Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

[George Joseph]

On Wed, Jun 6, 2018 at 1:51 AM Olivier  wrote:

>
>
> 2018-06-05 20:29 GMT+02:00 George Joseph :
>
>>
>>
>> On Tue, Jun 5, 2018 at 10:59 AM Olivier  wrote:
>>
>>>
>>>
>>> 2018-06-05 15:27 GMT+02:00 George Joseph :
>>> Thank  you very much, George for replying.
>>>


 On Tue, Jun 5, 2018 at 3:35 AM Olivier  wrote:

> Hi,
>
> After a long discussion with a friend, I would like to ask here:
>
> 1.According SIP RFCs, is possible/recommended to have different values
> in From and P-Asserted-Id fields ?
> For instance, From field showing 123456789 and P-Asserted-Id showing
> 987654321 (beside privacy considerations) ?
>

 Possible? yes absolutely.

>>>
>>> How would you then configure both headers, respectively ?
>>>
>>> From memory, in previous testings, whenever  CALLERID was set to
>>> WHATEVER, P-Asserted-Id was also set to WHATEVER and vice versa, so that I
>>> inferred from this that P-Asserted-Id was meant for Privacy
>>> considerations and nothing else (see [1])
>>>
>>
>> PAI should be used to indicate the calling party's identification
>> regardless of privacy concerns.  In the dialplan you can use the CALLERPRES
>> function to control privacy on a call by call basis.
>>
>>
>>
> I'm sorry but I still have a doubt ...
> Let me re-phrase my question:
>
> My setup is:
> Asterisk <--- PJSIP ---> Bob
>
> For a reason, I want Bob's phone to receive a call with the following
> headers:
>
> From: "Foo" ;tag=as75ee8c7c
> P-Asserted-Id: "Foo" >;whatever
>
> My dialplan is:
> same = n,Set(CALLERID(num)=999)
> XXX
> same = n,Dial(PJSIP/123456@bob)
>

> What shall I replace XXX with to allow me to set 8 in the user part of
> P-Asserted-Id URI (see example above) ?
>

You don't need anything for XXX.  Just the "Set(CALLERID(num)=999)" should
do it.


> CALLERPRES would change From or P-Asserted-Id but not having different
> user parts in URI, would it ?
>

CALLERPRES affects From and Contact but doesn't affect PAI at all.  It does
add a Privacy header though.
BTW, you should use CALLERID(pres) instead of CALLERPRES.


>
> To my knowledge, a possible way to implement what I'm after is to "turn
> off" P-Asserted-Id feature, add a custom P-Asserted-Id header with
> PJSIP_HEADER.
> Am I missing something ?
>

Either that or I am. :)   The example you have above should work.



>
>
>
>>
>>
>>>
>>>
>>> [1]
>>> https://www.voip-info.org/p-asserted-identity-and-remote-party-id-header/
>>>
>>>
 Recommended? who knows?  Implementations are all over the place.  I've
 always thought of the From header as identifying the user agent making the
 request which kinda agrees with RFC3261.   The PAI header should contain
 the identity of the original caller.


>
> 2. When Bob forwards to Cory a call coming from Alice, would expect
> Diversion/History-Info header to include Alice's number ?
>

 No.  The diversion header shows who the diverter is.
 https://tools.ietf.org/html/rfc5806

>>>
>
> Thank for  this reference: I think I confused diverter/caller/callee roles
> when I first read this document.
>
> So, if Bob forwards to Cory a call from Alice, in which headers would you
> expect Alice and Bob numbers to respectively appear ?
>
>
Well, if you just have 3 user agents without asterisk in the middle
Alice sends INVITE to Bob.
Bob returns a 302 to Alice with Cory as the Contact and Bob as the Diversion
Alice sends an ACK to Bob.
Alice sends a new INVITE to Cory.

If Asterisk is in the middle then...
Alice sends INVITE to Bob via Asterisk
Asterisk sends INVITE to Bob with Alice in From/PAI
Bob returns a 302 to Asterisk with Cory in Contact and Bob in Diversion
Asterisk returns a 181 "Call is being forwarded" to Alice.
Asterisk goes back to the dialplan to find Cory.
Asterisk sends an INVITE to Cory with Alice in From/PAI and Bob in
Diversion.
When Cory answers, Asterisk sends back a 200 OK to Alice with Cory in PAI
and Bob in Diversion


>
>
>
>>
>> Best regards
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>

 --
 George Joseph
 Digium, Inc. | Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 Check us out at: www.digium.com & www.asterisk.org


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

 Check out the new Asterisk community forum at:
 https://community.asterisk.org/

 New to Asterisk? Start here:
>

Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

[Olivier]

2018-06-05 20:29 GMT+02:00 George Joseph :

>
>
> On Tue, Jun 5, 2018 at 10:59 AM Olivier  wrote:
>
>>
>>
>> 2018-06-05 15:27 GMT+02:00 George Joseph :
>> Thank  you very much, George for replying.
>>
>>>
>>>
>>> On Tue, Jun 5, 2018 at 3:35 AM Olivier  wrote:
>>>
 Hi,

 After a long discussion with a friend, I would like to ask here:

 1.According SIP RFCs, is possible/recommended to have different values
 in From and P-Asserted-Id fields ?
 For instance, From field showing 123456789 and P-Asserted-Id showing
 987654321 (beside privacy considerations) ?

