Re: [asterisk-users] allowguest defaults to yes for SIP

[Lee Howard]

Michael Wyres wrote:
> The way I see it, the reason you have encountered some resistance to your 
> opinion in regards to whether guest access should be allowed by default or 
> should not be, is not because your opinion is "right" or "wrong" - everyone 
> is entitled to an opinion - and your stance has merit, certainly - I don't 
> think anyone is actually disputing that.  It is more that a lot of the people 
> on this list have been using Asterisk for a LNG time, and have 
> explained why it might be advantageous to have guest access enabled by 
> default.  There are definitely uses for this functionality, as has been 
> demonstrated by a number of examples contained in this thread.
>   

I certainly understand why someone would want such a feature.  Again, I 
think that it's a feature that should not be enabled by default.  I 
realize that some people that are using this feature would be 
inconvenienced if this default were to change.  I think that 
inconvenience is far-outweighed by the benefits in avoiding exploitation 
who are unaware of this feature.

I don't know how long, exactly, a LNG time is.  Certainly there 
are plenty of people who have used it longer than I have.  I started 
investigating and studying Asterisk in 1999.  I started using it in 
2002.  If that's not long enough to deserve a voice, then I understand.

> Isn't this why you joined the list?  To learn more about the product, and get 
> ideas and assistance from the more experienced users of the product?
>   

I've been a list member for a very long time.  Back in that day I was 
accustomed to joining the users list for every software I used with any 
interest.  The point of joining the list was, yes, to learn, but also to 
share and to provide feedback to developers.

> You raised your concern, and Tilghman (a senior developer at Digium) 
> explained the reasoning behind the default setting.  He suggested that you 
> take your concern to the tracker and post a patch.  You resisted.

In case you weren't aware, I *DID* open a case on the bug tracker, and I 
*DID* write a patch as requested.  However, an eager bug marshal decided 
to close my case before the patch was written and asked me to come to 
this list to discuss the subject.  So Tilghman was asking me to create a 
*NEW* ticket and to post the patch there... yet all the while there were 
discussions going on on asterisk-dev about the very same subject which, 
as clearly stated, superseded my contribution due to merit.  In other 
words, there was little point for me to write any patch until after 
those whose opinions count due to merit are done (but even then, I still 
wrote and contributed a patch).

But please understand, I've been down this path before many times.  I 
wasn't trying to be resistant.  Instead, I was merely cognizant of the 
fact that I had already done enough to express my opinions and that to 
continue restating them over and over would have been futile and 
argumentative.

> Now, the default extensions.conf contains the following snippet:
>
> 
>
> [default]
> ;
> ; By default we include the demo.  In a production system, you
> ; probably don't want to have the demo there.
> ;
> include => demo
>
> 
>
> Now, a lot of people never RTFM for anything.  Moreover, how many people 
> actually read the EULA for any piece of software they use?  It's not 
> Asterisk/Digium's fault if people don't read the available documentation that 
> they provide.  The quite plainly clear statement above is "in a production 
> system, you probably don't want to have the demo there".  Did you read that 
> bit?  Did you wonder why that bit is there?

Yes, I did read that.  This led me to immediately remove the demo.  It 
did not, however, lead me to set allowguest=no.

> When I first started working with Asterisk, I clearly remember that line (or 
> something very similar) piquing my curiousity to dig a little deeper as to 
> why that statement was made.  Lo, I discovered that this was because by 
> default, guest access is allowed.
>   

You certainly took it further than I did.  I accepted what it said at 
face-value.  I didn't continue to investigate.  I can't help but think 
there are others like me who will not read between the lines to learn 
that guest access is enabled by default.  Indeed, the language in 
doc/security.txt doesn't currently make this clear, either... reading it 
at face value I see a bias against using the "default" context for 
anything involving tolls, but it still doesn't say that unauthenticated 
callers are permitted by default.  Again, you were more inquisitive than 
I was.  I applaud you for it.  Do we expect that level of 
inquisitiveness from all users?

> I too found the default access odd at first, but I chose to understand the 
> reasoning from people who knew better, instead of chucking a hissy fit.

I'm sorry, I'm not sure I understand your definition of hissy fit.  If 
you view my behavior as a hissy fit then I do apologize.  

Re: [asterisk-users] allowguest defaults to yes for SIP

[Tzafrir Cohen]

On Fri, Nov 13, 2009 at 12:38:22PM +1300, Matt Riddell wrote:
> On 13/11/09 12:33 PM, Tzafrir Cohen wrote:
> > On Fri, Nov 13, 2009 at 12:19:54PM +1300, Matt Riddell wrote:
> >
> >> Maybe the best way would be to make it that the default context only
> >> provides the info from the examples unless you provide an option:
> >>
> >> read_security_document=yes
> >
> > Asterisk used to require that you set have 'TELEPHONY=yes' in
> > /etc/{sysconfig,default}/asterisk to start running. This is no longer
> > the case. Such requirements are not the thing that will make the user
> > read the documentation, and they get in the way of automating the
> > installation.
> 
> Yeah, but would you automate an install with additional contents in the 
> default context?

The init.d script is not installed automatically on 'make install' . At
least not if you have an existing one. Likewise is 
/etc/{default,sysconfig}/asterisk .

Are you one of the selected few who actually read licenses before
pressing on 'I agree'? If that definition is what it takes to get
Asterisk working, it will become yet another confirmation button.

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

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Re: [asterisk-users] Health IVR Recordings

[Alex Balashov]

Nazir Ahmed Vaid wrote:

> We are looking for Pre-Recorded IVRs for Health Services in English and 
> other languages. If anyone is aware of a source kindly advise. We are 
> launching a TRIAGE SERVICE and we need these Recorded IVRs for this purpose.

What makes you think that generic recordings of medical terminology 
(or whatever is meant by "Health Services") are going to work?

Triage and emergency room intake is a rather specific sub-domain of 
the medical lexicon in both its clinical and administrative dimensions.

These are unlikely to exist unless someone has already built your 
intended product, and rather similarly at that.

-- 
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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Re: [asterisk-users] Multimedia PBX Solution

[Alex Balashov]

Nazir Ahmed Vaid wrote:

> We are planning to develop a Multimedia PABX to connect about 500 or 
> more personnel for Voice, Video and Text 
> Communication. www.*gvsc*net.net  is a similar solution 
> but we wish to have our own independent solution. Please advise if 
> anyone can offer a ready to go end to end Asterisk based solution.

1) If someone else is offering a "ready-to-go," "end-to-end" product 
and you purchase it, would that not conflict with your goals of having 
your "own independent solution?"

2) A bit of marketing and communication advice, which you can take or 
leave:  Reduce the frequency with which the word "solution" appears in 
your sentences by about ... 5000%.

-- Alex

-- 
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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[asterisk-users] Multimedia PBX Solution

[Nazir Ahmed Vaid]

We are planning to develop a Multimedia PABX to connect about 500 or more
personnel for Voice, Video and Text Communication. www.*gvsc*net.net is a
similar solution but we wish to have our own independent solution. Please
advise if anyone can offer a ready to go end to end Asterisk based solution.

-- 
السلام عليكم ورحمة الله وبركاته


Nazir Ahmed Vaid
Cell:+92300-828

eHealth Services (Pvt) Ltd.
http://www.ehealth-services.com

NexSource Pakistan (Pvt) Ltd.

ASK Development
http://www.askdevelopment.org
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[asterisk-users] Health IVR Recordings

[Nazir Ahmed Vaid]

We are looking for Pre-Recorded IVRs for Health Services in English and
other languages. If anyone is aware of a source kindly advise. We are
launching a TRIAGE SERVICE and we need these Recorded IVRs for this purpose.


-- 
السلام عليكم ورحمة الله وبركاته


Nazir Ahmed Vaid
Cell:+92300-828

eHealth Services (Pvt) Ltd.
http://www.ehealth-services.com

NexSource Pakistan (Pvt) Ltd.

ASK Development
http://www.askdevelopment.org
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Re: [asterisk-users] allowguest defaults to yes for SIP

[Michiel van Baak]

On 12:38, Fri 13 Nov 09, Matt Riddell wrote:
> On 13/11/09 12:33 PM, Tzafrir Cohen wrote:
> > On Fri, Nov 13, 2009 at 12:19:54PM +1300, Matt Riddell wrote:
> >
> >> Maybe the best way would be to make it that the default context only
> >> provides the info from the examples unless you provide an option:
> >>
> >> read_security_document=yes
> >
> > Asterisk used to require that you set have 'TELEPHONY=yes' in
> > /etc/{sysconfig,default}/asterisk to start running. This is no longer
> > the case. Such requirements are not the thing that will make the user
> > read the documentation, and they get in the way of automating the
> > installation.
> 
> Yeah, but would you automate an install with additional contents in the 
> default context?

We do. It's the only way to get ENUM running on new boxen ;)

and yes I know, I'm not the beginning user anymore.

-- 

Michiel van Baak
mich...@vanbaak.eu
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

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


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Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters

[Olivier]

2009/11/12 Tzafrir Cohen 

> On Thu, Nov 12, 2009 at 01:43:48PM +0100, Olivier wrote:
> > 2009/11/11 Tzafrir Cohen 
> >
> > > On Wed, Nov 11, 2009 at 08:24:53PM +0100, Olivier wrote:
>
> > > > What about adding per-span section headers like Asterisk .conf files
> ?
> > > > [span1]
> > > > group_lines 1
> > > > pri_termtype
> > > > SPAN/1  TE
> > > > SPAN/2  TE
> > > >
> > > > [span2]
> > > > group_lines 2
> > > > pri_termtype
> > > > SPAN/2  TE
> > >
> > > This implies you will know span numbers in advance. I would like better
> > > ways to specify configuration.
> > >
> >
> > Really ?
> > I used this [span1] header as an example. Using any other string would be
> > fine for me as what matters, if I'm not mistaken, is the group_lines
> number
> > :
> >
> > [foo]
> > group_lines 1
> > pri_termtype
> >  SPAN/1  TE
> >  SPAN/2  TE
> >
> > [bar]
> > group_lines 2
>
> How can you tell which spans / channels will use each section?
>

My understanding of Dahdi is that I mostly need a group number to use with
Dial application :
Dial(DAHDI/g1/0123456789).





To get that dahdi-channels.conf file generated with dahdi_genconf, the only
missing feature (if my understanding is correct) is to be able to group
together a couple of ports so that I could either include in my diaplans,
lines such as Dial(DAHDI/g1/0123456789) or Dial(DAHDI/g2/9876543210).

So with a /etc/dahdi/genconf_parameters like this ...

[foo]
group_lines 1
pri_termtype
  SPAN/1TE


[bar]
group_lines 2
pri_termtype
  SPAN/2TE


... I think we've got everything needed to generate a
/etc/asterisk/dahdi-channels.conf file this :

; Span 1: B4/0/1 "B4XXP (PCI) Card 0 Span 1" (MASTER) AMI/CCS
group=1,11
context=remote
switchtype = euroisdn
signalling = bri_cpe_ptmp
channel => 1-2
context = default
group = 63

; Span 2: B4/0/2 "B4XXP (PCI) Card 0 Span 2" AMI/CCS
group=2,12
context=remote
switchtype = euroisdn
signalling = bri_cpe_ptmp
channel => 4-5
context = default
group = 63


So I don't understand where I would have to "tell which spans / channels
will use each section". The only purpose of sections within
genconf_parameters is to set the scope of parameters like group_lines.

Am I correct to think I can't today generate
/etc/asterisk/dahdi-channels.conf files in which 2 groups of BRI ports are
defined ?


>
> --
>   Tzafrir Cohen
> icq#16849755  
> jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
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Re: [asterisk-users] Will Digium iaxy stop working with asterisk 1.6; as it is discontinued?

[Tilghman Lesher]

On Thursday 12 November 2009 21:18:18 Joseph wrote:
> Digium has discontinued their ATA iaxy adapter; don't blame them, too
> expensive so they can not compete.
>
> The adapter is upgraded automaticaly when it is connected to new asterisk
> version; since this adapter is discontinued will it still work with
> asterisk 1.6 and beyond or will it be\ just a "door stopper"?

There is no reason why it should not continue to work.  However, there are
certain features that the IAXy will never have.  One particular item is that
you'll need to turn off calltoken support for peers/users which specify an
IAXy, as the firmware will never be modified to support that.  In addition,
the IAXy will never support any codec in the extended space (it probably
doesn't have enough CPU to master other codecs anyway).

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] solution for NAT issues?

[Ron]

i have also tried setting qualify='yes' but cpu usage spiked to 100%.

Ron wrote:
> Hi All,
> 
> 
> I been having issues on my users behind NAT, even if i hard set a 
> specific port on the phone, there are some network that NAT's it out to 
> a different port, in turn, some time later the phone could not be 
> reached by the server. i think because on the server, e.g. the user is 
> still registered on port 49923 but when the request is sent to that port 
>   the NAT router does not forward port 49923 to port of the IP phone, 
> maybe nat mapping has expired or something.
> 
> I have tried STUN, still sometimes the phones just cannot be reached.
> is there any other software to manage binding of ports on specific users 
> so that the routers always keeps the port mapped to port of the ip phone .
> TIA
> 
> Regards,
> Ron
> 
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[asterisk-users] Will Digium iaxy stop working with asterisk 1.6; as it is discontinued?

[Joseph]

Digium has discontinued their ATA iaxy adapter; don't blame them, too expensive 
so they can not compete.

