> On Wednesday 24 November 2010 11:07:40 Bruce B wrote:
> > This is not to bash the Asterisk project or Digium. Don't respond if you
> > have a difference of opinion as I am not looking for personal opinions
> > but rather JUST WONDERING THE TECHNICAL/TACTICAL CAUSE of not
> > documenting thing that are really hidden from the community and one has
> > to go through a much manual process to find out about.
>
> It's not a tactical or even a conscious decision.  The documentation arose
> at voip-info organically, from the days when there was only a single
> committer to Asterisk.  There's never been much of a coordinated attempt
> to replace that resource, mostly due to the large amount of labor that we
> would need.  There are certainly efforts to document Asterisk and keep the
> documentation up to date (see the Asterisk Documentation Project).
> Anything that you would like to contribute in this regard would be welcome.
>

I thought that's what http://wiki.asterisk.org was supposed to be for?

-- 
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--Warren Selby, dCAP
http://www.selbytech.com
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Re: [asterisk-users] DAHDI phantom pickup when ringing

[Jonathan Hunter]

Tzafrir,

On 24 November 2010 18:12, Tzafrir Cohen  wrote:

>
> Can you replicate those phantom answers without calling all channels?
>
> Try:
>
>  originate DAHDI/7 application Echo
>
> Does that line answer without you picking up the phone? Or does it
> require a combination of several channels ringing at the same time?
>

Good thinking - I tried this, and also Lye's suggestion to swap channels.

When calling just one channel at a time (the command above didn't work for
me from the command line - is it a Manager command? Either way, I set up an
extension to Dial() just that DAHDI channel to test it), I didn't manage to
reproduce the problem.

Admittedly this isn't an exhaustive test - I may just not have triggered the
right conditions this time - but it does indicate to me that it has
something to do with ringing multiple channels at once.

I then swapped the channels around (5 became 23, and 7 became 14). This
seemed to work better at first - I was able to make three or four calls
without any phantom pickups. Sadly, DAHDI/14 then picked up and proved that
it wasn't quite fixed :)

We've certainly eliminated a couple of things already, which is good...

Thank you, all, for your continued help!

Jonathan
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Re: [asterisk-users] action at registering or de-registering

[Sherwood McGowan]

On Wed, Nov 24, 2010 at 4:24 PM, Hans Witvliet  wrote:
> On Wed, 2010-11-24 at 15:47 -0600, Sherwood McGowan wrote:
>> On Wed, Nov 24, 2010 at 3:08 PM, Hans Witvliet  wrote:
>> > On Wed, 2010-11-24 at 08:29 -0500, Ryan Bullock wrote:
>> >> On Asterisk 1.8 when a SIP peer resgisters or unregisters it generates
>> >> a PeerStatus event. I don't know if this is in 1.4/1.6 as well, but
>> >> should be easy enough to test.
>> >>
>> >> Here is an example of what I see on the manager interface during a
>> >> register/unregister:
>> >>
>> >> Event: PeerStatus
>> >> Privilege: system,all
>> >> ChannelType: SIP
>> >> Peer: SIP/twinkle
>> >> PeerStatus: Registered
>> >> Address: 192.168.56.1:5068
>> >>
>> >> Event: PeerStatus
>> >> Privilege: system,all
>> >> ChannelType: SIP
>> >> Peer: SIP/twinkle
>> >> PeerStatus: Unregistered
>> >>
>> >> I think that should work for whatever you need to do.
>> >>
>> >
>> > I'm doing a fresh install, so 1.8 is what i'm going to use.
>> >
>> > What i want to check, is whether to person who is doing a register, is
>> > realy the person at the other end of a VPN-tunnel.
>> > With openvpn i'm absolutely sure which person is at a certain
>> > vpn-ip-addres. I must check if the registering is faked or not.
>> >
>> > As ong as linphone (or for that matter any other softphone) does not
>> > have a possibility for using the libraries from opensc, there is no
>> > other way...
>> >
>> > So next couple of weeks i'll start exploring AMI,
>> >
>> > Thanks!
>> >
>
>> >
>>
>> Well, if that's all you need (restricting registrations for a SIP
>> endpoint to a specific IP address), try one of the following
>> methods...
>>
>> Method 1:
>> In the endpoint definition, set the host to the vpn ip address, rather
>> than setting it to dynamic. This disallows registrations. Then, use
>> qualify=yes so Asterisk "knows" when the endpoint is available
>> (responding to OPTIONS requests).
>>
>> Method 2:
>> Use the permit,deny, and mask settings to define what ip address
>> and/or network the endpoint should be at, thereby locking out use from
>> another address.
>> (http://www.voip-info.org/wiki/view/Asterisk+sip+permit-deny-mask)
>>
>> Either of those should resolve your needs
>
>
> No, don't think so, (unless mistaken)
> Everybody got a dynamic address from openvpn, something in 10.225.0.0/16
> You never know what you wil get, so it got to be dynamic.
>
> Anybody within that range is a valid user (otherwise he could not set up
> the vpn-tunnel). But any rogue co-worker should not be able to register
> as another co-worker, so method-2 won't do either.
>
> sip/tls might have been a solution, but private keys are locked on a
> card, and can ony be reached with the pkcs11-libs from opensc.
>
> Hans
>
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>

Ah, I see, sorry I misunderstood what you needed. Good luck

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Re: [asterisk-users] action at registering or de-registering

[Hans Witvliet]

On Wed, 2010-11-24 at 15:47 -0600, Sherwood McGowan wrote:
> On Wed, Nov 24, 2010 at 3:08 PM, Hans Witvliet  wrote:
> > On Wed, 2010-11-24 at 08:29 -0500, Ryan Bullock wrote:
> >> On Asterisk 1.8 when a SIP peer resgisters or unregisters it generates
> >> a PeerStatus event. I don't know if this is in 1.4/1.6 as well, but
> >> should be easy enough to test.
> >>
> >> Here is an example of what I see on the manager interface during a
> >> register/unregister:
> >>
> >> Event: PeerStatus
> >> Privilege: system,all
> >> ChannelType: SIP
> >> Peer: SIP/twinkle
> >> PeerStatus: Registered
> >> Address: 192.168.56.1:5068
> >>
> >> Event: PeerStatus
> >> Privilege: system,all
> >> ChannelType: SIP
> >> Peer: SIP/twinkle
> >> PeerStatus: Unregistered
> >>
> >> I think that should work for whatever you need to do.
> >>
> >
> > I'm doing a fresh install, so 1.8 is what i'm going to use.
> >
> > What i want to check, is whether to person who is doing a register, is
> > realy the person at the other end of a VPN-tunnel.
> > With openvpn i'm absolutely sure which person is at a certain
> > vpn-ip-addres. I must check if the registering is faked or not.
> >
> > As ong as linphone (or for that matter any other softphone) does not
> > have a possibility for using the libraries from opensc, there is no
> > other way...
> >
> > So next couple of weeks i'll start exploring AMI,
> >
> > Thanks!
> >

> >
> 
> Well, if that's all you need (restricting registrations for a SIP
> endpoint to a specific IP address), try one of the following
> methods...
> 
> Method 1:
> In the endpoint definition, set the host to the vpn ip address, rather
> than setting it to dynamic. This disallows registrations. Then, use
> qualify=yes so Asterisk "knows" when the endpoint is available
> (responding to OPTIONS requests).
> 
> Method 2:
> Use the permit,deny, and mask settings to define what ip address
> and/or network the endpoint should be at, thereby locking out use from
> another address.
> (http://www.voip-info.org/wiki/view/Asterisk+sip+permit-deny-mask)
> 
> Either of those should resolve your needs


No, don't think so, (unless mistaken)
Everybody got a dynamic address from openvpn, something in 10.225.0.0/16
You never know what you wil get, so it got to be dynamic.