>>>
>>> Possible? yes absolutely.
>>>
>>
>> How would you then configure both headers, respectively ?
>>
>> From memory, in previous testings, whenever  CALLERID was set to
>> WHATEVER, P-Asserted-Id was also set to WHATEVER and vice versa, so that I
>> inferred from this that P-Asserted-Id was meant for Privacy
>> considerations and nothing else (see [1])
>>
>
> PAI should be used to indicate the calling party's identification
> regardless of privacy concerns.  In the dialplan you can use the CALLERPRES
> function to control privacy on a call by call basis.
>
>
>
I'm sorry but I still have a doubt ...
Let me re-phrase my question:

My setup is:
Asterisk <--- PJSIP ---> Bob

For a reason, I want Bob's phone to receive a call with the following
headers:

From: "Foo" ;tag=as75ee8c7c
P-Asserted-Id: "Foo" >;whatever

My dialplan is:
same = n,Set(CALLERID(num)=999)
XXX
same = n,Dial(PJSIP/123456@bob)

What shall I replace XXX with to allow me to set 8 in the user part of
P-Asserted-Id URI (see example above) ?
CALLERPRES would change From or P-Asserted-Id but not having different user
parts in URI, would it ?

To my knowledge, a possible way to implement what I'm after is to "turn
off" P-Asserted-Id feature, add a custom P-Asserted-Id header with
PJSIP_HEADER.
Am I missing something ?



>
>
>>
>>
>> [1] https://www.voip-info.org/p-asserted-identity-and-remote-par
>> ty-id-header/
>>
>>
>>> Recommended? who knows?  Implementations are all over the place.  I've
>>> always thought of the From header as identifying the user agent making the
>>> request which kinda agrees with RFC3261.   The PAI header should contain
>>> the identity of the original caller.
>>>
>>>

 2. When Bob forwards to Cory a call coming from Alice, would expect
 Diversion/History-Info header to include Alice's number ?

>>>
>>> No.  The diversion header shows who the diverter is.
>>> https://tools.ietf.org/html/rfc5806
>>>
>>

Thank for  this reference: I think I confused diverter/caller/callee roles
when I first read this document.

So, if Bob forwards to Cory a call from Alice, in which headers would you
expect Alice and Bob numbers to respectively appear ?




>
> Best regards
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


>>>
>>> --
>>> George Joseph
>>> Digium, Inc. | Software Developer
>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>>> Check us out at: www.digium.com & www.asterisk.org
>>>
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> George Joseph
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or u

Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

[George Joseph]

On Tue, Jun 5, 2018 at 10:59 AM Olivier  wrote:

>
>
> 2018-06-05 15:27 GMT+02:00 George Joseph :
> Thank  you very much, George for replying.
>
>>
>>
>> On Tue, Jun 5, 2018 at 3:35 AM Olivier  wrote:
>>
>>> Hi,
>>>
>>> After a long discussion with a friend, I would like to ask here:
>>>
>>> 1.According SIP RFCs, is possible/recommended to have different values
>>> in From and P-Asserted-Id fields ?
>>> For instance, From field showing 123456789 and P-Asserted-Id showing
>>> 987654321 (beside privacy considerations) ?
>>>
>>
>> Possible? yes absolutely.
>>
>
> How would you then configure both headers, respectively ?
>
> From memory, in previous testings, whenever  CALLERID was set to WHATEVER,
> P-Asserted-Id was also set to WHATEVER and vice versa, so that I inferred
> from this that P-Asserted-Id was meant for Privacy considerations and
> nothing else (see [1])
>

PAI should be used to indicate the calling party's identification
regardless of privacy concerns.  In the dialplan you can use the CALLERPRES
function to control privacy on a call by call basis.




>
>
> [1]
> https://www.voip-info.org/p-asserted-identity-and-remote-party-id-header/
>
>
>> Recommended? who knows?  Implementations are all over the place.  I've
>> always thought of the From header as identifying the user agent making the
>> request which kinda agrees with RFC3261.   The PAI header should contain
>> the identity of the original caller.
>>
>>
>>>
>>> 2. When Bob forwards to Cory a call coming from Alice, would expect
>>> Diversion/History-Info header to include Alice's number ?
>>>
>>
>> No.  The diversion header shows who the diverter is.
>> https://tools.ietf.org/html/rfc5806
>>
>>
>>
>>
>>>
>>> Best regards
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>> --
>> George Joseph
>> Digium, Inc. | Software Developer
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>> Check us out at: www.digium.com & www.asterisk.org
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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_
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Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

[Olivier]

2018-06-05 15:27 GMT+02:00 George Joseph :
Thank  you very much, George for replying.