The adapter is upgraded automaticaly when it is connected to new asterisk 
version; since this adapter is discontinued will it still work with asterisk 
1.6 
and beyond or will it be\ just a "door stopper"?

-- 
Joseph

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Re: [asterisk-users] RTPAUDIOQOS

[covici]

OK, thanks -- will have to try and see what I get.

Darryl Dunkin  wrote:

> I add this line in our in/out contexts:
> exten => h,1,Noop(QOS=${RTPAUDIOQOS})
> 
> Then grep for 'QOS' in asterisk-verbose (assuming you have verbose logging 
> on). I'm sure you could output it anwhere else as well with a system 
> call/echo.
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of 
> cov...@ccs.covici.com
> Sent: Thursday, November 12, 2009 06:19
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] RTPAUDIOQOS
> 
> OK, how do you get such information -- at times it would be very useful
> to know.
> 
> Darryl Dunkin  wrote:
> 
> > Sorry to reply so late, I am months behind and catching up.
> > 
> >  
> > 
> > I have been inspecting this on my own systems, and the results are 
> > inconsistent to say the least. I’ve been dumping these to the verbose logs 
> > for some time and monitoring them, but I have not been able to determine 
> > why the numbers are so far off. I am more concerned with the packets lost 
> > due to priority queuing within our network.
> > 
> >  
> > 
> > Here is an example just today:
> > 
> > ssrc=583450581
> > 
> > themssrc=1093951555
> > 
> > lp=0
> > 
> > rxjitter=0.003219
> > 
> > rxcount=1100
> > 
> > txjitter=0.000275
> > 
> > txcount=1108
> > 
> > rlp=57702
> > 
> > rtt=0.036000
> > 
> >  
> > 
> > If the txcount is only 1108, how can the remote lost packet count be 57702? 
> > Unless the call was nearly inaudible?
> > 
> >  
> > 
> > I did verify with this end user, and the call was just fine. Is this an 
> > issue with the phone at the remote end misreporting?
> > 
> >  
> > 
> > From: asterisk-users-boun...@lists.digium.com 
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mindaugas 
> > Kezys
> > Sent: Tuesday, September 22, 2009 01:01
> > To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > Subject: Re: [asterisk-users] RTPAUDIOQOS
> > 
> >  
> > 
> > Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified
> > 
> >  
> > 
> > Regards,
> > 
> > Mindaugas Kezys
> > 
> > http://www.kolmisoft.com
> > 
> > VoIP Billing and Routing Solutions
> > 
> >  
> > 
> > From: asterisk-users-boun...@lists.digium.com 
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL 
> > INDRODIYA
> > Sent: 2009 m. rugsėjo 22 d. 09:28
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [asterisk-users] RTPAUDIOQOS
> > 
> >  
> > 
> > hey all,
> > 
> > can any body know what this parameter stands for 
> > 
> > i got RTPAUDIOQOS while i have SIP channels 
> > 
> > but could not understand then what this parameter tell
> > 
> > ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000
> > 
> > if any one know plese help me to or give any documentation link
> > 
> > regards
> > Dhaval
> > 
> > 
> > 
> > Alternatives:
> > 
> > 
> > ___
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> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>  John Covici
>  cov...@ccs.covici.com
> 
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How do
you spend it?

 John Covici
 cov...@ccs.covici.com

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Re: [asterisk-users] Need opinion about GSM codec for Internet

[Martin]

If you doesn't need transcoding, you doesn't need any licenses...
Martin

- Original Message - 
From: "Vinícius Fontes" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Friday, November 06, 2009 11:43 AM
Subject: Re: [asterisk-users] Need opinion about GSM codec for Internet


In my opinion, GSM sounds great but not as good as G.729. So if you can't 
afford 
getting G.729, GSM is the way to go.



Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP



- "Alejandro Cabrera Obed"  escreveu:

> Dear all, I have implemented an Asterisk SIP server for a WAN VPN over
> Internet. We have users distributed along all my country (Argentina)
> that register to my Asterisk in order to talk among them.
>
> I'll plan to have voice and voicemail with GSM codec, because we can't
> afford the payment for the G.729 licenses (it's an administrative
> problem of our company, not an echonomical problem). So in this way
> Asterisk won't care about codec traslations, this sounds good.
>
> What do you think about the use of GSM codec for Internet calls ??? Do
> you think GSM is the best narrow-band codec if I can't use G.729 ???
>
> Thank you !!!
>
> Alejandro
>
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Re: [asterisk-users] Request for Review: Building Queues with Asterisk

[Leif Madsen]

Barry L. Kline wrote:
> Leif Madsen wrote:
> 
>> Please review and let me know how it goes for you!
> 
> Where is it?

Ah yes, in my eagerness to get ready for dinner with the g/fs parents, I have 
forgotten to post where this exists :)

I posted it to the issue tracker here:

https://issues.asterisk.org/view.php?id=16237

Enjoy!
Leif.

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Re: [asterisk-users] RTPAUDIOQOS

[Darryl Dunkin]

I add this line in our in/out contexts:
exten => h,1,Noop(QOS=${RTPAUDIOQOS})

Then grep for 'QOS' in asterisk-verbose (assuming you have verbose logging on). 
I'm sure you could output it anwhere else as well with a system call/echo.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of 
cov...@ccs.covici.com
Sent: Thursday, November 12, 2009 06:19
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RTPAUDIOQOS

OK, how do you get such information -- at times it would be very useful
to know.

Darryl Dunkin  wrote:

> Sorry to reply so late, I am months behind and catching up.
> 
>  
> 
> I have been inspecting this on my own systems, and the results are 
> inconsistent to say the least. I’ve been dumping these to the verbose logs 
> for some time and monitoring them, but I have not been able to determine why 
> the numbers are so far off. I am more concerned with the packets lost due to 
> priority queuing within our network.
> 
>  
> 
> Here is an example just today:
> 
> ssrc=583450581
> 
> themssrc=1093951555
> 
> lp=0
> 
> rxjitter=0.003219
> 
> rxcount=1100
> 
> txjitter=0.000275
> 
> txcount=1108
> 
> rlp=57702
> 
> rtt=0.036000
> 
>  
> 
> If the txcount is only 1108, how can the remote lost packet count be 57702? 
> Unless the call was nearly inaudible?
> 
>  
> 
> I did verify with this end user, and the call was just fine. Is this an issue 
> with the phone at the remote end misreporting?
> 
>  
> 
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mindaugas Kezys
> Sent: Tuesday, September 22, 2009 01:01
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] RTPAUDIOQOS
> 
>  
> 
> Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified
> 
>  
> 
> Regards,
> 
> Mindaugas Kezys
> 
> http://www.kolmisoft.com
> 
> VoIP Billing and Routing Solutions
> 
>  
> 
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA
> Sent: 2009 m. rugsėjo 22 d. 09:28
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] RTPAUDIOQOS
> 
>  
> 
> hey all,
> 
> can any body know what this parameter stands for 
> 
> i got RTPAUDIOQOS while i have SIP channels 
> 
> but could not understand then what this parameter tell
> 
> ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000
> 
> if any one know plese help me to or give any documentation link
> 
> regards
> Dhaval
> 
> 
> 
> Alternatives:
> 
> 
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Re: [asterisk-users] Termination Question

[B.Masoud @ SH]

That could work, but I have no control over server B, not server C !

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife
Sent: Friday, November 13, 2009 3:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Termination Question

 

I have no first-hand experience with the fussy idiosyncrasies, but the BIG
PICTURE is to have server A set up the call, and then "reinvite" the media
directly from B to C.  The call control messages flow to server A, the media
goes directly.   If you don't have "NAT traversal Kung-Fu", I suggest using
IAX2 over SIP.  

-K

 

 

 

- Original Message - 

From: B.Masoud @ SH   

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

Sent: Thursday, November 12, 2009 6:10 PM

Subject: Re: [asterisk-users] Termination Question

 

So how can I let A makes a PEER connection between B & C, and ONLY log the
call information?

 

Thanks.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife
Sent: Thursday, November 12, 2009 6:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Termination Question

 

...and with a packet switched transport layer, the 'hairpin' route through A
may create problematic levels of latency--latency that would perhaps NOT
have been problematic on a classic circuit switched route, so it's
definitely advisable to nail up a connection between b and c.

 

-K

 

 

- Original Message - 

From: Tarek Sawah   

To: Asterisk Users   

Sent: Thursday, November 12, 2009 8:28 AM

Subject: Re: [asterisk-users] Termination Question

 

for the sake of bandwidth you are supposed to connect each two servers
together.. otherwise calls between B && C will have to go through A .

-- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria:
+963 944 618286 USA: +1 347 562 2308 




  _  


From: i...@saudihome.com
To: asterisk-users@lists.digium.com
Date: Thu, 12 Nov 2009 16:13:10 +0300
Subject: [asterisk-users] Termination Question

Hello,

I would like to know how the following scenario works:

 

I have 3 Asterisk servers, A,B & C,  each one is located in a different
country.

Asterisk A is the main one, and both B & C are connected to it.

 

My question is, when a call is originated from B to C, it will have to go
through A, but does A makes a peer connection between B & C to eliminate
bandwidth and latency, or the call has to go through A ???

 

Thanks.

 

 


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Re: [asterisk-users] Termination Question

[Karl Fife]

I have no first-hand experience with the fussy idiosyncrasies, but the BIG 
PICTURE is to have server A set up the call, and then "reinvite" the media 
directly from B to C.  The call control messages flow to server A, the media 
goes directly.   If you don't have "NAT traversal Kung-Fu", I suggest using 
IAX2 over SIP.  
-K



- Original Message - 
  From: B.Masoud @ SH 
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
  Sent: Thursday, November 12, 2009 6:10 PM
  Subject: Re: [asterisk-users] Termination Question


  So how can I let A makes a PEER connection between B & C, and ONLY log the 
call information?

   

  Thanks.

   

  From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife
  Sent: Thursday, November 12, 2009 6:10 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Termination Question

   

  ...and with a packet switched transport layer, the 'hairpin' route through A 
may create problematic levels of latency--latency that would perhaps NOT have 
been problematic on a classic circuit switched route, so it's definitely 
advisable to nail up a connection between b and c.

   

  -K

   

   

  - Original Message - 

From: Tarek Sawah 

To: Asterisk Users 

Sent: Thursday, November 12, 2009 8:28 AM

Subject: Re: [asterisk-users] Termination Question

 

for the sake of bandwidth you are supposed to connect each two servers 
together.. otherwise calls between B && C will have to go through A .

-- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: 
+963 944 618286 USA: +1 347 562 2308 







From: i...@saudihome.com
To: asterisk-users@lists.digium.com
Date: Thu, 12 Nov 2009 16:13:10 +0300
Subject: [asterisk-users] Termination Question

Hello,

I would like to know how the following scenario works:

 

I have 3 Asterisk servers, A,B & C,  each one is located in a different 
country.

Asterisk A is the main one, and both B & C are connected to it.

 

My question is, when a call is originated from B to C, it will have to go 
through A, but does A makes a peer connection between B & C to eliminate 
bandwidth and latency, or the call has to go through A ???

 

Thanks.

 

 




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Re: [asterisk-users] allowguest defaults to yes for SIP

[Michael Wyres]

>-Original Message-
>From: asterisk-users-boun...@lists.digium.com 
>[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Howard
>Sent: Friday, 13 November 2009 06:16
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [asterisk-users] allowguest defaults to yes for SIP

>I could be wrong, but I don't generally consider myself stupid or 
>lazy... and yet this default setting as "yes" took me by surprise, 
>obviously.

This has nothing to do with "stupidity" or "laziness".

The way I see it, the reason you have encountered some resistance to your 
opinion in regards to whether guest access should be allowed by default or 
should not be, is not because your opinion is "right" or "wrong" - everyone is 
entitled to an opinion - and your stance has merit, certainly - I don't think 
anyone is actually disputing that.  It is more that a lot of the people on this 
list have been using Asterisk for a LNG time, and have explained why it 
might be advantageous to have guest access enabled by default.  There are 
definitely uses for this functionality, as has been demonstrated by a number of 
examples contained in this thread.

Isn't this why you joined the list?  To learn more about the product, and get 
ideas and assistance from the more experienced users of the product?

You raised your concern, and Tilghman (a senior developer at Digium) explained 
the reasoning behind the default setting.  He suggested that you take your 
concern to the tracker and post a patch.  You resisted.  The open source 
community (despite what some think) is a highly organised community, with 
structures in place to get things like that done.

If you consistently did end runs around established corporate procedures in 
your workplace, you'd expect a foot up the ass from management.  Tilghman was 
as politely as possible asking you to follow the established procedures.  You 
chose to resist.

Now, the default extensions.conf contains the following snippet:



[default]
;
; By default we include the demo.  In a production system, you
; probably don't want to have the demo there.
;
include => demo



Now, a lot of people never RTFM for anything.  Moreover, how many people 
actually read the EULA for any piece of software they use?  It's not 
Asterisk/Digium's fault if people don't read the available documentation that 
they provide.  The quite plainly clear statement above is "in a production 
system, you probably don't want to have the demo there".  Did you read that 
bit?  Did you wonder why that bit is there?  When I first started working with 
Asterisk, I clearly remember that line (or something very similar) piquing my 
curiousity to dig a little deeper as to why that statement was made.  Lo, I 
discovered that this was because by default, guest access is allowed.  