Anybody within that range is a valid user (otherwise he could not set up
the vpn-tunnel). But any rogue co-worker should not be able to register
as another co-worker, so method-2 won't do either.

sip/tls might have been a solution, but private keys are locked on a
card, and can ony be reached with the pkcs11-libs from opensc.

Hans

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Re: [asterisk-users] Avoided deadlock Error

[bayardo . sanchez]

Othe problem is small time my hdd is full of recording 
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-Original Message-
From: Stefan Schmidt 
Sender: asterisk-users-boun...@lists.digium.com
Date: Wed, 24 Nov 2010 22:59:56 
To: Asterisk Users Mailing List - Non-Commercial 
Discussion
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] Avoided deadlock Error

Am 24.11.2010 13:48, schrieb Bayardo Sanchez:
> My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem
> is :
> 
> Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
> deadlock for '0x861f6d8', 9 retries!
> Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
> deadlock for '0x85a6420', 9 retries!
> Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
> deadlock for '0x85bc510', 9 retries!
> Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
> deadlock for '0x85f9e68', 9 retries!
> Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
> deadlock for '0x85e1db0', 9 retries!
> 
> this error comes only when I call spain saturated my CLI with the message
> error
> 
> 
hello,

as tilghman noticed 1.2 is EOL, but i still use it too and i see a bunch
of this messages on different servers and they dont cause any problem at
all.

if you have some problems with this (except the warning message) you
should upgrade.

best regards

stefan

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Re: [asterisk-users] Avoided deadlock Error

[Stefan Schmidt]

Am 24.11.2010 13:48, schrieb Bayardo Sanchez:
> My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem
> is :
> 
> Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
> deadlock for '0x861f6d8', 9 retries!
> Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
> deadlock for '0x85a6420', 9 retries!
> Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
> deadlock for '0x85bc510', 9 retries!
> Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
> deadlock for '0x85f9e68', 9 retries!
> Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
> deadlock for '0x85e1db0', 9 retries!
> 
> this error comes only when I call spain saturated my CLI with the message
> error
> 
> 
hello,

as tilghman noticed 1.2 is EOL, but i still use it too and i see a bunch
of this messages on different servers and they dont cause any problem at
all.

if you have some problems with this (except the warning message) you
should upgrade.

best regards

stefan

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Re: [asterisk-users] action at registering or de-registering

[Sherwood McGowan]

On Wed, Nov 24, 2010 at 3:08 PM, Hans Witvliet  wrote:
> On Wed, 2010-11-24 at 08:29 -0500, Ryan Bullock wrote:
>> On Asterisk 1.8 when a SIP peer resgisters or unregisters it generates
>> a PeerStatus event. I don't know if this is in 1.4/1.6 as well, but
>> should be easy enough to test.
>>
>> Here is an example of what I see on the manager interface during a
>> register/unregister:
>>
>> Event: PeerStatus
>> Privilege: system,all
>> ChannelType: SIP
>> Peer: SIP/twinkle
>> PeerStatus: Registered
>> Address: 192.168.56.1:5068
>>
>> Event: PeerStatus
>> Privilege: system,all
>> ChannelType: SIP
>> Peer: SIP/twinkle
>> PeerStatus: Unregistered
>>
>> I think that should work for whatever you need to do.
>>
>
> I'm doing a fresh install, so 1.8 is what i'm going to use.
>
> What i want to check, is whether to person who is doing a register, is
> realy the person at the other end of a VPN-tunnel.
> With openvpn i'm absolutely sure which person is at a certain
> vpn-ip-addres. I must check if the registering is faked or not.
>
> As ong as linphone (or for that matter any other softphone) does not
> have a possibility for using the libraries from opensc, there is no
> other way...
>
> So next couple of weeks i'll start exploring AMI,
>
> Thanks!
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

Well, if that's all you need (restricting registrations for a SIP
endpoint to a specific IP address), try one of the following
methods...

Method 1:
In the endpoint definition, set the host to the vpn ip address, rather
than setting it to dynamic. This disallows registrations. Then, use
qualify=yes so Asterisk "knows" when the endpoint is available
(responding to OPTIONS requests).

Method 2:
Use the permit,deny, and mask settings to define what ip address
and/or network the endpoint should be at, thereby locking out use from
another address.
(http://www.voip-info.org/wiki/view/Asterisk+sip+permit-deny-mask)

Either of those should resolve your needs

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Re: [asterisk-users] Audiocodes firmware

[Joseph]

On 10/14/10 15:38, Bryant Zimmerman wrote:
>For which device models?
>
>
> From: "Mark Murawski" 
>Sent: Thursday, October 14, 2010 3:26 PM
>To: asterisk-users@lists.digium.com
>Subject: [asterisk-users] Audiocodes firmware
>
>Does anyone have links to the most recent audiocodes firmware?
>
>

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What firmware are you running? I think the latest one is 5.8

-- 
Joseph

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Re: [asterisk-users] Someone has hacked into our system

[Joseph]

On 11/24/10 10:39, Gary Kuznitz  wrote:

>> Look for "allowguest" default is "yes"
>> I change it to allowguest=no
>> In addition you might want to restrict some countries in your dial-plan, 
>> here is my list:
>
>This would be great.  Can I put this anyplace in extensions.conf?
>Or does it need to go after [DLPN_DialPlanl]  ?
>
>Thanks,
>
>Gary Kuznitz

This is in sip.conf

[general]
context=default ; Default context for incoming calls
allowguest=no   ; Allow or reject guest calls (default is yes)
...

-- 
Joseph

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Re: [asterisk-users] action at registering or de-registering

[Hans Witvliet]

On Wed, 2010-11-24 at 08:29 -0500, Ryan Bullock wrote:
> On Asterisk 1.8 when a SIP peer resgisters or unregisters it generates
> a PeerStatus event. I don't know if this is in 1.4/1.6 as well, but
> should be easy enough to test.
> 
> Here is an example of what I see on the manager interface during a
> register/unregister:
> 
> Event: PeerStatus
> Privilege: system,all
> ChannelType: SIP
> Peer: SIP/twinkle
> PeerStatus: Registered
> Address: 192.168.56.1:5068
> 
> Event: PeerStatus
> Privilege: system,all
> ChannelType: SIP
> Peer: SIP/twinkle
> PeerStatus: Unregistered
> 
> I think that should work for whatever you need to do.
> 

I'm doing a fresh install, so 1.8 is what i'm going to use.

What i want to check, is whether to person who is doing a register, is
realy the person at the other end of a VPN-tunnel.
With openvpn i'm absolutely sure which person is at a certain
vpn-ip-addres. I must check if the registering is faked or not.

As ong as linphone (or for that matter any other softphone) does not
have a possibility for using the libraries from opensc, there is no
other way...

So next couple of weeks i'll start exploring AMI,

Thanks!


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Re: [asterisk-users] Someone has hacked into our system

[Gary Kuznitz]

Thank you for the reply.

On 23 Nov 2010 at 18:51, John (John Novack ) 
commented about Re: [asterisk-users] Someone has hacked into our :

> 
> 
> Gary Kuznitz wrote:
> > Thank you for the reply...
> >
> > Comments below...
> > On 22 Nov 2010 at 17:23, Tilghman (Tilghman Lesher > us...@lists.digium.com>) commented about Re: [asterisk-users] Someone has 
> > hacked
> > into our :
> >
> >
> >> On Monday 22 November 2010 17:10:31 Gary Kuznitz wrote:
> >>  
> >>> I have the log now. I'd like to know what to look for in trying to figure
> >>> out how the calls are getting originated. I'd be happy to shere all the
> >>> information. I just don't want to post information on this public list 
> >>> that
> >>> might show other people how to get in to our box.
> >>>
> >> allowguest=yes in sip.conf, with a context= in the [general] section that
> >> is permitted to make outbound calls?
> >>  
> > I'm trying to understand exactly what this means.
> >
> > I found a sip.conf in /etc/asterisk
> > I have a [general] section.
> > I don't have allowguest=yes.  Is that good or am I supposed to have it?
> >
> I believe what you SHOULD have is;
> allowguest=no
> Not sure if that is the default behavior or not
> > If I'm supposed to have it can it go any place in the [general] section?
> > I have in the [general] section a line with:
> > context = default
> > Is this where I would remove default and enter the IP addresses that are 
> > allowed to
> > make calls?
> >
> Your default context in extensions.conf should basiclly lead nowhere.
> I have mine set up to play an insane laugh then hangup
> Probably safe to say NEVER use context default for any outbound calling

I don't have any context in extensions.conf
I do have context = default in sip.conf
Should I remove that line?
Could you give me an example of what you have in your extensions.conf?

Thank you,

Gary Kuznitz
> 
> You should also have, in general:
> 
> alwaysauthreject=yes
> This seems pretty effective in stopping some hacking
> These are simple fixes.
> I will let others comment on other more detailed firewalling
> 
> John Novack
> 
> > What would a line with IP address look like?  Could you give me an example?
> > If that isn't where the IP address that are allowed supposed to be where 
> > would I put
> > them?
> >
> > Thank you,
> >
> > Gary Kuznitz
> >
> >
> >> Just a guess, but there have been
> >> more than a few such discussions on the list about that configuration, plus
> >> a README-SERIOUSLY.bestpractices.txt in the root directory of every 
> >> Asterisk
> >> source tree.  You DID read that file, right?
> >>
> >> -- 
> >> Tilghman Lesher
> >> Digium, Inc. | Senior Software Developer
> >> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> >> Check us out at: www.digium.com&  www.asterisk.org
> >>
> >> -- 
> >> _
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >> New to Asterisk? Join us for a live introductory webinar every Thurs:
> >> http://www.asterisk.org/hello
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >>  
> >
> >
> >
> 
> -- 
> 
> Dog is my Co-pilot
> 



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Re: [asterisk-users] Recommended *WRT router to run Asterisk?

[Gilles]

On Fri, 19 Nov 2010 10:15:40 -0500, jon pounder 
wrote:
>What is nice is when the $50 hardware and the $1000 hardware run exactly 
>the same software so other than the drivers for the hardware itself, 
>everything else behaves the same way and its easy to move around 
>configurations to grow. (I am not talking about asterisk specifically, 
>just generally about routers, backup devices, media servers, etc)

And what's even better, is when the customer is fine with spending
$1000 for a box that only costs $50 ;-)

Thanks everyone for the great feedback. I'll investigate the different
options listed above.


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Re: [asterisk-users] Someone has hacked into our system

[Gary Kuznitz]

On 23 Nov 2010 at 16:54, Joseph (Joseph ) commented about 
Re: [asterisk-users] Someone has hacked into our :

> On 11/23/10 14:18, Gary Kuznitz  wrote:
> >Thank you for the reply...
> >
> >Comments below...
> >On 22 Nov 2010 at 17:23, Tilghman (Tilghman Lesher  >us...@lists.digium.com>) commented about Re: [asterisk-users] Someone has 
> >hacked
> >into our :
> >
> >> On Monday 22 November 2010 17:10:31 Gary Kuznitz wrote:
> >> > I have the log now. I'd like to know what to look for in trying to figure
> >> > out how the calls are getting originated. I'd be happy to shere all the
> >> > information. I just don't want to post information on this public list 
> >> > that
> >> > might show other people how to get in to our box.
> >>
> >> allowguest=yes in sip.conf, with a context= in the [general] section that
> >> is permitted to make outbound calls?
> >
> >I'm trying to understand exactly what this means.
> >
> >I found a sip.conf in /etc/asterisk
>   
> >I have a [general] section.
> >I don't have allowguest=yes.  Is that good or am I supposed to have it?
> 
> Look for "allowguest" default is "yes"
> I change it to allowguest=no
> In addition you might want to restrict some countries in your dial-plan, here 
> is my list:

This would be great.  Can I put this anyplace in extensions.conf?
Or does it need to go after [DLPN_DialPlanl]  ?

Thanks,

Gary Kuznitz

> [blocked-numbers]
> ;block bahamas, etc
>  exten => _91900.,1,congestion; N11
>  exten => _91XXX976.,1,congestion ; N11
>  exten => _91XXX555.,1,congestion ; N11
>  exten => _91X11.,1,congestion; N11
>  exten => _91867.,1,congestion; Yukon (sorry mike)
> 
>  ;exten => _1NPA Country
>  exten => _91232.,1,congestion;   Sierra Leone
>  exten => _91242.,1,congestion;   BAHAMAS
>  exten => _91246.,1,congestion;   BARBADOS
>  exten => _91264.,1,congestion;   ANGUILLA
>  exten => _91268.,1,congestion;   ANTIGUA/BARBUDA
>  exten => _91284.,1,congestion;   BRITISH VIRGIN ISLANDS
>  exten => _91345.,1,congestion;   CAYMAN ISLANDS
>  exten => _91441.,1,congestion;   BERMUDA
>  exten => _91473.,1,congestion;   GRENADA
>  exten => _91649.,1,congestion;   TURKS & CAICOS ISLANDS
>  exten => _91664.,1,congestion;   MONTSERRAT
>  exten => _91758.,1,congestion;   ST. LUCIA
>  exten => _91767.,1,congestion;   DOMINICA
>  exten => _91784.,1,congestion;   ST. VINCENT & GRENADINES
>  exten => _91809.,1,congestion;   DOMINICAN REPUBLIC
>  exten => _91829.,1,congestion;   DOMINICAN REPUBLIC
>  exten => _91868.,1,congestion;   TRINIDAD AND TOBAGO
>  exten => _91869.,1,congestion;   ST. KITTS AND NEVIS
>  exten => _91876.,1,congestion;   JAMAICA
> 
> -- 
> Joseph



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Re: [asterisk-users] DAHDI phantom pickup when ringing

[Tzafrir Cohen]

On Sun, Nov 21, 2010 at 11:13:00PM +, Jonathan Hunter wrote:
> Hi,
> 
> I've been experiencing trouble with my DAHDI channels for some time and have
> finally decided to try and resolve the issue.
> 
> Essentially, the problem I am having is that when a call comes in, and my
> DAHDI phones therefore ring, Asterisk thinks that one of the handsets has
> picked up to answer the incoming call - whereas in actual fact it is still
> on hook. The call then gets instantly dropped (the phone is on-hook, after
> all), and the caller has to redial.
> 
> Sample log (this is an incoming call from SIP/, that was "phantom
> answered" by DAHDI/7. There is no time delay between the 'answered' line and
> the 'hungup' line):
> 
> -- DAHDI/5-1 is ringing
> -- DAHDI/3-1 is ringing
> -- DAHDI/7-1 is ringing
> -- DAHDI/2-1 is ringing
> -- DAHDI/4-1 is ringing
> -- DAHDI/6-1 is ringing
> -- DAHDI/7-1 answered SIP/-011f
> -- Hanging up on 'DAHDI/6-1'
> -- Hungup 'DAHDI/6-1'
> -- Hanging up on 'DAHDI/4-1'
> -- Hungup 'DAHDI/4-1'
> -- Hanging up on 'DAHDI/2-1'
> -- Hungup 'DAHDI/2-1'
> -- Hanging up on 'DAHDI/5-1'
> -- Hungup 'DAHDI/5-1'
> -- Hanging up on 'DAHDI/3-1'
> -- Hungup 'DAHDI/3-1'
> -- Hanging up on 'DAHDI/1-1'
> -- Hungup 'DAHDI/1-1'
> -- Hanging up on 'DAHDI/7-1'
> -- Hungup 'DAHDI/7-1'
>   == Spawn extension (macro-stdexten, s, 3) exited non-zero on
> 'SIP/-011f'

Can you replicate those phantom answers without calling all channels?