Digium has made that available in the distribution for EVERYONE to read, and 
extensions.conf is probably the most accessed file in an Asterisk system not 
using RealTime, so people who choose to ignore reading the excellent notes and 
annotations in all of the default configuration files is doing themselves a 
disservice.

I too found the default access odd at first, but I chose to understand the 
reasoning from people who knew better, instead of chucking a hissy fit.

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Re: [asterisk-users] Need Adapter/Gateway with PSTN-interface

[Martin]

Grandstream HT503. For me works just fine. 1xFXO 1xFXS port. Each port has its 
own sip account.
Martin
  - Original Message - 
  From: jonas kellens 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Thursday, November 12, 2009 5:38 AM
  Subject: Re: [asterisk-users] Need Adapter/Gateway with PSTN-interface


  I've read (through google) that the Linksys SPA-products do not have good 
voice quality on the PSTN-line.

  Grandstream HT486 is also just lifeline and EOL.

  The only I come up with is Patton-gateways but these are not at all cheap !

  Jonas.

  On Thu, 2009-11-12 at 10:13 +, Steve Howes wrote: 
On 12 Nov 2009, at 09:33, jonas kellens wrote:

> I am looking for a gateway/ATA that can take conversations on the  
> analogue line (PSTN) and send them to the Asterisk server on the  
> private network.
>
> I was experimenting with the Atcom AG-188N but the "FXO"-port only  
> supports lifeline, so it's not a real FXO-port that can send  
> incoming calls to my private Asterisk-server.
>
> Could someone advice on a gateway that can take analogue calls and  
> transfer them on my local network ?!
>
> I know about the Digium-cards. Are there alternatives ?

Google could tell you this Try the Linksys/Sipura type products

S




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Re: [asterisk-users] Termination Question

[B.Masoud @ SH]

So how can I let A makes a PEER connection between B & C, and ONLY log the
call information?

 

Thanks.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife
Sent: Thursday, November 12, 2009 6:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Termination Question

 

...and with a packet switched transport layer, the 'hairpin' route through A
may create problematic levels of latency--latency that would perhaps NOT
have been problematic on a classic circuit switched route, so it's
definitely advisable to nail up a connection between b and c.

 

-K

 

 

- Original Message - 

From: Tarek Sawah   

To: Asterisk Users   

Sent: Thursday, November 12, 2009 8:28 AM

Subject: Re: [asterisk-users] Termination Question

 

for the sake of bandwidth you are supposed to connect each two servers
together.. otherwise calls between B && C will have to go through A .

-- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria:
+963 944 618286 USA: +1 347 562 2308 





  _  


From: i...@saudihome.com
To: asterisk-users@lists.digium.com
Date: Thu, 12 Nov 2009 16:13:10 +0300
Subject: [asterisk-users] Termination Question

Hello,

I would like to know how the following scenario works:

 

I have 3 Asterisk servers, A,B & C,  each one is located in a different
country.

Asterisk A is the main one, and both B & C are connected to it.

 

My question is, when a call is originated from B to C, it will have to go
through A, but does A makes a peer connection between B & C to eliminate
bandwidth and latency, or the call has to go through A ???

 

Thanks.

 

 


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[asterisk-users] TDM400p , asteriskNow and may other woes.....

[Humanx2000]

Hello all,

I am new to asterisk and have spent a good 4 or 5 days trying to get
things sorted out. I initially installed it in Fedora Core 11 and
compiled mods + asterisk. After much problems, I went with asteriskNow.
The biggest problem I am have is getting some kind of base configuration
going. I have been all over Google, but what I oftwen find is
conflicting or outdated information. Commands to use that no longer work
because things have changed from zaptel to dahdi.

I have a TDM400P with 1 FXO module and 1 FXS module installed. The card
is readily seen. I cannot get a dial tone (and I did plug in the power).
And am unsure what to do.

1. /etc/init.d/asterisk does not exist, so I have no idea how the system
is even starting.
2. The /etc/asterisk folder has zapata.conf.template AND
chan_dahdi.conf.template. Which one am I supposed to use?

Anyone want to take some pity? Just looking to get to the point of a
dial tone. At least will know things are working. I can go on from
there. But at this point I am stuck. Not trying to take the lazy way
out, just trying to get a handle on this. BELIEVE ME I have put forth a
GREAT deal of effort. Went to the irc channel and though there were some
200 users, most were prob just idleing. Been all over the forums. And
every google "setup tdm400p asterisk" page that exists.

Just want to plug in regular telephone, and dial out through my
telephone company.

Thanks


dmesg

Nov  8 00:01:37 localhost kernel: Module 0: Installed -- AUTO FXS/DPO
Nov  8 00:01:37 localhost kernel: Module 1: Not installed
Nov  8 00:01:37 localhost kernel: Module 2: Not installed
Nov  8 00:01:37 localhost kernel: Module 3: Installed -- AUTO FXO (FCC mode)
Nov  8 00:01:37 localhost kernel: Found a Wildcard TDM: Wildcard TDM400P
REV E/F (2 modules)
Nov  8 00:01:37 localhost kernel: dahdi_transcode: Loaded.


--
dahdi_cfg -vv
--
DAHDI Tools Version - 2.2.0

DAHDI Version: 2.2.0.2
Echo Canceller(s): MG2
Configuration
==


Channel map:

Channel 01: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04)

2 channels to configure.

Setting echocan for channel 1 to mg2
Setting echocan for channel 4 to mg2


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[asterisk-users] Home line noise problem

[robert boardman]

I Have a home line connected to a tdm400p with 3 extensions and a siemens
sip-dect , it seems to work fine but during a call there is always a digital
squeal every so often does anyone know what this could be?

Robb
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Re: [asterisk-users] Can't connect to voip provider over NAT

[Landy Landy]

> Have you tried "nat=yes" in the
> definition in sip.conf?

Yes, I have that definition in sip.conf. Now, I'm getting the following error   

-- SIP/voipprovider-094132d8 is making progress passing it to SIP/102-09423d58
-- Got SIP response 603 "Declined" back from 208.xx.xx.xx
-- SIP/voipprovider-094132d8 is busy
  == Everyone is busy/congested at this time (1:1/0/0)

and I get a "This account number is not valid on the headset".

I've called my provider and they've said that everything is fine at their end. 
I don't know why I'm getting the message saying the account is not valid.



  

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Re: [asterisk-users] allowguest defaults to yes for SIP

[Matt Riddell]

On 13/11/09 12:33 PM, Tzafrir Cohen wrote:
> On Fri, Nov 13, 2009 at 12:19:54PM +1300, Matt Riddell wrote:
>
>> Maybe the best way would be to make it that the default context only
>> provides the info from the examples unless you provide an option:
>>
>> read_security_document=yes
>
> Asterisk used to require that you set have 'TELEPHONY=yes' in
> /etc/{sysconfig,default}/asterisk to start running. This is no longer
> the case. Such requirements are not the thing that will make the user
> read the documentation, and they get in the way of automating the
> installation.

Yeah, but would you automate an install with additional contents in the 
default context?

-- 
Cheers,

Matt Riddell
Director
___

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http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
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Re: [asterisk-users] allowguest defaults to yes for SIP

[Tzafrir Cohen]

On Fri, Nov 13, 2009 at 12:19:54PM +1300, Matt Riddell wrote:

> Maybe the best way would be to make it that the default context only 
> provides the info from the examples unless you provide an option:
> 
> read_security_document=yes

Asterisk used to require that you set have 'TELEPHONY=yes' in
/etc/{sysconfig,default}/asterisk to start running. This is no longer
the case. Such requirements are not the thing that will make the user
read the documentation, and they get in the way of automating the
installation.

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

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Re: [asterisk-users] allowguest defaults to yes for SIP

[Matt Riddell]

On 13/11/09 9:37 AM, Lee Howard wrote:
> Michiel van Baak wrote:
>> When I started working with asterisk, and found my first issue, I
>> created a patch, put it on the tracker, followed up on the comments, and
>> stuff got in.
>
> I'm sincerely pleased to know that you've had a different experience
> than have I.

I've had an experience which is a little of both.

I've had some patches accepted, and other not accepted (MySQL userfield2-5).

I think it's really important that not every patch gets accepted, and I 
really like the discussion which has taken place on this one.

Basically the two sides of the argument are:

For: I put stuff in my default context, now people can use it without 
authentication - I didn't expect this.

Against: I'm a new user, I tried to get Asterisk working but had 
authentication problems, now I'm moving to Microsoft OCS (or 3cx or 
whatever).

I kinda think that you want to make it as easy as possible for new users 
to at least run an echo test (and maybe make a call through to Digium).

Once they've done that they're going to need to edit config files.

If there is strong wording in the config files explaining that they 
shouldn't be adding anything here without first reading the security 
document I think it would suffice.

Maybe the best way would be to make it that the default context only 
provides the info from the examples unless you provide an option:

read_security_document=yes

or whatever.

I know that it seems really easy for most of us to chuck a couple of sip 
devices into the config and set up some extensions, but for a new user, 
any step at all they need to make before getting a "call" working is bad.

The average new user won't know much about VoIP, nor much (if anything) 
about Linux, and seeing some text interface provide some random error 
when they try it for the first time will just turn them away.

>> If you read the page about contributing code to asterisk, it clearly
>> states that the dev mailinglist is the place to discuss development.
>> If you post comments there, people will read it, comment on it, and if
>> more people agree with the ideas it will get implemented.
>>
>> It's how all OpenSource projects work.
>
> I truly wish it were.  I've seen more than a few that didn't.

:) just consider yourself lucky it's not glibc or something you're 
trying to commit to :)

The people with commit access tend to just say no.  Even if the change 
stops something from breaking on multiple platforms (see eglibc discussion).

Basically to get a change into Asterisk, you need a reasonably good 
percentage of people agreeing that the change is worthwhile (and the 
best way to implement it).

Don't get me wrong, I understand the change you're proposing, just that 
it may not be the 100% best way to do it, and it needs to be carefully 
thought out before proceeding with something which may have a large 
impact on new users.

Think what it's like for the 3G video people who have a huge patchset 
that they wrote before bringing it up for discussion only to hear it was 
the wrong way to do it.

At least the patch is small :D

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Matt Riddell
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Re: [asterisk-users] allowguest defaults to yes for SIP

[Matt Riddell]

On 13/11/09 8:30 AM, SIP wrote:
> Eh... if VoIP fraud weren't so rampant, and I didn't constantly see
> mailings to the Asterisk list about "How do I secure my system from the
> people who've been costing me tons of money lately," I would say that
> having a lax stance on security in exchange for additional usability
> might be a good thing.  But as is, that's simply not the case. The
> 'usability' you get from this is really only questionably essential in
> its ability to save time, but the security one would get from a change
> could save some people actual money -- not just time.

The problem there is normally lax usernames and passwords.  Not that 
there is default access to the echo test.

> As someone who used to design systems and networks, I would vote for
> security over nebulous desire to keep the status quo.

Because you're already using Asterisk.  If it had been too hard at the 
start maybe you wouldn't.

> True, you can't keep stupid people from doing stupid things, but given a
> choice between protecting the ignorant from a bad situation or catering
> to those who want to avoid an extra step or two on installation, I'd
> side with protecting the ignorant every time. There's always a trade-off
> between usability and security, and I'm of the opinion that security is
> the more important of the two when dealing with systems connected to the
> Internet. Call me a cynic. :)

The ignorant won't have changed the default context - they likely won't 
even know how to edit a config file - so they're safe.

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Re: [asterisk-users] Request for Review: Building Queues with Asterisk

[Michiel van Baak]

On 17:19, Thu 12 Nov 09, Leif Madsen wrote:
> I have been working on some documentation for how to build queues for 
> Asterisk. 
> This is an introduction for getting device state working for queues, and 
> building queues. It contains the documentation file (text format) and also 
> has 
> the .tar.gz file of the /etc/asterisk/ directory I was using for testing.
> 
> The modules.conf file has autoload=no enabled, and just loads the modules 
> that 
> were required for the example (along with probably a couple extra modules, 
> but 
> the list of modules has been toned down).
> 
> Please review and let me know how it goes for you!

Where can we find all of this ?
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Re: [asterisk-users] allowguest defaults to yes for SIP

[Matt Riddell]

On 13/11/09 3:59 AM, Danny Nicholas wrote:
> Without the allowguest=no, Asterisk doesn't put up any defense against an
> unauthorized guest.  You still have NAT/Firewall/IPTABLE "defenses", for
> what they are worth.  The trick is to get what you need without allowing
> what you don't want.

A slight clarification - I wouldn't say it's defences.

By default these calls are sent to the default context (which should not 
have the capability to make calls other than test the system).

So, yes you are allowing unauthenticated calls, but to the echo test etc.

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Re: [asterisk-users] Trouble registering Cisco 7942

[Warren Selby]

Just checked with my actual config file, and it's not a sanitation mistake,
that's how I've actually got mine setup.  Like I said earlier, I've never
even messed with that section of my config before...I set mine up based on a
combination of configs I've found around the net (I think you've already
linked to them in another post to the list).