Try:

  originate DAHDI/7 application Echo

Does that line answer without you picking up the phone? Or does it
require a combination of several channels ringing at the same time?

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

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[asterisk-users] Disable connected line updates for dahdi PRI channel

[Michael Smith]

Hi,

Starting in Asterisk 1.8.0, Asterisk supports connected line updates. 
This is fantastic for SIP. How can I prevent them from being sent to a 
PRI channel?

I'm having problems when a call is answered by an internal SIP 
extension, then transferred (blind or attended) to another internal SIP 
extension. One of my PRI providers can't handle the ROSE_ETSI_EctInform 
APDU and drops the call.

Looking through the chan_dahdi and sig_pri code, I don't see any 
configuration flag to block the updates from going through. I was even 
hoping I could set __CONNECTED_LINE_CALLER_SEND_MACRO to something 
bogus, but it looks like if the macro fails to execute in any way, the 
code will just go ahead and update the connected line data.

I wouldn't be surprised if the provider has the same issue with 
redirecting updates.

Here's what happens when external number 87133306 calls into my PRI, 
extension  answers, does an attended transfer to 0102, and completes 
the transfer. The provider eventually hangs up with "Message not 
compatible with call state (101)".

channel.c: Released clone lock on 'SIP/-0007'
channel.c: Done Masquerading DAHDI/i1/87133306-3 (6)
chan_dahdi.c: Requested indication 26 on channel DAHDI/i1/87133306-3
chan_dahdi.c: Requested indication 17 on channel DAHDI/i1/87133306-3
channel.c: Bridge stops because we're zombie or need a soft hangup: 
c0=SIP/-0007, c1=SIP/-0006, flags: Yes,Yes,No,No
channel.c: Bridge stops bridging channels SIP/-0007 and 
SIP/-0006
chan_dahdi.c: Requested indication 22 on channel DAHDI/i1/87133306-3
sig_pri.c: Received AST_CONTROL_CONNECTED_LINE on DAHDI/i1/87133306-3
chan_dahdi.c: 1 Adding facility ie contents to send in FACILITY message:
chan_dahdi.c: 1 ASN.1 dump
chan_dahdi.c: 1   Context Specific/C [1 0x01]  Len:24 <18>
chan_dahdi.c: 1 Integer(2 0x02) <02> Len:1 <01>
chan_dahdi.c: 1   <03> - "~"
chan_dahdi.c: 1 OID(6 0x06) <06> Len:6 <06>
chan_dahdi.c: 1   <04 00 82 71 01 05> - "~~~q~~"
chan_dahdi.c: 1 Sequence/C(48 0x30) <30> Len:11 <0B>
chan_dahdi.c: 1   Enumerated(10 0x0A) <0A> Len:1 <01>
chan_dahdi.c: 1 <01> - "~"
chan_dahdi.c: 1   Context Specific/C [0 0x00]  Len:6 <06>
chan_dahdi.c: 1 Context Specific [0 0x00] <80> Len:4 <04>
chan_dahdi.c: 1   <30 31 30 32> - "0102"
chan_dahdi.c: 1 ASN.1 end
chan_dahdi.c: 1 INVOKE Component Context Specific/C [1 0x01]
chan_dahdi.c: 1   invokeId Integer(2 0x02) = 3 0x0003
chan_dahdi.c: 1   operationValue OID(6 0x06) = 4.0.369.1.5
chan_dahdi.c: 1   operationValue = ROSE_ETSI_EctInform
chan_dahdi.c: 1   EctInform Sequence/C(48 0x30)
chan_dahdi.c: 1   callStatus Enumerated(10 0x0A) = 1 0x0001
chan_dahdi.c: 1   redirectionNumber PresentedNumberUnscreened
chan_dahdi.c: 1   Explicit Context Specific/C [0 0x00]
chan_dahdi.c: 1   presentationAllowedNumber PartyNumber
chan_dahdi.c: 1   unknownPartyNumber Context Specific [0 0x00] = "0102"
chan_dahdi.c: 1
chan_dahdi.c: 1 > DL-DATA request
chan_dahdi.c: 1 > Protocol Discriminator: Q.931 (8)  len=34
chan_dahdi.c: 1 > TEI=0 Call Ref: len= 2 (reference 43/0x2B) (Sent to 
originator)
chan_dahdi.c: 1 > Message Type: FACILITY (98)
chan_dahdi.c: 1 TEI=0 Transmitting N(S)=43, window is open V(A)=43 K=7
chan_dahdi.c: 1
chan_dahdi.c: 1 > Protocol Discriminator: Q.931 (8)  len=34
chan_dahdi.c: 1 > TEI=0 Call Ref: len= 2 (reference 43/0x2B) (Sent to 
originator)
chan_dahdi.c: 1 > Message Type: FACILITY (98)
chan_dahdi.c: 1 > [1c 1b 91 a1 18 02 01 03 06 06 04 00 82 71 01 05 30 0b 
0a 01 01 a0 06 80 04 30 31 30 32]
chan_dahdi.c: 1 > Facility (len=29, codeset=0) [ 0x91, 0xA1, 0x18, 0x02, 
0x01, 0x03, 0x06, 0x06, 0x04, 0x00, 0x82, 'q', 0x01, 0x05, '0', 0x0B, 
0x0A, 0x01, 0x01, 0xA0, 0x06, 0x80, 0x04, '0102' ]
channel.c: Hanging up channel 'SIP/-0006'
app_dial.c: Exiting with DIALSTATUS=ANSWER.
pbx.c: Spawn extension (inbound_all,s,2) exited non-zero on 
'SIP/-0007'
pbx.c:   == Spawn extension (inbound_all, s, 2) exited non-zero on 
'SIP/-0007'
channel.c: Soft-Hanging up channel 'SIP/-0007'
channel.c: Hanging up zombie 'SIP/-0007'
chan_sip.c: Stopping retransmission on 
'4e4d6d5a1152cff85ed7c18c6ed2b...@172.20.45.10' of Request 103: Match Found
res_rtp_asterisk.c: No remote address on RTP instance '0x7f3a46298488' 
so dropping frame
res_rtp_asterisk.c: No remote address on RTP instance '0x7f3a46298488' 
so dropping frame
res_rtp_asterisk.c: No remote address on RTP instance '0x7f3a46298488' 
so dropping frame
chan_sip.c: Stopping retransmission on 
'565a02b66888481b6862e89b53af3...@172.20.45.10' of Request 103: Match Found
res_rtp_asterisk.c: No remote address on RTP instance '0x7f3a46298488' 
so dropping frame
res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa29ef8'
res_rtp_asterisk.c: No remote address on RTP instance '0x7f3a46298488' 
so dropping frame
chan_dahdi.c: 1
chan_dahdi.c: 1 < Protocol Discriminator: Q.931 (8)  len=9
chan_dahdi.c: 1 < TEI=0 Call Ref: len

Re: [asterisk-users] kernel: dahdi: Detected time shift.