Thanks,
--Warren Selby

On Thu, Nov 12, 2009 at 4:11 PM, Stephen Reese  wrote:

> On Wed, Nov 11, 2009 at 9:34 PM, Warren Selby 
> wrote:
> > The 7960 and 79x2 use different sip firmwares and as far a I have seen
> > the 7960 does not have the same port issue the 7941/2 seems to have
> > (which technically is not a problem, just an implementation of the sip
> > protocol that you don't typically see).
> >
> > As to your issue, are you still seeing the same error messages in the
> > ssh logs?  I haven't ever had to use the register with proxy settings
> > in my configs, but I've only worked with the 79x1 series phones, not
> > the x2.
> >
> > I've actually got a post up on my blog addressing setting up a 7941 in
> > a situation similar to yours:
> >
> > http://www.selbytech.com/2009/10/setup-cisco-7941-or-7961-with-asterisk/
> >
> > In that post is a sanitized version of my conf file that I use on my
> > own deskphone, if you'd like to download it and try it out with your
> > setup.
> >
>
> My config is very similar though my only question is you have
> registerWithProxy set to true though nothing defined. Was this a
> sanitation mistake?
>
> 
> 
> 
> 
> 
> 
> 
> true
> 
>
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Re: [asterisk-users] Request for Review: Building Queues with Asterisk

[Barry L. Kline]

Leif Madsen wrote:

> Please review and let me know how it goes for you!

Where is it?

Barry


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[asterisk-users] Request for Review: Building Queues with Asterisk

[Leif Madsen]

I have been working on some documentation for how to build queues for Asterisk. 
This is an introduction for getting device state working for queues, and 
building queues. It contains the documentation file (text format) and also has 
the .tar.gz file of the /etc/asterisk/ directory I was using for testing.

The modules.conf file has autoload=no enabled, and just loads the modules that 
were required for the example (along with probably a couple extra modules, but 
the list of modules has been toned down).

Please review and let me know how it goes for you!

Thanks!
Leif Madsen.

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Re: [asterisk-users] Trouble registering Cisco 7942

[Stephen Reese]

On Wed, Nov 11, 2009 at 9:34 PM, Warren Selby  wrote:
> The 7960 and 79x2 use different sip firmwares and as far a I have seen
> the 7960 does not have the same port issue the 7941/2 seems to have
> (which technically is not a problem, just an implementation of the sip
> protocol that you don't typically see).
>
> As to your issue, are you still seeing the same error messages in the
> ssh logs?  I haven't ever had to use the register with proxy settings
> in my configs, but I've only worked with the 79x1 series phones, not
> the x2.
>
> I've actually got a post up on my blog addressing setting up a 7941 in
> a situation similar to yours:
>
> http://www.selbytech.com/2009/10/setup-cisco-7941-or-7961-with-asterisk/
>
> In that post is a sanitized version of my conf file that I use on my
> own deskphone, if you'd like to download it and try it out with your
> setup.
>

My config is very similar though my only question is you have
registerWithProxy set to true though nothing defined. Was this a
sanitation mistake?








true


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Re: [asterisk-users] allowguest defaults to yes for SIP

[Lee Howard]

Michiel van Baak wrote:
> When I started working with asterisk, and found my first issue, I
> created a patch, put it on the tracker, followed up on the comments, and
> stuff got in.

I'm sincerely pleased to know that you've had a different experience 
than have I.

> If you read the page about contributing code to asterisk, it clearly
> states that the dev mailinglist is the place to discuss development.
> If you post comments there, people will read it, comment on it, and if
> more people agree with the ideas it will get implemented.
>
> It's how all OpenSource projects work.

I truly wish it were.  I've seen more than a few that didn't.

Thanks,

Lee.


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Re: [asterisk-users] allowguest defaults to yes for SIP

[Michiel van Baak]

On 11:16, Thu 12 Nov 09, Lee Howard wrote:
> Danny Nicholas wrote:
> > "Gentlemens clubs" usually don't have any.  While LH probably has a valid
> > point, jumping on Til isn't the way to bring it home.  You can't protect the
> > stupid or lazy from themselves.  If you can't do this right, pay someone
> > else to.
> 
> You're suggesting that if I pay someone they'll be able to get the 
> default setting for allowguest changed to "no" ?

No, he was saying that if you dont know the system you are going to
setup, and dont have the time/resources to read up on how it works, you
can always hire someone who knows how stuff works.

> 
> I could be wrong, but I don't generally consider myself stupid or 
> lazy... and yet this default setting as "yes" took me by surprise, 
> obviously.

No-one told you you are stupid or lazy.
It's just that this option only allows unwanted stuff if the
configuration is made to do that.

> 
> So either I am stupid or lazy or there is a risk here that can catch 
> even others off-guard.
> 
> I've been down this contribution road-path a half-dozen times before 
> with Asterisk.  So forgive me if I don't play it out to the final futile 
> note.
> 
> In ESR's CatB there's the idea where the maintainer encourages (and 
> wants) bug reporting, feedback, and other non-code forms of contribution 
> (as well as code contributions).  He refers to it as grooming 
> co-developers.  That's not how Asterisk development works... here you 
> can contribute if you're already in the meritocracy, but if you're not, 
> then you have more than a difficult time in trying to even contribute in 
> small non-monetary ways.

This is so untrue.
When I started working with asterisk, and found my first issue, I
created a patch, put it on the tracker, followed up on the comments, and
stuff got in. Sometimes it takes some time before the first review of
your patch is happening. This is mainly because the developers are
really busy, and only part of the developers is being paid to do this
stuff for asterisk, all the others are doing it in their free time.

If you read the page about contributing code to asterisk, it clearly
states that the dev mailinglist is the place to discuss development.
If you post comments there, people will read it, comment on it, and if
more people agree with the ideas it will get implemented.

It's how all OpenSource projects work.

-- 

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Re: [asterisk-users] Codec interface

[Tzafrir Cohen]

On Thu, Nov 12, 2009 at 09:22:41AM -0500, Bill Shaw wrote:
> Hi All,
> 
> I need to interface a codec-type device to Asterisk.  The device uses a 
> TI TLV320AIC1110 codec in 15 bit linear data mode with a 2.048 MHz clock 
> supplied by the device.  I am about to start on a custom hardware design 
> to interface this device to  the computer,  but thought I'd ask here 
> before I get started on it.  Does anyone know of a hardware interface 
> that is already being manufactured that can tie a codec-based device 
> into Asterisk?

There's codec_dahdi , that implements g729 and g723 through a specific
Digium card.

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Re: [asterisk-users] Need Adapter/Gateway with PSTN-interface

[Ira]

At 02:25 AM 11/12/2009, you wrote:
>FWIW, I've had a few recommendations for the Linksys SPA3000. However,
>I haven't tried this for myself yet since I'm still in the planning
>stage of replacing my current Asterisk machine. In my case, I
>currently have a full-size tower and I'm planning to move to a
>mini-itx machine that doesn't have a PCI slot for my TDM400 card.

I was able to assemble a MiniITX box with a laptop HD and Atom 330 
that had room for my TDM400, so it's possible if you want.  I've not 
seen one assembled that would work, but I got all the parts I needed 
from NewEgg.

Ira 


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Re: [asterisk-users] my kernel is dazed and confused

[Tzafrir Cohen]

On Thu, Nov 12, 2009 at 09:31:11AM -0500, Dr. Michael J. Chudobiak wrote:
> Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason 
> a0 on CPU 0.
> Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely 
> on the PCI bus.
> Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue

NMI - Non Maskable Interrupt. This is a rather generic error message.
Search a bit to see how to make some more sense of the messages
following it.

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Re: [asterisk-users] allowguest defaults to yes for SIP

[Danny Nicholas]

I did not mean to state or imply that you are lazy or stupid;  It's just
that some folks expect to spend 10 minutes reading a PDF, set up Asterisk
and all is well - That's not what Open Source is about.  If you want limited
or no risk, you have to pay the piper.  I'll bet there are thousands of
pieces of code that are great that don't get through the contribution
process.  You can't have any type of *cracy without crazy :)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Howard
Sent: Thursday, November 12, 2009 1:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] allowguest defaults to yes for SIP

Danny Nicholas wrote:
> "Gentlemens clubs" usually don't have any.  While LH probably has a valid
> point, jumping on Til isn't the way to bring it home.  You can't protect
the
> stupid or lazy from themselves.  If you can't do this right, pay someone
> else to.

You're suggesting that if I pay someone they'll be able to get the 
default setting for allowguest changed to "no" ?

I could be wrong, but I don't generally consider myself stupid or 
lazy... and yet this default setting as "yes" took me by surprise, 
obviously.

So either I am stupid or lazy or there is a risk here that can catch 
even others off-guard.

I've been down this contribution road-path a half-dozen times before 
with Asterisk.  So forgive me if I don't play it out to the final futile 
note.

In ESR's CatB there's the idea where the maintainer encourages (and 
wants) bug reporting, feedback, and other non-code forms of contribution 
(as well as code contributions).  He refers to it as grooming 
co-developers.  That's not how Asterisk development works... here you 
can contribute if you're already in the meritocracy, but if you're not, 
then you have more than a difficult time in trying to even contribute in 
small non-monetary ways.

So anyway, I've been down this road a half-dozen times already, and it 
ends up being futile, frustrating, and time-consuming.  I'm too busy 
today to be interested in playing.

Thanks,

Lee.

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Re: [asterisk-users] allowguest defaults to yes for SIP

[SIP]

Eh... if VoIP fraud weren't so rampant, and I didn't constantly see
mailings to the Asterisk list about "How do I secure my system from the
people who've been costing me tons of money lately," I would say that
having a lax stance on security in exchange for additional usability
might be a good thing.  But as is, that's simply not the case. The
'usability' you get from this is really only questionably essential in
its ability to save time, but the security one would get from a change
could save some people actual money -- not just time.

As someone who used to design systems and networks, I would vote for
security over nebulous desire to keep the status quo.

True, you can't keep stupid people from doing stupid things, but given a
choice between protecting the ignorant from a bad situation or catering
to those who want to avoid an extra step or two on installation, I'd
side with protecting the ignorant every time. There's always a trade-off
between usability and security, and I'm of the opinion that security is
the more important of the two when dealing with systems connected to the
Internet. Call me a cynic. :)

N.


Danny Nicholas wrote:
> "Gentlemens clubs" usually don't have any.  While LH probably has a valid
> point, jumping on Til isn't the way to bring it home.  You can't protect the
> stupid or lazy from themselves.  If you can't do this right, pay someone
> else to.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Howard
> Sent: Thursday, November 12, 2009 12:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] allowguest defaults to yes for SIP
>
> Tilghman Lesher wrote:
>   
>> The issue in question was suspended, while the reporter makes the case on
>> 
> the
>   
>> Asterisk-dev mailing list, which is not this list.  The opinions there
>> 
> amongst 
>   
>> contributors (meritocracy, not democracy) are that keeping the sample
>> configuration as it is now is probably the way to go.
>>   
>> 
>
> Sigh... of course.  It's a gentlemen's club and only members have a say.
>
>   
>> If you want to create a new issue and attach your patch there, I'll look
>> 
> at
>   
>> it.
>> 
>
> I sent a patch.  I pointed you at a case.  That should have been FAR 
> more than enough for my attempt at contribution to be acceptable.
>
> Thanks,
>
> Lee.
>
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Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters

[Tzafrir Cohen]

On Thu, Nov 12, 2009 at 01:43:48PM +0100, Olivier wrote:
> 2009/11/11 Tzafrir Cohen 
> 
> > On Wed, Nov 11, 2009 at 08:24:53PM +0100, Olivier wrote:

> > > What about adding per-span section headers like Asterisk .conf files ?
> > > [span1]
> > > group_lines 1
> > > pri_termtype
> > > SPAN/1  TE
> > > SPAN/2  TE
> > >
> > > [span2]
> > > group_lines 2
> > > pri_termtype
> > > SPAN/2  TE
> >
> > This implies you will know span numbers in advance. I would like better
> > ways to specify configuration.
> >
> 
> Really ?
> I used this [span1] header as an example. Using any other string would be
> fine for me as what matters, if I'm not mistaken, is the group_lines number
> :
> 
> [foo]
> group_lines 1
> pri_termtype
>  SPAN/1  TE
>  SPAN/2  TE
> 
> [bar]
> group_lines 2

How can you tell which spans / channels will use each section?

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Re: [asterisk-users] allowguest defaults to yes for SIP

[Lee Howard]

Danny Nicholas wrote:
> "Gentlemens clubs" usually don't have any.  While LH probably has a valid
> point, jumping on Til isn't the way to bring it home.  You can't protect the
> stupid or lazy from themselves.  If you can't do this right, pay someone
> else to.

You're suggesting that if I pay someone they'll be able to get the 
default setting for allowguest changed to "no" ?

I could be wrong, but I don't generally consider myself stupid or 
lazy... and yet this default setting as "yes" took me by surprise, 
obviously.

So either I am stupid or lazy or there is a risk here that can catch 
even others off-guard.

I've been down this contribution road-path a half-dozen times before 
with Asterisk.  So forgive me if I don't play it out to the final futile 
note.

In ESR's CatB there's the idea where the maintainer encourages (and 
wants) bug reporting, feedback, and other non-code forms of contribution 
(as well as code contributions).  He refers to it as grooming 
co-developers.  That's not how Asterisk development works... here you 
can contribute if you're already in the meritocracy, but if you're not, 
then you have more than a difficult time in trying to even contribute in 
small non-monetary ways.

So anyway, I've been down this road a half-dozen times already, and it 
ends up being futile, frustrating, and time-consuming.  I'm too busy 
today to be interested in playing.

Thanks,

Lee.

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Re: [asterisk-users] "POTS 4K linear codec"

[Jeff LaCoursiere]

On Thu, 12 Nov 2009, Cary Fitch wrote:

> I am not sure what the problems are and the reasons for the basic 64K modems
> used in VOIP are.  I understand the compressed codecs that get the bandwidth
> down to 20-30 K.  And perhaps the 64K units give much better potential audio
> than you would get on a normal POTS line.
>
> But, as phone circuits VOIP/SIP doesn't seem to perform as well as plane old
> phones.
>
> Multiple transcodings cause issues.  Today a cell phone or a POTS line phone
> can send DTMF clearly enough to operate a credit card or other interactive
> tone based system at the far end.  With SIP it is sometimes "chancy".
>
> Is there a plain 64K codec that would simply pass through the SIP server and
> be handed off to a PRI or phone co. trunk on a T1 on the other side of the
> SIP server?  Digital 64K telco sounds very good as a phone conversation.
>
> Cary Fitch

Isn't that ulaw/alaw?

j

>
>
>
>
>
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Re: [asterisk-users] allowguest defaults to yes for SIP

[Danny Nicholas]

"Gentlemens clubs" usually don't have any.  While LH probably has a valid
point, jumping on Til isn't the way to bring it home.  You can't protect the
stupid or lazy from themselves.  If you can't do this right, pay someone
else to.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Howard
Sent: Thursday, November 12, 2009 12:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] allowguest defaults to yes for SIP

Tilghman Lesher wrote:
> The issue in question was suspended, while the reporter makes the case on
the
> Asterisk-dev mailing list, which is not this list.  The opinions there
amongst 
> contributors (meritocracy, not democracy) are that keeping the sample
> configuration as it is now is probably the way to go.
>   

Sigh... of course.  It's a gentlemen's club and only members have a say.

> If you want to create a new issue and attach your patch there, I'll look
at
> it.

I sent a patch.  I pointed you at a case.  That should have been FAR 
more than enough for my attempt at contribution to be acceptable.

Thanks,

Lee.

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Re: [asterisk-users] SIP source address error

[Dave Platt]

> It's set to bind to 0.0.0.0, which IIRC is nothing strange.
> 
> The question remains: how can a remote Asterisk server be receiving  
> SIP packets that still contain the private net IP address of a client?

It sounds to me as if the client hasn't been told to use its
gateway's public IP address in the SIP conversation, and as if
the client isn't sending its outbound packets through a gateway/NAT
which is SIP-aware and can rewrite the SIP data accordingly.

There are several approaches which can work:

-  The gateway is properly configured to forward its external
   ports to the client, and the client is manually configured to
   use the gateway's external IP address in its SIP protocol
   exchanges.

-  The gateway does port forwarding and NAT properly, and is
   also SIP-aware - it intercepts and rewrites the contents of
   the outbound SIP packets, changing the IP address and port
   given by the client to its own IP address and whatever
   external port it has NAT'ed / redirected to the client.

-  The gateway does port forwarding and NAT properly, and the
   client is configured to use STUN to figure out what public
   IP address/port its packets are being NAT'ed to.

-  The client doesn't talk directly to the outside peers, but
   goes through a SIP proxy running on the gateway.

In your case, it sounds as if the client and gateway aren't
doing one of these things.  As a result, the client's SIP
protocol packets still contain its private-net IP and port,
at the time they reach the remote server.

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Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1

[Steve Howes]

On 12 Nov 2009, at 17:09, Cyprus VoIP wrote
>> DNS/Gateway ok on server?
> Yes. The problem is with the FreePBX modules. I forced the mirror file
> to include version 2.5, and I get a list, but when I try to install  
> the
> modules, it says that the modules need FreePBX version 2.5.0alpha or  
> rc1
> or higher, but although 2.5.2 is indeed higher, it's rejected. I've
> given up on this software and will continue to edit my .conf files
> manually. what a waste of time :-(

Well, its clearly not an Asterisk issue, so yes it is a waste of time :)

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Re: [asterisk-users] allowguest defaults to yes for SIP

[Lee Howard]

Tilghman Lesher wrote:
> The issue in question was suspended, while the reporter makes the case on the
> Asterisk-dev mailing list, which is not this list.  The opinions there 
> amongst 
> contributors (meritocracy, not democracy) are that keeping the sample
> configuration as it is now is probably the way to go.
>   

Sigh... of course.  It's a gentlemen's club and only members have a say.

> If you want to create a new issue and attach your patch there, I'll look at
> it.

I sent a patch.  I pointed you at a case.  That should have been FAR 
more than enough for my attempt at contribution to be acceptable.

Thanks,

Lee.

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Re: [asterisk-users] allowguest defaults to yes for SIP

[Tilghman Lesher]

On Thursday 12 November 2009 12:08:39 Lee Howard wrote:
> Tilghman Lesher wrote:
> > On Thursday 12 November 2009 09:53:17 Lee Howard wrote:
> >> And yet this point is not even made clear in the doc/security.txt file.
> >> It says to not use "default" for anything you don't want to get abused,
> >> but it doesn't say *why*.  So I can envision, then, someone reading the
> >> document and then changing context=internal in the [general] section of
> >> sip.conf... and thinking that they responded correctly to what the
> >> document said.
> >
> > You've just made a case for enhancing the documentation, not for changing
> > the defaults.  If you contribute documentation changes to this effect on
> > the issue tracker, I would be more than happy to commit them.
>
> The patch is attached.  Feel free to add it to bug tracker issue ID
> 16226 which some maintainer was happy enough to close already.

The issue in question was suspended, while the reporter makes the case on the
Asterisk-dev mailing list, which is not this list.  The opinions there amongst 
contributors (meritocracy, not democracy) are that keeping the sample
configuration as it is now is probably the way to go.

If you want to create a new issue and attach your patch there, I'll look at
it.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] allowguest defaults to yes for SIP

[Lee Howard]

Tilghman Lesher wrote:

On Thursday 12 November 2009 09:53:17 Lee Howard wrote:
  


These people should need to deliberately use allowguest=yes.  I would
venture to guess that these people already know who they are and
deliberately have this set.  I would venture to guess that there are
far, far more people who have it turned on by default who really don't
want it that way than there are who expected it to be that way and
desire it to so be.



And the people who use this probably believe that YOU should be the one
who has to deliberately turn this option off.  I would venture to guess that
90% of all statistics are made up on the spot, including this one and the
two you specified above.
  


I made it clear that they were guesses.  But, please *DO* take a vote on 
this.  I'm not seeing anyone but you stand up to support the default 
setting.  Unless you take a vote there's really nothing I can do but guess.


The fact that this problem is being exploited leads me to believe that 
this is far-more prevalent a problem than just my single case.  Take 
care of your users when you can do something so easily.  Don't 
deliberately let them learn things the hard way on the basis that they 
should have known better.  The mere fact that this issue is addressed in 
doc/security.txt should be an indication that there is a common risk 
that could be averted.



And yet this point is not even made clear in the doc/security.txt file.
It says to not use "default" for anything you don't want to get abused,
but it doesn't say *why*.  So I can envision, then, someone reading the
document and then changing context=internal in the [general] section of
sip.conf... and thinking that they responded correctly to what the
document said.



You've just made a case for enhancing the documentation, not for changing
the defaults.  If you contribute documentation changes to this effect on the
issue tracker, I would be more than happy to commit them.


The patch is attached.  Feel free to add it to bug tracker issue ID 
16226 which some maintainer was happy enough to close already.


And, for what it's worth let me restate my vote that the default for 
allowguest be changed to "no" on the basis of keeping ignorant people 
from making a stupid mistake.


Thanks,

Lee.

--- asterisk-1.4.21.2/doc/security.txt.old	2009-11-12 09:53:03.0 -0800
+++ asterisk-1.4.21.2/doc/security.txt	2009-11-12 09:56:38.0 -0800
@@ -48,12 +48,15 @@
 
 Therefore, you should NOT allow access to outgoing or toll services in
 contexts that are accessible (especially without a password) from incoming
-channels, be they IAX channels, FX or other trunks, or even untrusted
-stations within you network.  In particular, never ever put outgoing toll
-services in the "default" context.  To make things easier, you can include
-the "default" context within other private contexts by using:
+channels, be they IAX channels, SIP channels, FX or other trunks, or even 
+untrusted stations within you network.  Keep in mind that the default setting
+for SIP configuration is allowguest=yes.  So unauthenticated SIP users will, 
+by default, be able to access the context specified in the [general] section.
+Therefore, never ever put outgoing toll services in the "public" context.  
+To make things easier, you can include the "default" context within other 
+private contexts by using:
 
-	include => default
+	include => public
 
 in the appropriate section.  A well designed PBX might look like this:
 
@@ -63,9 +66,9 @@
 
 [local]
 exten => _9NXXNXXX,1,Dial(Zap/g2/${EXTEN:1})
-include => default
+include => public
 
-[default]
+[public]
 exten => 6123,Dial(Zap/1)
 
 
--- asterisk-1.4.21.2/configs/sip.conf.sample.old	2009-11-12 09:57:19.0 -0800
+++ asterisk-1.4.21.2/configs/sip.conf.sample	2009-11-12 09:58:41.0 -0800
@@ -24,7 +24,7 @@
 ;
 
 [general]
-context=default			; Default context for incoming calls
+context=public			; Default context for incoming calls
 ;allowguest=no			; Allow or reject guest calls (default is yes)
 allowoverlap=no			; Disable overlap dialing support. (Default is yes)
 ;allowtransfer=no		; Disable all transfers (unless enabled in peers or users)
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Re: [asterisk-users] allowguest defaults to yes for SIP

[Tilghman Lesher]

On Thursday 12 November 2009 09:53:17 Lee Howard wrote:
> Tilghman Lesher wrote:
> > On Thursday 12 November 2009 07:47:34 Lee Howard wrote:
> >> In your sip.conf file allowguest defaults to yes.  This means that
> >> anyone that can reach the SIP ports on that system has access to make
> >> unauthenticated calls, by default.  The administrator actually has to go
> >> in and turn it off to prevent unauthenticated SIP calls (in whatever
> >> context [general] points at).
> >
> > Actually, they only have access to your default context.  Whether you
> > make available outgoing calls in your default context is your choice.  By
> > default, there is no capability of making outgoing calls from your
> > default context.
>
> Well, yes, the default configuration is useless.  But, let's say I
> follow doc/security.txt exactly and have this:
>
> [default]
> exten => 6123,Dial(Zap/1)
>
> ... therefore, by default, an unauthenticated user from anywhere can
> call the extension Zap/1.  It's not my point whether or not this poses a
> financial risk.  My point is that this is an insecure default behavior
> to have allowguest=yes.
>
> >> Does anyone else agree with me that this is a poor default?  I'd like to
> >> see the default setting changed.
> >
> > The purpose of the allowguest option is to allow persons to call into
> > your system from a zero-knowledge position.  This allows you to publish a
> > general SIP address as a point of contact.
>
> These people should need to deliberately use allowguest=yes.  I would
> venture to guess that these people already know who they are and
> deliberately have this set.  I would venture to guess that there are
> far, far more people who have it turned on by default who really don't
> want it that way than there are who expected it to be that way and
> desire it to so be.

And the people who use this probably believe that YOU should be the one
who has to deliberately turn this option off.  I would venture to guess that
90% of all statistics are made up on the spot, including this one and the
two you specified above.

> > The reason why it is set that way in the
> > sample configuration is to make it easy for new users to get to that
> > magic moment when Asterisk first responds to their call (in essence, to
> > get the user "hooked").
>
> This is a poor excuse for a poor default security setting.

It's not a security setting; it's a functionality setting.  You see it behind
rose-tinted spectacles because in your specific case, you don't have a
use for it.  That's fine, but please do not extrapolate from your limited
use cases what the global settings should be.

> >> It seems to me that this default is the reason behind the
> >> doc/security.txt bias against using the "default" context for toll
> >> calls.
> >
> > Correct, you should be using something like "internal" instead.
>
> And yet this point is not even made clear in the doc/security.txt file.
> It says to not use "default" for anything you don't want to get abused,
> but it doesn't say *why*.  So I can envision, then, someone reading the
> document and then changing context=internal in the [general] section of
> sip.conf... and thinking that they responded correctly to what the
> document said.

You've just made a case for enhancing the documentation, not for changing
the defaults.  If you contribute documentation changes to this effect on the
issue tracker, I would be more than happy to commit them.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] How to send DTMF on Zaptel with 50ms tone duration and 50ms gap between the digits?

[Zeeshan Zakaria]

Hi,

After some testing I've found out that my client's hardware recognizes DTMF
only if digits are sent 50ms apart with 50ms of tone duration. This was
tested using a test device which generates DTMF.

Now asterisk doesn't do it by default because digits going out from Asterisk
are not being recognized.

Using command sendDTMF, I can control inter-digit duration, and using
toneduration I can control duration of tone per digit. But I can't find a
way to do both at the same time

Application sendDTMF simply ignores the value set in toneduration and sends
DTMF at some default value, which I don't know what it is, but it is
obviously not 50ms because the hardware can't reliably recognized the
digits.

Is there a way I can send digits with 50ms tone duration and 50ms gap
between them?

-- 
Zeeshan A Zakaria
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Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1

[Steve Howes]

On 12 Nov 2009, at 16:54, Nelson Granados wrote:

> Dear Steve,
>
> Do you have your DNS settings ok?