[William Stillwell (Lists)]

I know there was a patch for dahdi to fix server lockups on time shift. (not
sure what version, but if you changed the time, the server would just go
crash.)

 

Do you have the latest version ?

 

Check your ntpd settings to make sure your time isn't bouncing all over the
place.

 

 

 

William Stillwell

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Wednesday, November 24, 2010 11:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] kernel: dahdi: Detected time shift.

 

Hello list,

I'm experiencing a lot of server freezes lately. The server just... freezes.

I notice in the log files (/var/log/asterisk/messages & /var/log/messages)
that logging stops at the time the server hangs. Logging continues when the
server has been restarted (which is the only solution).

So it is not a proces that hangs, it's the entire server (CentOS5.5 +
Asterisk + MySQL).

I really have no idea what can be causing these sudden freezes. Memory stays
mostly at 250MB of 512 MB total, CPU is 97% to 100% idle...

/var/log/asterisk/debug tells me nothing, no lines that indicate something
strange before the freeze (debug level 9).

I have no core.pid file in /tmp, when I look after rebooting the server.

The only thing I have is a high level of mentionning of "kernel: dahdi:
Detected time shift." in /var/log/messages.


What is causing this kernel message ? Could this be the cause of the server
freeze ?

Thank you for every feedback you can give me.


Kind regards,
Jonas.

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Re: [asterisk-users] kernel: dahdi: Detected time shift.

[Shaun Ruffell]

On 11/24/2010 11:02 AM, Shaun Ruffell wrote:
> On 11/24/2010 10:43 AM, Jonas Kellens wrote:
>> The only thing I have is a high level of mentionning of "kernel: dahdi:
>> Detected time shift." in /var/log/messages.
>>
>>
>> What is causing this kernel message ? Could this be the cause of the
>> server freeze ?
>>
> 
> When a system does not have a hardware device installed DAHDI uses the
> system time in order to approximate a telephony clock source (what was
> previously referred to as dahdi_dummy).  Since it uses the system time
> to accomplish this, if there is a large gap in the time (typically
> because NTP is adjusting the time) DAHDI just reports that it thinks
> it's very far behind or ahead of where it should be, and doesn't even
> try to "mix" any audio for that interval.
> 
> So, I think the "detected time shift" is more a symptom of something
> else causing locks as opposed to the source.
> 
> Is your server keeping accurate wall-time?
> 

Another case where I regularly see "detected time shift" is when I'm
running DAHDI in a virtual machine and go through a suspend / resume
cycle.  On resume when the system clock is updated DAHDI will notice the
shift.

-- 
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] kernel: dahdi: Detected time shift.

[Shaun Ruffell]

On 11/24/2010 10:43 AM, Jonas Kellens wrote:
> Hello list,
> 
> I'm experiencing a lot of server freezes lately. The server just... freezes.
> 
> I notice in the log files (/var/log/asterisk/messages &
> /var/log/messages) that logging stops at the time the server hangs.
> Logging continues when the server has been restarted (which is the only
> solution).
> 
> So it is not a proces that hangs, it's the entire server (CentOS5.5 +
> Asterisk + MySQL).
> 
> I really have no idea what can be causing these sudden freezes. Memory
> stays mostly at 250MB of 512 MB total, CPU is 97% to 100% idle...
> 
> /var/log/asterisk/debug tells me nothing, no lines that indicate
> something strange before the freeze (debug level 9).
> 
> I have no core.pid file in /tmp, when I look after rebooting the server.
> 
> The only thing I have is a high level of mentionning of "kernel: dahdi:
> Detected time shift." in /var/log/messages.
> 
> 
> What is causing this kernel message ? Could this be the cause of the
> server freeze ?
> 

When a system does not have a hardware device installed DAHDI uses the
system time in order to approximate a telephony clock source (what was
previously referred to as dahdi_dummy).  Since it uses the system time
to accomplish this, if there is a large gap in the time (typically
because NTP is adjusting the time) DAHDI just reports that it thinks
it's very far behind or ahead of where it should be, and doesn't even
try to "mix" any audio for that interval.

So, I think the "detected time shift" is more a symptom of something
else causing locks as opposed to the source.

Is your server keeping accurate wall-time?

-- 
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] kernel: dahdi: Detected time shift.

[Mark Deneen]

On Wed, Nov 24, 2010 at 11:43 AM, Jonas Kellens
 wrote:
> Hello list,
>
> I'm experiencing a lot of server freezes lately. The server just... freezes.
>
> I notice in the log files (/var/log/asterisk/messages & /var/log/messages)
> that logging stops at the time the server hangs. Logging continues when the
> server has been restarted (which is the only solution).
>
> So it is not a proces that hangs, it's the entire server (CentOS5.5 +
> Asterisk + MySQL).
>
> I really have no idea what can be causing these sudden freezes. Memory stays
> mostly at 250MB of 512 MB total, CPU is 97% to 100% idle...
>
> /var/log/asterisk/debug tells me nothing, no lines that indicate something
> strange before the freeze (debug level 9).
>
> I have no core.pid file in /tmp, when I look after rebooting the server.
>
> The only thing I have is a high level of mentionning of "kernel: dahdi:
> Detected time shift." in /var/log/messages.
>
>
> What is causing this kernel message ? Could this be the cause of the server
> freeze ?
>
> Thank you for every feedback you can give me.

Jonas,

Do you have a monitor attached to the server?  If the kernel is
crashing, you might be able to catch the stack trace there.

-M

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[asterisk-users] Why doesn't Asterisk project document certain important features of Asterisk officially?

[Bruce B]

Hi Everyone,

I am wondering why documentation of some of the vital parts of Asterisk is
hosted on voipinfo.org (unreliable is some parts) and not on asterisk.org?
For example the list of AMI events are not well documented and one has to
guess which version supports which event. The documentation file for AMI for
Asterisk 1.4 is really only a startup guide and it doesn't even provide a
full list available events.

I am wondering if this is a labor tedious job for the programmers or is it a
tactical move in part of a party with some sort of interest? Because the
programmer(s) who does the programming for something like the AMI events may
as well do something known as Copy & Paste (very easy process of pressing
keys: Ctrl+C + Ctrl+V) of what s/he has changed in the source code and make
it available in a text document without much fancy editing even.

This is not to bash the Asterisk project or Digium. Don't respond if you
have a difference of opinion as I am not looking for personal opinions but
rather JUST WONDERING THE TECHNICAL/TACTICAL CAUSE of not documenting thing
that are really hidden from the community and one has to go through a much
manual process to find out about.

Thanks for the input.

-Bruce
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[asterisk-users] IPv6: What You Need to Know Now

[Randy R]

Yes, the thanksgiving holiday is here (in the USA)! But also, the fear
of running out of IP addresses next year has raised its ugly head and
since we don't do Thanksgiving in Europe, we have some serious talking
to do about this problem.

This Friday at 12 Noon EST, Olle Johansson will be joining us to
describe the state of the migration to IP v6 in VoIP-dom. Olle (@oej)
needs no introduction. His company, Edvina was founded in 1987. Since
then, they’ve promoted open standards, for networking, for application
integration and for software. Olle is a pillar of our community and an
outspoken critic of all that stands in the way of progress in his
areas of expertise.

We plan to cover the following issues (and more) :

- IPv6 – how to get it into your network today
- VoIP and IPv6 – why is this a good marriage?
- Experiences from Asterisk 1.8 IPv6 support
- Living in a dual stack world
- Dual stack considered bad for VoIP?
- Implementations out there – a call for help? Which devices used in
the VUC community supports IPv6? Any experiences?

If you can't be there, consider listening to the recording at a convenient time.

You can also listen live via mp3 stream now, or via SIP (especially
tasty in g722 wideband) or Skype, or even... PSTN. Watch the back
channel text on IRC. You'll find all the info at http://vuc.me - most
of the contact numbers are in the top banner.

SIP:200...@login.zipdx.com
Skype:vuc.me
mp3 stream: http://www.call2stream.com/stream/vuc-me.m3u
PSTN: +1 425-906-3916
iNum: +883 5100 123 94882

IRC: #vuc on Freenode.net or http://vuc.me/irc
VUC in your local time zone: http://vuc.me/next

Also, we are giving away a Gigaset SIP/DECT phone that day, so relax,
join us, mute if you need to belch after all that food and drink and
get ready for IPv6!

/r

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[asterisk-users] kernel: dahdi: Detected time shift.

[Jonas Kellens]

Hello list,

I'm experiencing a lot of server freezes lately. The server just... freezes.

I notice in the log files (/var/log/asterisk/messages & 
/var/log/messages) that logging stops at the time the server hangs. 
Logging continues when the server has been restarted (which is the only 
solution).


So it is not a proces that hangs, it's the entire server (CentOS5.5 + 
Asterisk + MySQL).


I really have no idea what can be causing these sudden freezes. Memory 
stays mostly at 250MB of 512 MB total, CPU is 97% to 100% idle...


/var/log/asterisk/debug tells me nothing, no lines that indicate 
something strange before the freeze (debug level 9).


I have no core.pid file in /tmp, when I look after rebooting the server.

The only thing I have is a high level of mentionning of "kernel: dahdi: 
Detected time shift." in /var/log/messages.



What is causing this kernel message ? Could this be the cause of the 
server freeze ?


Thank you for every feedback you can give me.


Kind regards,
Jonas.
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Re: [asterisk-users] callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk

[Olivier]

2010/11/20 Olivier 

>
> Depending on what telco Charlie is connected to would change the CallerId
> presented to Charlie from being Alice's or Bob's Cid.
>
> When a call is forwarded, Charlie's telco receives different ANI and CID :
> some (seems to) favor ANI and some CID.
>
> An interesting thing to test is to let Bob issue a simple call to Charlie
> using a fake CID such as 0123456789.
> Will Charlie's phone display this non-existent number or not ?
>


Another trap is recently discovered is :
sometimes, the call is presented to Charlie with Alice's name and Bob's
number !!

I met this one when Charlie is subscribed to an analog line with Caller Name
Service enabled.
For a reason I don't know, it seems that the numbers choosen to display the
name and the number are different (ANI and CID).
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Re: [asterisk-users] [Asterisk-users] asterisk-1.8.0 compilation error

[Paul Belanger]

On 10-11-24 06:09 AM, RAJNIKANT VANZA wrote:
> make[1]: *** [cdr_webservice.o] Error 1
> make: *** [cdr] Error 2
>
What is cdr_webservice.o ?

-- 
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[asterisk-users] Originate Response.

[Rodrigo Lang]

Hi to all.

I am conducting several tests with the Asterisk manager and I noticed
something that I believe to be a problem.

When I generate a call with the Action Originate with the Async option true,
the event OriginateResponse returns normally. But when I generate a call in
the same way, without the Async option, the event OriginateResponse does not
come.

Is this a bug? It was fixed in some version?

I use Asterisk version 1.6.0.28


Thanks in advance.
-- 
Rodrigo Lang
Opening your mind - Just another Open Source
site
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Re: [asterisk-users] SPA942 on speaker phone does not hang up?

[Peder]

It is the phone itself:  go to Regional tab and scroll down to Reorder Delay
and make it 255.  That tells it not to play re-order tone and just hangup.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith
Sent: Wednesday, November 24, 2010 5:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SPA942 on speaker phone does not hang up?

 

Hello all,

I am using Linksys SPA942 in my current installation activity. I see a
peculiar behavior: A call is made and the SPA942 uses its speaker. When the
far end of a call hangs up , the SPA942 stays off hook, and after a time
plays a fast busy. The user then has to press the line presence button to
hang up the phone.

 

I think I must be missing some sip.conf parameter. My sip.conf is pretty
simple for these extensions; here is what I am using now:

 

[extension1234]

mailbox=1...@default

type=friend

context=users

host=dynamic

secret=verysecret

 

I have looked at the sample sip.conf and did not get any clues, also the
SPA900 Admin Manual doesn't say anything about it.

 

Thanks

Cassius

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Re: [asterisk-users] action at registering or de-registering

[Ryan Bullock]

On Asterisk 1.8 when a SIP peer resgisters or unregisters it generates
a PeerStatus event. I don't know if this is in 1.4/1.6 as well, but
should be easy enough to test.

Here is an example of what I see on the manager interface during a
register/unregister:

Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/twinkle
PeerStatus: Registered
Address: 192.168.56.1:5068

Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/twinkle
PeerStatus: Unregistered

I think that should work for whatever you need to do.

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[asterisk-users] Avoided deadlock Error

[Bayardo Sanchez]

My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem
is :

Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
deadlock for '0x861f6d8', 9 retries!
Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
deadlock for '0x85a6420', 9 retries!
Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
deadlock for '0x85bc510', 9 retries!
Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
deadlock for '0x85f9e68', 9 retries!
Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
deadlock for '0x85e1db0', 9 retries!

this error comes only when I call spain saturated my CLI with the message
error

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[asterisk-users] DTMF CallerID

[Antonio Modesto]

Hi,

Does anyone know if CID is already working with Digium TDM800P card
using DTMF signalling?

(I'm brazillian)

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Re: [asterisk-users] Asterisk 1.6 and Music on Hold

[Sherwood McGowan]

On Wed, Nov 24, 2010 at 5:35 AM, Olivier CALVANO  wrote:
> 2010/11/24 Sherwood McGowan :
>> On Wed, Nov 24, 2010 at 4:30 AM, Olivier CALVANO  wrote:
>>> Hi
>>>
>>> i have a small problems on Asterisk 1.6 with the MusiconOld :
>>>
>>> musiconhold.conf:
>>>
>>> [Sound_1]
>>> mode=quietmp3
>>> directory=/var/lib/asterisk/moh/Sound_1
>>>
>>> in extensions.conf:
>>>
>>> exten => 0532xx,1,Answer
>>> exten => 0532xx,2,MusicOnHold(Sound_1)
>>> exten => 0532xx,3,Dial(SIP/ACCOUNT001,180,t)
>>> exten => 0532xx,4,Hangup
>>>
>>>
>>>
>>>
>>> When i call to the number, i have the Music "Sound_1" but the SIP Phone
>>> don't ring ...
>>>
>>> Where is my error ?
>>>
>>>
>>> and second question, can i said at asterisk that when he receive the call,
>>> he play the music from first second ? and repeat at the end of the music.
>>>
>>> Thanks for your help
>>>
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>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>> First, if you don't use the Music on hold command prior to the dial,
>> do you hear ringing? It seems to me that what's going on here is that
>> you're overriding the progress notification that results from the
>> device responding to the invite with "TRYING" or "RINGING" by running
>> MOH. If the ringing doesn't occur even when you remove the MOH
>> command, your device is probably not signaling properly and you'll
>> need to use the "r" option in your Dial command.
>
>
>
> Hi
>
> Thanks for your help, yes, if i don't put the music on hold command, the phone
> ringing. I have change for put the "r" but no effect
>
> bye
> olivier
>
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Olivier,
Your MusicOnHold(Sound_1) command is overriding the progress
indications that Asterisk would normally provide. Do you intend to
play music on hold, or are you just wishing to set the class for that
call? If the latter, use Set(CHANNEL(musicclass)=Sound_1). That would
NOT play the Music on hold, thereby allowing Asterisk to provide the
progress indications. If you mean to play the music, you're going to
have to understand that you won't be able to hear indications (Please
read http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial) such as
ringing.