Yes, but its not me with the problem. ;)

Steve

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Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1

[Cyprus VoIP]

>> The problem is that the online module update is not working for me
>> (Cannot connect to online repository (mirror.freepbx.org). Online
>> modules are not available.) and I couldn't find online a working
>> solution :-(
> 
> DNS/Gateway ok on server?
> 
Yes. The problem is with the FreePBX modules. I forced the mirror file 
to include version 2.5, and I get a list, but when I try to install the 
modules, it says that the modules need FreePBX version 2.5.0alpha or rc1 
or higher, but although 2.5.2 is indeed higher, it's rejected. I've 
given up on this software and will continue to edit my .conf files 
manually. what a waste of time :-(

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Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1

[Nelson Granados]

Dear Steve,

Do you have your DNS settings ok?
Otherwise include these settings(DNS1 DNS2) in your network configuration.

Regards,

Nelson Granados

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cyprus VoIP
Sent: Thursday, November 12, 2009 11:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1

>> I tried to install Asterisk + Asterisk addons + FreePBX (latest  
>> versions
>> of all), but in the FreePBX screen, I don't have the option to set  
>> ring
>> groups and IVRs
>>
>> Can anyone tell me what I'm doing wrong?
> 
> You are not posting on the FreePBX forums? ;)
> 
I figured "Asterisk-Users" would know ;)
 >
> The solution however, is to install the modules using the module admin.
> 
The problem is that the online module update is not working for me 
(Cannot connect to online repository (mirror.freepbx.org). Online 
modules are not available.) and I couldn't find online a working 
solution :-(
> 
> Steve
> 


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Re: [asterisk-users] Can't connect to voip provider over NAT

[Carlos Chavez]

On Thu, 2009-11-12 at 14:50 +1100, Michael Wyres wrote:
> Have you tried "nat=yes" in the definition in sip.conf?
> 
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy
> Sent: Thursday, 12 November 2009 13:30
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Can't connect to voip provider over NAT
> 
> Hello.
> 
> I'm trying to test an Asterisk server by using a VOIP provider for 
> international calls but, I'm having problems trying to get my server 
> communicate with theirs. I don't know if I'm having all these issues becuase 
> I'm behind NAT or what. I have the following in my server's sip.conf:
> 
> [provider]
> type=peer
> host=
> username=
> secret=
> port=5060
> canreinvite=YES
> dtmfmode=rfc2833
> 
> I've tried opening all ports to test this but, still doesn't work. Now, I 
> need to know which especific ports to open in order to allow sip flow 
> correctly. Also enabled/opened ports 5060 - 5070 and the rtp: rtpstart=1
> rtpend=2
> 
> Don't know what else to try. Please help.
> 
> Thanks in advanced for your help.
> 
> 
I think this is more a problem that you are not setting your external
IP address correctly so the provider can send RTP back to you.  Make
sure you have either "externip", "externhost" or "stunaddr"(1.6) set
correctly.  The do a "sip show settings" in the CLI to see if the
correct address is set.  If you are behind nat "canreinvite" should be
set to no.


-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1

[Steve Howes]

On 12 Nov 2009, at 16:29, Cyprus VoIP wrote:
> The problem is that the online module update is not working for me
> (Cannot connect to online repository (mirror.freepbx.org). Online
> modules are not available.) and I couldn't find online a working
> solution :-(

DNS/Gateway ok on server?

S

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Re: [asterisk-users] Libpri-1.4.10.2 Released

[Barry L. Kline]

Karl Fife wrote:

> 
> Perhaps there's an arcane way to query lipbri the older releases from the 
> CLI?  Can anyone speak to that?
> 

Quick and dirty:

strings /usr/lib/libpri.so

That's CLI, tho' not the one you're talking about.

Barry


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Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1

[Cyprus VoIP]

>> I tried to install Asterisk + Asterisk addons + FreePBX (latest  
>> versions
>> of all), but in the FreePBX screen, I don't have the option to set  
>> ring
>> groups and IVRs
>>
>> Can anyone tell me what I'm doing wrong?
> 
> You are not posting on the FreePBX forums? ;)
> 
I figured "Asterisk-Users" would know ;)
 >
> The solution however, is to install the modules using the module admin.
> 
The problem is that the online module update is not working for me 
(Cannot connect to online repository (mirror.freepbx.org). Online 
modules are not available.) and I couldn't find online a working 
solution :-(
> 
> Steve
> 


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Re: [asterisk-users] SendText

[Thomas Perron]

OK.
Thanks



On Thu, Nov 12, 2009 at 4:33 AM, Tarek Sawah  wrote:

> i have my own SMS provider as we sell SMS .. so i have setup my call center
> with SMS sending for several services and alerts like a Missed Call when i'm
> not registered it will send me an sms to alert me.
> it's pretty the same as Matt discribed.. you call an AGI which may use cURL
> to hit the HTTP API
>
> -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria:
> +963 944 618286 USA: +1 347 562 2308
>
>
>
> --
> Date: Mon, 9 Nov 2009 22:19:08 -0500
> From: thomas.per...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] SendText
>
>
> Will text messages work to non-SIP enpoints using your logic/code?
> thank you
>
> On Mon, Nov 9, 2009 at 8:59 PM, Matt Riddell wrote:
>
>  On 10/11/09 12:58 PM, Thomas Perron wrote:
> > Does anyone have any success with sending a text message from
> > extensions.conf
> > to an PSTN endpoint such as a cell phone?
> >
> > If so, kindly send configuration for this part.  I am working on an IVR
> > and want
> > callers to get a text message at a particular part of the call, after
> > dialing a defined character (such as 22).
>
> We use clickatel.
>
> Basically we use the PHP API and call it via an AGI which sends texts.
>
> Therefore the extensions.conf is pretty sparse:
>
> exten => s,1,Read(destination)
> exten => s,2,AGI(agi://127.0.0.1/send_sms.php)
>
> Pseudo code for send_sms is:
>
> 1. Read AGI variables
> 2. Get destination variable
> 3. Include clickatel API file
> 4. call send_sms function
>
> We also provide an API from our telephone exchanges, but to be fair
> you're likely better off just using clickatel yourself :D
>
> --
> Cheers,
>
> Matt Riddell
> Director
> ___
>
> http://www.venturevoip.com/news.php (Daily Asterisk News)
> http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
> http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
>
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>
>
> --
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> more.
>
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Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1

[Steve Howes]

On 12 Nov 2009, at 15:38, Cyprus VoIP wrote:
> I tried to install Asterisk + Asterisk addons + FreePBX (latest  
> versions
> of all), but in the FreePBX screen, I don't have the option to set  
> ring
> groups and IVRs
>
> Can anyone tell me what I'm doing wrong?

You are not posting on the FreePBX forums? ;)

The solution however, is to install the modules using the module admin.

Steve

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Re: [asterisk-users] allowguest defaults to yes for SIP

[Lee Howard]

Tilghman Lesher wrote:
> On Thursday 12 November 2009 07:47:34 Lee Howard wrote:
>   
>> In your sip.conf file allowguest defaults to yes.  This means that
>> anyone that can reach the SIP ports on that system has access to make
>> unauthenticated calls, by default.  The administrator actually has to go
>> in and turn it off to prevent unauthenticated SIP calls (in whatever
>> context [general] points at).
>> 
>
> Actually, they only have access to your default context.  Whether you make
> available outgoing calls in your default context is your choice.  By default,
> there is no capability of making outgoing calls from your default context.
>   

Well, yes, the default configuration is useless.  But, let's say I 
follow doc/security.txt exactly and have this:

[default]
exten => 6123,Dial(Zap/1)

... therefore, by default, an unauthenticated user from anywhere can 
call the extension Zap/1.  It's not my point whether or not this poses a 
financial risk.  My point is that this is an insecure default behavior 
to have allowguest=yes.


>> Does anyone else agree with me that this is a poor default?  I'd like to
>> see the default setting changed.
>> 
>
> The purpose of the allowguest option is to allow persons to call into your
> system from a zero-knowledge position.  This allows you to publish a general
> SIP address as a point of contact.

These people should need to deliberately use allowguest=yes.  I would 
venture to guess that these people already know who they are and 
deliberately have this set.  I would venture to guess that there are 
far, far more people who have it turned on by default who really don't 
want it that way than there are who expected it to be that way and 
desire it to so be.

> The reason why it is set that way in the
> sample configuration is to make it easy for new users to get to that magic
> moment when Asterisk first responds to their call (in essence, to get the user
> "hooked").
>   

This is a poor excuse for a poor default security setting.

>> It seems to me that this default is the reason behind the
>> doc/security.txt bias against using the "default" context for toll calls.
>> 
>
> Correct, you should be using something like "internal" instead.

And yet this point is not even made clear in the doc/security.txt file.  
It says to not use "default" for anything you don't want to get abused, 
but it doesn't say *why*.  So I can envision, then, someone reading the 
document and then changing context=internal in the [general] section of 
sip.conf... and thinking that they responded correctly to what the 
document said.

If this default is to persist then I think that it behooves the 
developers to at least make this exposure clear to the users.  
Therefore, the in the [general] section of sip.conf the context should 
not be set to "default", but rather to "unauthorized" or "public" or 
"open" or "free" or something that makes it clear that this is where 
unauthenticated SIP calls go.

Thanks,

Lee.


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[asterisk-users] solution for NAT issues?

[Ron]

Hi All,


I been having issues on my users behind NAT, even if i hard set a 
specific port on the phone, there are some network that NAT's it out to 
a different port, in turn, some time later the phone could not be 
reached by the server. i think because on the server, e.g. the user is 
still registered on port 49923 but when the request is sent to that port 
  the NAT router does not forward port 49923 to port of the IP phone, 
maybe nat mapping has expired or something.

I have tried STUN, still sometimes the phones just cannot be reached.
is there any other software to manage binding of ports on specific users 
so that the routers always keeps the port mapped to port of the ip phone .
TIA

Regards,
Ron

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[asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1

[Cyprus VoIP]

Hello,

I tried to install Asterisk + Asterisk addons + FreePBX (latest versions 
of all), but in the FreePBX screen, I don't have the option to set ring 
groups and IVRs
.
Can anyone tell me what I'm doing wrong?

Thanks,

Andreas

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Re: [asterisk-users] "POTS 4K linear codec"

[Jared Smith]

On Thu, 2009-11-12 at 08:53 -0600, Cary Fitch wrote:
> Digital 64K telco sounds very good as a phone conversation.

Digital 64k audio coming across a T1 is essentially identical to the
ulaw codec in VoIP.  Digital 64k audio coming across an E1 is
essentially identical to the alaw codec.

-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] allowguest defaults to yes for SIP

[Tilghman Lesher]

On Thursday 12 November 2009 07:47:34 Lee Howard wrote:
> In your sip.conf file allowguest defaults to yes.  This means that
> anyone that can reach the SIP ports on that system has access to make
> unauthenticated calls, by default.  The administrator actually has to go
> in and turn it off to prevent unauthenticated SIP calls (in whatever
> context [general] points at).

Actually, they only have access to your default context.  Whether you make
available outgoing calls in your default context is your choice.  By default,
there is no capability of making outgoing calls from your default context.

> Does anyone else agree with me that this is a poor default?  I'd like to
> see the default setting changed.

The purpose of the allowguest option is to allow persons to call into your
system from a zero-knowledge position.  This allows you to publish a general
SIP address as a point of contact.  The reason why it is set that way in the
sample configuration is to make it easy for new users to get to that magic
moment when Asterisk first responds to their call (in essence, to get the user
"hooked").

> It seems to me that this default is the reason behind the
> doc/security.txt bias against using the "default" context for toll calls.

Correct, you should be using something like "internal" instead.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] allowguest defaults to yes for SIP

[Tilghman Lesher]

On Thursday 12 November 2009 09:00:45 Dan Journo wrote:
> Am I correct in saying that the without allowguest=no anyone can connect
> and make calls through the default context?
>
> If allowguest is set to no, how can I ensure that incoming calls can still
> be received from our DDI supplier?

You're correct in stating that this is the purpose of the allowguest
configuration option.  If you disable it, only peers with which you have
established settings will be able to call into your system.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] allowguest defaults to yes for SIP

[Tilghman Lesher]

On Thursday 12 November 2009 08:59:16 Danny Nicholas wrote:
> Without the allowguest=no, Asterisk doesn't put up any defense against an
> unauthorized guest.  You still have NAT/Firewall/IPTABLE "defenses", for
> what they are worth.  The trick is to get what you need without allowing
> what you don't want.

Don't assume that all guests are uninvited.  The allowguest setting permits
you to publish a SIP address at which new customers may make initial contact.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Termination Question

[Karl Fife]

...and with a packet switched transport layer, the 'hairpin' route through A 
may create problematic levels of latency--latency that would perhaps NOT have 
been problematic on a classic circuit switched route, so it's definitely 
advisable to nail up a connection between b and c.

-K


- Original Message - 
  From: Tarek Sawah 
  To: Asterisk Users 
  Sent: Thursday, November 12, 2009 8:28 AM
  Subject: Re: [asterisk-users] Termination Question


  for the sake of bandwidth you are supposed to connect each two servers 
together.. otherwise calls between B && C will have to go through A .