Does that clear it up? Basically, you cna't have Music On Hold AND
Ringing for a channel going at the same time, they're mutually
exclusive

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Re: [asterisk-users] SPA942 on speaker phone does not hang up?

[Sherwood McGowan]

On Wed, Nov 24, 2010 at 5:16 AM, Cassius Smith  wrote:
> Hello all,
> I am using Linksys SPA942 in my current installation activity. I see a
> peculiar behavior: A call is made and the SPA942 uses its speaker. When the
> far end of a call hangs up , the SPA942 stays off hook, and after a time
> plays a fast busy. The user then has to press the line presence button to
> hang up the phone.
> I think I must be missing some sip.conf parameter. My sip.conf is pretty
> simple for these extensions; here is what I am using now:
> [extension1234]
> mailbox=1...@default
> type=friend
> context=users
> host=dynamic
> secret=verysecret
> I have looked at the sample sip.conf and did not get any clues, also the
> SPA900 Admin Manual doesn't say anything about it.
> Thanks
> Cassius
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>

Something that would help you GREATLY in determining what's going on
is to run a packet capture (such as with tshark or tcpdump) and read
the "conversation" between your device and Asterisk

e.g.: tshark -f 'port 5060' -w /tmp/test.pcap -S

Run the above command (if you have tshark, which is the terminal
version of Wireshark), perform a test call that duplicates this issue,
then press CTRL-C on the terminal, download /tmp/test.pcap, and open
it in Wireshark. Of course, you'll need to know how to read a SIP
conversation, but that's beyond the scope of this email

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Re: [asterisk-users] Asterisk 1.6 and Music on Hold

[Olivier CALVANO]

2010/11/24 Sherwood McGowan :
> On Wed, Nov 24, 2010 at 4:30 AM, Olivier CALVANO  wrote:
>> Hi
>>
>> i have a small problems on Asterisk 1.6 with the MusiconOld :
>>
>> musiconhold.conf:
>>
>> [Sound_1]
>> mode=quietmp3
>> directory=/var/lib/asterisk/moh/Sound_1
>>
>> in extensions.conf:
>>
>> exten => 0532xx,1,Answer
>> exten => 0532xx,2,MusicOnHold(Sound_1)
>> exten => 0532xx,3,Dial(SIP/ACCOUNT001,180,t)
>> exten => 0532xx,4,Hangup
>>
>>
>>
>>
>> When i call to the number, i have the Music "Sound_1" but the SIP Phone
>> don't ring ...
>>
>> Where is my error ?
>>
>>
>> and second question, can i said at asterisk that when he receive the call,
>> he play the music from first second ? and repeat at the end of the music.
>>
>> Thanks for your help
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>               http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> First, if you don't use the Music on hold command prior to the dial,
> do you hear ringing? It seems to me that what's going on here is that
> you're overriding the progress notification that results from the
> device responding to the invite with "TRYING" or "RINGING" by running
> MOH. If the ringing doesn't occur even when you remove the MOH
> command, your device is probably not signaling properly and you'll
> need to use the "r" option in your Dial command.



Hi

Thanks for your help, yes, if i don't put the music on hold command, the phone
ringing. I have change for put the "r" but no effect

bye
olivier

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[asterisk-users] TDM calls fall after some minutes

[luca capra]

Hello,

it's my first post on this list, I hope not to bore youwith my novice 
questions..


We're using a TDM400 with 3 fxo modules connected to pstn.
Call goes inbound/outbound correctly, after playing a bit on some 
dahdi-channels.conf/chan_dahdi.conf options.


The "big" problem is that after 5 minutes and 16'' of outbound call, 
suddenly the line fall down. Likely someone an hangup.

I'm looking the signalling method (actually fxs_ks) but with no luck.

Does somebody had similar problems ?

Thanks in advance
Luca
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[asterisk-users] SPA942 on speaker phone does not hang up?

[Cassius Smith]

Hello all,
I am using Linksys SPA942 in my current installation activity. I see a
peculiar behavior: A call is made and the SPA942 uses its speaker. When the
far end of a call hangs up , the SPA942 stays off hook, and after a time
plays a fast busy. The user then has to press the line presence button to
hang up the phone.

I think I must be missing some sip.conf parameter. My sip.conf is pretty
simple for these extensions; here is what I am using now:

[extension1234]
mailbox=1...@default
type=friend
context=users
host=dynamic
secret=verysecret

I have looked at the sample sip.conf and did not get any clues, also the
SPA900 Admin Manual doesn't say anything about it.

Thanks
Cassius


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[asterisk-users] [Asterisk-users] asterisk-1.8.0 compilation error

[RAJNIKANT VANZA]

Hi all,

I want to upgared from asterisk-1.6.2.6  version to asterisk-1.8.0 version.

When i execute "make" command for compilation i have seen below errors.

In file included from
/usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/cdr.h:31

/usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/data.h:233: error:
field ‘AST_DATA_IPADDR’ has incomplete type
/usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/data.h:794: error:
return type is an incomplete type
/usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/data.h: In function
‘ast_data_retrieve_ipaddr’:
/usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/data.h:799: warning:
‘return’ with a value, in function returning void
make[1]: *** [cdr_webservice.o] Error 1
make: *** [cdr] Error 2

Please, help me for resolve above errors.
Thanks in advance.
-- 
Best Regards,
Rajnikant Vanza
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Re: [asterisk-users] Asterisk 1.6 and Music on Hold

[Sherwood McGowan]

On Wed, Nov 24, 2010 at 4:30 AM, Olivier CALVANO  wrote:
> Hi
>
> i have a small problems on Asterisk 1.6 with the MusiconOld :
>
> musiconhold.conf:
>
> [Sound_1]
> mode=quietmp3
> directory=/var/lib/asterisk/moh/Sound_1
>
> in extensions.conf:
>
> exten => 0532xx,1,Answer
> exten => 0532xx,2,MusicOnHold(Sound_1)
> exten => 0532xx,3,Dial(SIP/ACCOUNT001,180,t)
> exten => 0532xx,4,Hangup
>
>
>
>
> When i call to the number, i have the Music "Sound_1" but the SIP Phone
> don't ring ...
>
> Where is my error ?
>
>
> and second question, can i said at asterisk that when he receive the call,
> he play the music from first second ? and repeat at the end of the music.
>
> Thanks for your help
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

First, if you don't use the Music on hold command prior to the dial,
do you hear ringing? It seems to me that what's going on here is that
you're overriding the progress notification that results from the
device responding to the invite with "TRYING" or "RINGING" by running
MOH. If the ringing doesn't occur even when you remove the MOH
command, your device is probably not signaling properly and you'll
need to use the "r" option in your Dial command.