  -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: 
+963 944 618286 USA: +1 347 562 2308 




--
  From: i...@saudihome.com
  To: asterisk-users@lists.digium.com
  Date: Thu, 12 Nov 2009 16:13:10 +0300
  Subject: [asterisk-users] Termination Question


  Hello,

  I would like to know how the following scenario works:



  I have 3 Asterisk servers, A,B & C,  each one is located in a different 
country.

  Asterisk A is the main one, and both B & C are connected to it.



  My question is, when a call is originated from B to C, it will have to go 
through A, but does A makes a peer connection between B & C to eliminate 
bandwidth and latency, or the call has to go through A ???



  Thanks.





--
  Windows 7: Unclutter your desktop. Learn more. 


--


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Re: [asterisk-users] allowguest defaults to yes for SIP

[Danny Nicholas]

Without the allowguest=no, Asterisk doesn't put up any defense against an
unauthorized guest.  You still have NAT/Firewall/IPTABLE "defenses", for
what they are worth.  The trick is to get what you need without allowing
what you don't want.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, November 12, 2009 9:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] allowguest defaults to yes for SIP

Am I correct in saying that the without allowguest=no anyone can connect and
make calls through the default context?

If allowguest is set to no, how can I ensure that incoming calls can still
be received from our DDI supplier?

Many Thanks
Dan

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: 12 November 2009 14:46
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] allowguest defaults to yes for SIP

Just my .02 - the guest context should torture or hangup instead of being
empty.  That might encourage a masochistic hacker though...

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator
TOOTAI
Sent: Thursday, November 12, 2009 8:42 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] allowguest defaults to yes for SIP

Lee Howard a écrit :
> In your sip.conf file allowguest defaults to yes.  This means that 
> anyone that can reach the SIP ports on that system has access to make 
> unauthenticated calls, by default.  The administrator actually has to go 
> in and turn it off to prevent unauthenticated SIP calls (in whatever 
> context [general] points at).
>
> Does anyone else agree with me that this is a poor default?  I'd like to 
> see the default setting changed.
>
> It seems to me that this default is the reason behind the 
> doc/security.txt bias against using the "default" context for toll calls.
>   
Agree. Another possibility would be to have a guestcontext defined in 
default. This context would exist but empty.

-- 
Daniel

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Re: [asterisk-users] BLF with SPA941?

[Ex Vito]

 Although I've never tested such feature on those devices, I know
 that it was only enabled in a recent firmware (6.1.3a/6.1.5a ?).

 Are you running it ?
--
  exvito

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[asterisk-users] "POTS 4K linear codec"

[Cary Fitch]

I am not sure what the problems are and the reasons for the basic 64K modems
used in VOIP are.  I understand the compressed codecs that get the bandwidth
down to 20-30 K.  And perhaps the 64K units give much better potential audio
than you would get on a normal POTS line.

But, as phone circuits VOIP/SIP doesn't seem to perform as well as plane old
phones.

Multiple transcodings cause issues.  Today a cell phone or a POTS line phone
can send DTMF clearly enough to operate a credit card or other interactive
tone based system at the far end.  With SIP it is sometimes "chancy".

Is there a plain 64K codec that would simply pass through the SIP server and
be handed off to a PRI or phone co. trunk on a T1 on the other side of the
SIP server?  Digital 64K telco sounds very good as a phone conversation.

Cary Fitch





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Re: [asterisk-users] allowguest defaults to yes for SIP

[Dan Journo]

Am I correct in saying that the without allowguest=no anyone can connect and 
make calls through the default context?

If allowguest is set to no, how can I ensure that incoming calls can still be 
received from our DDI supplier?

Many Thanks
Dan

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: 12 November 2009 14:46
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] allowguest defaults to yes for SIP

Just my .02 - the guest context should torture or hangup instead of being
empty.  That might encourage a masochistic hacker though...

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator
TOOTAI
Sent: Thursday, November 12, 2009 8:42 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] allowguest defaults to yes for SIP

Lee Howard a écrit :
> In your sip.conf file allowguest defaults to yes.  This means that 
> anyone that can reach the SIP ports on that system has access to make 
> unauthenticated calls, by default.  The administrator actually has to go 
> in and turn it off to prevent unauthenticated SIP calls (in whatever 
> context [general] points at).
>
> Does anyone else agree with me that this is a poor default?  I'd like to 
> see the default setting changed.
>
> It seems to me that this default is the reason behind the 
> doc/security.txt bias against using the "default" context for toll calls.
>   
Agree. Another possibility would be to have a guestcontext defined in 
default. This context would exist but empty.

-- 
Daniel

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Re: [asterisk-users] my kernel is dazed and confused

[Dr. Michael J. Chudobiak]

On 11/12/2009 09:42 AM, Francesco Peeters wrote:
> Dr. Michael J. Chudobiak wrote:
>> Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason
>> a0 on CPU 0.
>> Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely
>> on the PCI bus.
>> Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue
>>
>> Would my Digium TDM410P cause an NMI, or is my computer failing?
>>
>> - Mike
>>
> Googling for the error seems to indicate a possible kernel bug... Are
> you using Ubuntu or Debian?...

I'm using Fedora 11, kernel 2.6.30.8-64.fc11.x86_64.

- Mike

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Re: [asterisk-users] allowguest defaults to yes for SIP

[Danny Nicholas]

Just my .02 - the guest context should torture or hangup instead of being
empty.  That might encourage a masochistic hacker though...

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator
TOOTAI
Sent: Thursday, November 12, 2009 8:42 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] allowguest defaults to yes for SIP

Lee Howard a écrit :
> In your sip.conf file allowguest defaults to yes.  This means that 
> anyone that can reach the SIP ports on that system has access to make 
> unauthenticated calls, by default.  The administrator actually has to go 
> in and turn it off to prevent unauthenticated SIP calls (in whatever 
> context [general] points at).
>
> Does anyone else agree with me that this is a poor default?  I'd like to 
> see the default setting changed.
>
> It seems to me that this default is the reason behind the 
> doc/security.txt bias against using the "default" context for toll calls.
>   
Agree. Another possibility would be to have a guestcontext defined in 
default. This context would exist but empty.

-- 
Daniel

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Re: [asterisk-users] allowguest defaults to yes for SIP

[Administrator TOOTAI]

Lee Howard a écrit :
> In your sip.conf file allowguest defaults to yes.  This means that 
> anyone that can reach the SIP ports on that system has access to make 
> unauthenticated calls, by default.  The administrator actually has to go 
> in and turn it off to prevent unauthenticated SIP calls (in whatever 
> context [general] points at).
>
> Does anyone else agree with me that this is a poor default?  I'd like to 
> see the default setting changed.
>
> It seems to me that this default is the reason behind the 
> doc/security.txt bias against using the "default" context for toll calls.
>   
Agree. Another possibility would be to have a guestcontext defined in 
default. This context would exist but empty.

-- 
Daniel

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Re: [asterisk-users] my kernel is dazed and confused

[Francesco Peeters]

Dr. Michael J. Chudobiak wrote:
> Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason 
> a0 on CPU 0.
> Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely 
> on the PCI bus.
> Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue
>
>
> Would my Digium TDM410P cause an NMI, or is my computer failing?
>
> - Mike
>
>
>   
Googling for the error seems to indicate a possible kernel bug... Are
you using Ubuntu or Debian?...


-- 
Francesco

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Re: [asterisk-users] allowguest defaults to yes for SIP

[Doug Lytle]

Lee Howard wrote:
> Does anyone else agree with me that this is a poor default?  I'd like to
> see the default setting changed.
>
>
I've always considered it to be good practice that something that may 
leave your system vulnerable, should be disabled by default.

So yes, I would agree.

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] state_interface backport issue

[Robert Broyles]

Any takers?
Still trying to get this resolved...

Thanks!

Robert Broyles wrote:
> It's my understanding that the backport is available now in 1.4. 
> However, seem to be having some issues with it. Just wondering if I 
> have everything setup right.
>
> I'm running 1.4.26.2 realtime.
> queue_members:
> `uniqueid` int(10) unsigned NOT NULL auto_increment,
>  `membername` varchar(40) default NULL,
>  `queue_name` varchar(128) default NULL,
>  `interface` varchar(128) default NULL,
>  `penalty` int(11) default NULL,
>  `paused` int(1) default NULL,
>  `state_interface` varchar(128) NOT NULL,
>
> Data:
> 1, Name, QUEUENAME, Local/1...@agents/n, 1, , SIP/100
>
> Local agents are setup setup in an 'agents' context.
>
> [agents]
> exten => 1050,1,Set(agentsip=${DB(agent_sip/1050)}
> exten => 1050,2,Dial(SIP/${agentsip})
>
> Queue shows the agent as unavailable when the SIP device (SIP/100) is 
> down. (as I would hope)... but shows the agent as available all the 
> other times.
>
> As a result my CLi is on fire with 'busy' notices, because it's trying 
> to ring an agent even when they are on a call. If I remove the 
> state_interface, it shows them as 'busy' in the queue, and doesn't 
> ring them.
>
> Let's see, what else did I forget? Other details:
>
> sip.conf: limitonpeers=yes
> and call-limit=5 on each SIP device
> queue.conf: ringinuse=no
>
> Anything else I should look for?
>
>
> Thanks!
>
> -Rob
>


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[asterisk-users] my kernel is dazed and confused

[Dr. Michael J. Chudobiak]

Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason 
a0 on CPU 0.
Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely 
on the PCI bus.
Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue


Would my Digium TDM410P cause an NMI, or is my computer failing?

- Mike



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Re: [asterisk-users] Termination Question

[Tarek Sawah]

for the sake of bandwidth you are supposed to connect each two servers 
together.. otherwise calls between B && C will have to go through A .

--
AHD Tarek Sawah

Integrated Digital Systems

CCNA, MCSE, RHCE, VoIP

Syria: +963 944 618286

USA: +1 347 562 2308






From: i...@saudihome.com
To: asterisk-users@lists.digium.com
Date: Thu, 12 Nov 2009 16:13:10 +0300
Subject: [asterisk-users] Termination Question
















Hello,

I would like to know how the following scenario works:

 

I have 3 Asterisk servers, A,B & C,  each one is
located in a different country.

Asterisk A is the main one, and both B & C are connected
to it.

 

My question is, when a call is originated from B to C, it
will have to go through A, but does A makes a peer connection between B & C
to eliminate bandwidth and latency, or the call has to go through A ???

 

Thanks.

 

  
_
Windows 7: Unclutter your desktop.
http://go.microsoft.com/?linkid=9690331&ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen:112009___
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Re: [asterisk-users] Dell Poweredge T105

[Olivier]

2009/11/12 Olivier 

> Hello,
>
> I someone successfully using Asterisk and Debian on an Opteron-enabled Dell
> Poweredge T105 ?
> If positive, which architecture (i386, amd, ...) w
>
If positive, which architecture (i386, amd, ...) was chosen ?

Regards
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[asterisk-users] Dell Poweredge T105

[Olivier]

Hello,

I someone successfully using Asterisk and Debian on an Opteron-enabled Dell
Poweredge T105 ?
If positive, which architecture (i386, amd, ...) w
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[asterisk-users] Codec interface

[Bill Shaw]

Hi All,

I need to interface a codec-type device to Asterisk.  The device uses a 
TI TLV320AIC1110 codec in 15 bit linear data mode with a 2.048 MHz clock 
supplied by the device.  I am about to start on a custom hardware design 
to interface this device to  the computer,  but thought I'd ask here 
before I get started on it.  Does anyone know of a hardware interface 
that is already being manufactured that can tie a codec-based device 
into Asterisk?

Thanks in advance,

Bill

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Re: [asterisk-users] RTPAUDIOQOS

[covici]

OK, how do you get such information -- at times it would be very useful
to know.

Darryl Dunkin  wrote:

> Sorry to reply so late, I am months behind and catching up.
> 
>  
> 
> I have been inspecting this on my own systems, and the results are 
> inconsistent to say the least. I’ve been dumping these to the verbose logs 
> for some time and monitoring them, but I have not been able to determine why 
> the numbers are so far off. I am more concerned with the packets lost due to 
> priority queuing within our network.
> 
>  
> 
> Here is an example just today:
> 
> ssrc=583450581
> 
> themssrc=1093951555
> 
> lp=0
> 
> rxjitter=0.003219
> 
> rxcount=1100
> 
> txjitter=0.000275
> 
> txcount=1108
> 
> rlp=57702
> 
> rtt=0.036000
> 
>  
> 
> If the txcount is only 1108, how can the remote lost packet count be 57702? 
> Unless the call was nearly inaudible?
> 
>  
> 
> I did verify with this end user, and the call was just fine. Is this an issue 
> with the phone at the remote end misreporting?
> 
>  
> 
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mindaugas Kezys
> Sent: Tuesday, September 22, 2009 01:01
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] RTPAUDIOQOS
> 
>  
> 
> Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified
> 
>  
> 
> Regards,
> 
> Mindaugas Kezys
> 
> http://www.kolmisoft.com
> 
> VoIP Billing and Routing Solutions
> 
>  
> 
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA
> Sent: 2009 m. rugsėjo 22 d. 09:28
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] RTPAUDIOQOS
> 
>  
> 
> hey all,
> 
> can any body know what this parameter stands for 
> 
> i got RTPAUDIOQOS while i have SIP channels 
> 
> but could not understand then what this parameter tell
> 
> ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000
> 
> if any one know plese help me to or give any documentation link
> 
> regards
> Dhaval
> 
> 
> 
> Alternatives:
> 
> 
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-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 cov...@ccs.covici.com

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Re: [asterisk-users] Incoming Call Ring

[Danny Nicholas]

Depending on your phone, you can use CallWaitingRing to ring the phone
anyway. I do this with Polycom 501's.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, November 12, 2009 6:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Incoming Call Ring

 

Hello,

 

I have Asterisk set up with 6 extensions. When a call comes in, I use a Dial
command to call all the extensions together until someone picks up.

 

The problem is, when there is an incoming call and an extension is in use,
if the extension puts down the phone while the incoming call is still
ringing, that extension doesn't ring. This is because when the Dial command
was executed, that extension was busy.

 

Is there any way to make that extension ring as soon as its available if
there is still an incoming call?

 

Thanks

Dan

 

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[asterisk-users] allowguest defaults to yes for SIP

[Lee Howard]

In your sip.conf file allowguest defaults to yes.  This means that 
anyone that can reach the SIP ports on that system has access to make 
unauthenticated calls, by default.  The administrator actually has to go 
in and turn it off to prevent unauthenticated SIP calls (in whatever 
context [general] points at).

Does anyone else agree with me that this is a poor default?  I'd like to 
see the default setting changed.

It seems to me that this default is the reason behind the 
doc/security.txt bias against using the "default" context for toll calls.

Thanks,

Lee.

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[asterisk-users] AST_CONFIG, MEETME_INFO and meetme.conf

[Olivier]

Hello,

To make my dialplan more robust, I thought I wouldn't include any
meetme-specific rules and I would exlusively rely on meetme.conf data.

For each dialed number, I would check if this number is used as a conference
room number in meetme.conf.

When I'm trying to implement this, I can see that :
1. AST_CONFIG is not convenient to parse lines like "conf=>1234", as several
lines are present and AST_CONFIG is dedicated to  key-value pairs.
2. MEETME_INFO is focused on live conferences (though issue 15450 extends
this behaviour).

How would you work around this without using Realtime ?

Regards
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Re: [asterisk-users] Incoming Call Ring

[Leif Neland]

  - Original Message - 
  From: Dan Journo 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Thursday, November 12, 2009 1:24 PM
  Subject: [asterisk-users] Incoming Call Ring


  Hello,

   

  I have Asterisk set up with 6 extensions. When a call comes in, I use a Dial 
command to call all the extensions together until someone picks up.

   

  The problem is, when there is an incoming call and an extension is in use, if 
the extension puts down the phone while the incoming call is still ringing, 
that extension doesn't ring. This is because when the Dial command was 
executed, that extension was busy.

   

  Is there any way to make that extension ring as soon as its available if 
there is still an incoming call?

   

You could put all 6 phones in a queue, and call that instead. But there will 
still be a delay before Asterisk calls the phone  again.



You could put the phones in a pickup-group, and the user could pick up the 
call, default is *8



Leif


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[asterisk-users] BLF with SPA941?

[Leif Neland]

Appearently SPA941 is less than a SPA942 without two ports, poe and backlight.

There is less features too, it doesn't support BLF.

Is it possible to hack 942-software into 941, or is there another workaround?

Leif
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[asterisk-users] Termination Question

[B.Masoud @ SH]

Hello,

I would like to know how the following scenario works:

 

I have 3 Asterisk servers, A,B & C,  each one is located in a different
country.

Asterisk A is the main one, and both B & C are connected to it.

 

My question is, when a call is originated from B to C, it will have to go
through A, but does A makes a peer connection between B & C to eliminate
bandwidth and latency, or the call has to go through A ???

 

Thanks.

 

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Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters

[Olivier]

2009/11/11 Tzafrir Cohen 

> On Wed, Nov 11, 2009 at 08:24:53PM +0100, Olivier wrote:
> > 2009/11/10 Tzafrir Cohen 
> >
> > > On Tue, Nov 10, 2009 at 06:06:12PM +0100, Olivier wrote:
> > > > Hello,
> > > >
> > > > 1. How can specify in /etc/dahdi/genconf_parameters file that a port
> from
> > > a
> > > > B410P board is to be "disabled".
> > >
> > > There's currently no way to do that.
> > >
> > > It should be trivial to implment. The more difficult part of it would
> be
> > > how to define exactly what spans / channels to disable.
> > >
> > > But why do you need that?
> > >
> >
> > I don't really know why I thought I needed that feature but as some
> gateways
> > implement this feature (the ability to enable or disable each port), I
> must
> > have told myself I may have missed here.
> >
> > On a general point of view, as most Dahdi cards have a light showing
> nearby
> > port status, it should ideally possible, to turn this light off when a
> port
> > is disabled.
> >
> > But I must also add it doesn't seem very important to me to have this
> > implemented.
>
> dahdi_genconf is an optional tool. Ideally it should need no
> configuration at all and generate configuration that Just Works (though
> the fact that it can do that indicates that the current defaults are
> broken).
>
> It should not be another configuration layer. If the configuration it
> has generated is not good enough, you can also manually edit it.
>
> >
> >
> > > > Playing with comments (see bellow) doesn't help : file
> > > > /etc/asterisk/dahdi-channels.conf is filled with 4 ports data.
> > > >
> > > > pri_termtype
> > > > SPAN/1  TE
> > > > SPAN/2  TE
> > > > SPAN/3  TE
> > > > #   SPAN/4  TE
> > >
> > > Currently pri_termtype is the only directive in dahdi_genconf that uses
> > > this "list" syntax. I'm not very happy with it.
> > >
> > > I'm not exactly sure if there should be some sort of generic way of
> > > adding per-span (span? channel? how do you define a span?) definitions.
> > > Think of ssh_config.
> > >
> >
> >
> > What about adding per-span section headers like Asterisk .conf files ?
> > [span1]
> > group_lines 1
> > pri_termtype
> > SPAN/1  TE
> > SPAN/2  TE
> >
> > [span2]
> > group_lines 2
> > pri_termtype
> > SPAN/2  TE
>
> This implies you will know span numbers in advance. I would like better
> ways to specify configuration.
>

Really ?
I used this [span1] header as an example. Using any other string would be
fine for me as what matters, if I'm not mistaken, is the group_lines number
:

[foo]
group_lines 1
pri_termtype
 SPAN/1  TE
 SPAN/2  TE

[bar]
group_lines 2


>
> >
> >
> > I don't think we need to define any further what a span is, beside that
> > rules that applied to the whole genconf_parameters ("no more than 1
> > group_lines statement") should apply to each section.
>
> --
>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] Incoming Call Ring

[Dan Journo]

Hello,

 

I have Asterisk set up with 6 extensions. When a call comes in, I use a
Dial command to call all the extensions together until someone picks up.

 

The problem is, when there is an incoming call and an extension is in
use, if the extension puts down the phone while the incoming call is
still ringing, that extension doesn't ring. This is because when the
Dial command was executed, that extension was busy.

 

Is there any way to make that extension ring as soon as its available if
there is still an incoming call?

 

Thanks

Dan

 

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[asterisk-users] Scheduling destruction of SIP dialog

[Mindaugas Kezys]

Hello,

I got situation which is unclear for me, hope somebody could explain this.

A calls to B

INVITE sent from A to B
B responds with 100 Trying
B responds with 183 Progress
After 10 seconds: Asterisk CLI: Scheduling destruction of SIP dialog '..' in
32000 ms (Method: INVITE)
Asterisk sends CANCEL _instantly_
B responds with 200 OK and 487 Request Terminated
Asterisk confirms 102 ACK
CLI: Really destroying SIP dialog '..' Method: INVITE
Call terminates

Asterisk version 1.4.18.1

Total call duration: 11s

Timeout on call to B is set to 60 seconds:
'SIP/0277027277...@prov7|60|S(7197)'

Call log is here: http://pastebin.ca/1667975



Why Asterisk decided to terminate the call?



Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions




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Re: [asterisk-users] SIP source address error

[Jaap Winius]

Quoting Matt Riddell :

>> [Nov 11 14:29:47] WARNING[6365]: chan_sip.c:1787 __sip_xmit:  
>> sip_xmit of 0xb63d5694 (len 444) to 192.168.8.30:5060 returned -1:  
>> Operation not permitted
>
> Are you binding to an address that the box doesn't own?
>
> Check the top of sip.conf.

It's set to bind to 0.0.0.0, which IIRC is nothing strange.

The question remains: how can a remote Asterisk server be receiving  
SIP packets that still contain the private net IP address of a client?


Jaap

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Re: [asterisk-users] Need Adapter/Gateway with PSTN-interface

[jonas kellens]

I've read (through google) that the Linksys SPA-products do not have
good voice quality on the PSTN-line.

Grandstream HT486 is also just lifeline and EOL.

The only I come up with is Patton-gateways but these are not at all
cheap !

Jonas.

On Thu, 2009-11-12 at 10:13 +, Steve Howes wrote:

> On 12 Nov 2009, at 09:33, jonas kellens wrote:
> 
> > I am looking for a gateway/ATA that can take conversations on the  
> > analogue line (PSTN) and send them to the Asterisk server on the  
> > private network.
> >
> > I was experimenting with the Atcom AG-188N but the "FXO"-port only  
> > supports lifeline, so it's not a real FXO-port that can send  
> > incoming calls to my private Asterisk-server.
> >
> > Could someone advice on a gateway that can take analogue calls and  
> > transfer them on my local network ?!
> >
> > I know about the Digium-cards. Are there alternatives ?
> 
> Google could tell you this Try the Linksys/Sipura type products
> 
> S


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Re: [asterisk-users] Need Adapter/Gateway with PSTN-interface

[Geoff Lane]

On Thursday, November 12, 2009, jonas kellens wrote:

> Could someone advice on a gateway that can take analogue calls and
> transfer them on my local network ?!

FWIW, I've had a few recommendations for the Linksys SPA3000. However,
I haven't tried this for myself yet since I'm still in the planning
stage of replacing my current Asterisk machine. In my case, I
currently have a full-size tower and I'm planning to move to a
mini-itx machine that doesn't have a PCI slot for my TDM400 card.

HTH,

-- 
Geoff


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Re: [asterisk-users] Need Adapter/Gateway with PSTN-interface

[Steve Howes]

On 12 Nov 2009, at 09:33, jonas kellens wrote:

> I am looking for a gateway/ATA that can take conversations on the  
> analogue line (PSTN) and send them to the Asterisk server on the  
> private network.
>
> I was experimenting with the Atcom AG-188N but the "FXO"-port only  
> supports lifeline, so it's not a real FXO-port that can send  
> incoming calls to my private Asterisk-server.
>
> Could someone advice on a gateway that can take analogue calls and  
> transfer them on my local network ?!
>
> I know about the Digium-cards. Are there alternatives ?

Google could tell you this Try the Linksys/Sipura type products

S

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[asterisk-users] Cisco 7970 SIP endless ringing...?

[ml01]

Anyone know what would cause an endless ringing situation?

I have a snom360 and cisco 7970 (sip 8.5.3). I have an incoming trunk
which dials both phones:

[gp710]
exten => _[*1-9].,1,Dial(SIP/li...@cisco7970&SIP/li...@snom360,60)
exten => _[*1-9].,n,Hangup

If a call comes in, I can answer the call on the cisco no problem. 
However if I answer on the snom360, the cisco never stops ringing.

-Dan

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Re: [asterisk-users] SendText

[Tarek Sawah]

i have my own SMS provider as we sell SMS .. so i have setup my call center 
with SMS sending for several services and alerts like a Missed Call when i'm 
not registered it will send me an sms to alert me.
it's pretty the same as Matt discribed.. you call an AGI which may use cURL to 
hit the HTTP API

--
AHD Tarek Sawah

Integrated Digital Systems

CCNA, MCSE, RHCE, VoIP

Syria: +963 944 618286

USA: +1 347 562 2308






Date: Mon, 9 Nov 2009 22:19:08 -0500
From: thomas.per...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SendText

Will text messages work to non-SIP enpoints using your logic/code?
thank you


On Mon, Nov 9, 2009 at 8:59 PM, Matt Riddell  wrote:




On 10/11/09 12:58 PM, Thomas Perron wrote:
> Does anyone have any success with sending a text message from
> extensions.conf
> to an PSTN endpoint such as a cell phone?
>
> If so, kindly send configuration for this part.  I am working on an IVR

> and want
> callers to get a text message at a particular part of the call, after
> dialing a defined character (such as 22).

We use clickatel.

Basically we use the PHP API and call it via an AGI which sends texts.


Therefore the extensions.conf is pretty sparse:

exten => s,1,Read(destination)
exten => s,2,AGI(agi://127.0.0.1/send_sms.php)

Pseudo code for send_sms is:


1. Read AGI variables
2. Get destination variable
3. Include clickatel API file
4. call send_sms function

We also provide an API from our telephone exchanges, but to be fair
you're likely better off just using clickatel yourself :D





--
Cheers,

Matt Riddell
Director
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http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)


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[asterisk-users] Need Adapter/Gateway with PSTN-interface

[jonas kellens]

I am looking for a gateway/ATA that can take conversations on the
analogue line (PSTN) and send them to the Asterisk server on the private
network.

I was experimenting with the Atcom AG-188N but the "FXO"-port only
supports lifeline, so it's not a real FXO-port that can send incoming
calls to my private Asterisk-server.

Could someone advice on a gateway that can take analogue calls and
transfer them on my local network ?!

I know about the Digium-cards. Are there alternatives ?

Jonas.
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