Cheers

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Re: [asterisk-users] Contradiction in GROUP() function

[Steve Davies]

On 24 November 2010 10:12, Steve Davies  wrote:
> I am confused. In Asterisk 1.2 and 1.4, in the code there is an error:
> "Setting a group requires an argument (group name)"
>
> But the syntax is shown as: "Syntax: GROUP([category])"
>
> The [category] square brackets indicate to me an "optional" parameter,
> which contradicts the error.
>
> Verison 1.6 is non-committal in its definition, but I always assumed
> that an empty string was still a valid category-name, so GROUP()=123
> is as valid as GROUP(X)=123.
>
> Could this be clarified?

I suspect from further reading the code that this might just be a
misleading error message.

Regards
Steve

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[asterisk-users] Asterisk 1.6 and Music on Hold

[Olivier CALVANO]

Hi

i have a small problems on Asterisk 1.6 with the MusiconOld :

musiconhold.conf:

[Sound_1]
mode=quietmp3
directory=/var/lib/asterisk/moh/Sound_1

in extensions.conf:

exten => 0532xx,1,Answer
exten => 0532xx,2,MusicOnHold(Sound_1)
exten => 0532xx,3,Dial(SIP/ACCOUNT001,180,t)
exten => 0532xx,4,Hangup




When i call to the number, i have the Music "Sound_1" but the SIP Phone
don't ring ...

Where is my error ?


and second question, can i said at asterisk that when he receive the call,
he play the music from first second ? and repeat at the end of the music.

Thanks for your help

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[asterisk-users] Contradiction in GROUP() function

[Steve Davies]

I am confused. In Asterisk 1.2 and 1.4, in the code there is an error:
"Setting a group requires an argument (group name)"

But the syntax is shown as: "Syntax: GROUP([category])"

The [category] square brackets indicate to me an "optional" parameter,
which contradicts the error.

Verison 1.6 is non-committal in its definition, but I always assumed
that an empty string was still a valid category-name, so GROUP()=123
is as valid as GROUP(X)=123.

Could this be clarified?

Many thanks,
Steve

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Re: [asterisk-users] astcanary ?

[Jonas Kellens]

On 11/24/2010 10:28 AM, --[ UxBoD ]-- wrote:




Hello,

I notice that the following proces is running :

astcanary /var/run/asterisk/alt.asterisk.canary.tweet.tweet.tweet 1527


What is this ??


Kind regards,
Jonas.

You are running Asterisk with priority set. Check 
/etc/asterisk/asterisk.conf for the line highpriority = yes ; Run 
realtime priority (same as -p at startup)

--
Thanks, Phil


And what happens when I choose to run Asterisk without priority ? What 
does this priority do ?!



Kind regards,
Jonas.
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Re: [asterisk-users] action at registering or de-registering

[Sherwood McGowan]

On Wed, Nov 24, 2010 at 1:20 AM, Hans Witvliet  wrote:
> Hi all,
>
> Perhaps someone has dealt with it before.
>
> I want to activate a bunch of my own scripts after someone has registred
> om my asterisk, or when his cient has de-registerded.
>
> have been skimming through AGI and AMI, and seen a lot of nice features,
> but not the (de-)registering events.
>
> Kind regards, Hans
>
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>

I don't think this is currently possible, but I could be wrong. If it
turns out that this is not possible, maybe you should suggest it as a
feature, I could see where it could be useful

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Re: [asterisk-users] Asterisk 1.6 and Username in Dial

[Sherwood McGowan]

No you can't

On Wed, Nov 24, 2010 at 2:34 AM, Olivier CALVANO  wrote:
> Hi
>
> i don't see a answer at my question
>
> Bye
> Jerome
>
>
>
>
>
> 2010/11/9 Olivier CALVANO :
>> Hi
>>
>> In Asterisk 1.6/realtime Mysql, we can't put a username/password in a
>> Dial Command ?:
>>
>> 'Dial', 'SIP/Username:passw...@mypeer/${EXTEN},180,r'
>>
>> Thanks
>> Olivier
>>
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

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Re: [asterisk-users] astcanary ?

[--[ UxBoD ]--]

- Original Message -


Hello, 

I notice that the following proces is running : 

astcanary /var/run/asterisk/alt.asterisk.canary.tweet.tweet.tweet 1527 


What is this ?? 


Kind regards, 
Jonas. 

You are running Asterisk with priority set. Check /etc/asterisk/asterisk.conf 
for the line highpriority = yes ; Run realtime priority (same as -p at startup) 
-- 
Thanks, Phil 
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Re: [asterisk-users] usage of account code in CDR

[Mindaugas Kezys]

We use it to determine who is the caller.

Regards,
Mindaugas Kezys

Kolmisoft UAB 
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL: http://www.kolmisoft.com
Find us on Facebook

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nikhil
Sent: Wednesday, November 24, 2010 6:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] usage of account code in CDR

please reply on this if u know

On 11/18/2010 09:24 AM, Nikhil wrote:
> Hi everyone
> Anyone please explain me How Account code is use for billing., 
> Thanks Nikhil
>


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Re: [asterisk-users] Asterisk 1.6 and Username in Dial

[Olivier CALVANO]

Hi

i don't see a answer at my question

Bye
Jerome





2010/11/9 Olivier CALVANO :
> Hi
>
> In Asterisk 1.6/realtime Mysql, we can't put a username/password in a
> Dial Command ?:
>
> 'Dial', 'SIP/Username:passw...@mypeer/${EXTEN},180,r'
>
> Thanks
> Olivier
>

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Re: [asterisk-users] DAHDI phantom pickup when ringing

[Jonathan Hunter]

On 24 November 2010 01:20, Lyle Giese  wrote:

> Post the revelent portions of your extension.conf.  Maybe you have a logic
> error somewhere.
>
> Thanks Lyle.

My extensions.conf is fairly simple in this regard; I use macro-stdexten:

 [macro-stdexten];
exten => s,1,NoOp('${CALLERID(NAME)}' [${CALLERID(NUM)}] calling [${ARG1}])
exten => s,n,Set(MBOXCONTEXT=)
exten => s,n,Dial(${ARG1},30)   ; Ring the interface, 30
seconds maximum
exten => s,n,MailboxExists(${macro_ext...@${mboxcontext})
exten => s,n,NoOp(Got mailbox status of '${VMBOXEXISTSSTATUS}')
exten =>
s,n,GotoIf($["${VMBOXEXISTSSTATUS}"="SUCCESS"]?s-Voicemail,1:s-NOANSWER,1)

and it is called with
"SIP/&DAHDI/1r1&DAHDI/3r1&DAHDI/5r1&DAHDI/7r3&SIP/&SIP/&SIP/&SIP/&SIP/&DAHDI/2&DAHDI/4&DAHDI/6"

Have you tried to move the set from channel 5 to 8 and 7 to 9? (to see if
> one or two of the fxs channels have gone bad in the chan bank?)
>
> Good idea, thank you - I will try this tonight.


> It could also be a power supply issue inside the Zhone that tries to 'trip'
> the ringing.
>

Hmm - not sure how I might determine whether this is the case or not.. It
only seems to occur on some channels, at the moment.

Thanks,

Jonathan
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[asterisk-users] astcanary ?

[Jonas Kellens]

Hello,

I notice that the following proces is running :

astcanary /var/run/asterisk/alt.asterisk.canary.tweet.tweet.tweet 1527


What is this ??


Kind regards,
Jonas.
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