>
>
> On Tue, Jun 5, 2018 at 3:35 AM Olivier  wrote:
>
>> Hi,
>>
>> After a long discussion with a friend, I would like to ask here:
>>
>> 1.According SIP RFCs, is possible/recommended to have different values in
>> From and P-Asserted-Id fields ?
>> For instance, From field showing 123456789 and P-Asserted-Id showing
>> 987654321 (beside privacy considerations) ?
>>
>
> Possible? yes absolutely.
>

How would you then configure both headers, respectively ?

>From memory, in previous testings, whenever  CALLERID was set to WHATEVER,
P-Asserted-Id was also set to WHATEVER and vice versa, so that I inferred
from this that P-Asserted-Id was meant for Privacy considerations and
nothing else (see [1])


[1]
https://www.voip-info.org/p-asserted-identity-and-remote-party-id-header/


> Recommended? who knows?  Implementations are all over the place.  I've
> always thought of the From header as identifying the user agent making the
> request which kinda agrees with RFC3261.   The PAI header should contain
> the identity of the original caller.
>
>
>>
>> 2. When Bob forwards to Cory a call coming from Alice, would expect
>> Diversion/History-Info header to include Alice's number ?
>>
>
> No.  The diversion header shows who the diverter is.
> https://tools.ietf.org/html/rfc5806
>
>
>
>
>>
>> Best regards
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> George Joseph
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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_
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Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

[George Joseph]

On Tue, Jun 5, 2018 at 3:35 AM Olivier  wrote:

> Hi,
>
> After a long discussion with a friend, I would like to ask here:
>
> 1.According SIP RFCs, is possible/recommended to have different values in
> From and P-Asserted-Id fields ?
> For instance, From field showing 123456789 and P-Asserted-Id showing
> 987654321 (beside privacy considerations) ?
>

Possible? yes absolutely.  Recommended? who knows?  Implementations are all
over the place.  I've always thought of the From header as identifying the
user agent making the request which kinda agrees with RFC3261.   The PAI
header should contain the identity of the original caller.


>
> 2. When Bob forwards to Cory a call coming from Alice, would expect
> Diversion/History-Info header to include Alice's number ?
>

No.  The diversion header shows who the diverter is.
https://tools.ietf.org/html/rfc5806




>
> Best regards
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

[Daniel Tryba]

On Tue, Jun 05, 2018 at 11:34:51AM +0200, Olivier wrote:
> 1.According SIP RFCs, is possible/recommended to have different values in
> From and P-Asserted-Id fields ?
> For instance, From field showing 123456789 and P-Asserted-Id showing
> 987654321 (beside privacy considerations) ?

Yes, most obviuos need for PAI is a call where anonimity is desired by
caller. Set the From to anonymous@anonymous.invalid and PAI to a real
user if the destination is trusted, any proxy that handles this message
that doesn't trust a destination will strip PAI thus ensuring privacy.
 
> 2. When Bob forwards to Cory a call coming from Alice, would expect
> Diversion/History-Info header to include Alice's number ?

No, diversion/history should contain Bob.

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Re: [asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general

[JM or AJS]

On 29/03/17 16:18, Olivier wrote:

Hello,

After reading [1] (in french), I would be very happy if I could get 
answers to:


1. Does this 13.7+20161113-3 package version has any relation with 
asterisk's version it complements ? Current asterisk version in repo 
is 13.14.0. Does this 13.7 complies with it ?
Debian's versioning scheme is all their own.  And I would not expect it 
to work with anything but a Debian-packaged Asterisk.


Stretch is currently the "testing" distribution.  This means that new 
versions of packages could appear at any time; but if a newly-introduced 
package breaks any other packages, they will be removed from "testing"  
(and replaced as soon as possible with newer, compatible versions)  
rather than allow packages to exist in the repository that cannot be 
co-installed.


If you really want to use a newer Asterisk version, the Debian source 
will contain a file called "rules", which is really a Makefile "in 
disguise".  This should give you a good clue as to how to hand-build an 
equivalent based on more up-to-date Source Code (if the compile-time 
options have not changed too much, then you might even get away with 
using it directly, but consider this a bodge).
2. From package description, is this package enough or not to allow 
transcoding with G711 ?

For instance, in the following situation:
SIP Phone < Opus > Asterisk < G111 ---> ITSP
All codecs can input and output raw, uncompressed PCM; so as long as you 
build all the relevant modules, your Asterisk will be able to transcode 
between any two codecs it supports.


(Is "G111" a typo for "G711" ?)
3. Can you share here any personal field experience with this codec, 
for home worker use case ?

Is there a better user experience with Opus than with G729 or G711 ?
Opus is, to the best of my knowledge, fully Open Source.  G729 was 
encumbered by patents in some jurisdictions, though it's now patent-free.


G.711 A-Law is what the PSTN uses natively, and that is unlikely to 
change anytime soon; though some VoIP providers are bringing Opus online 
already.  If you have many phones connected to your Asterisk, then you 
may run into CPU limitations transcoding incoming and outgoing calls 
between G711 and Opus.  But that depends on your Asterisk server.  If 
you are recording calls, Asterisk will already have to convert both the 
incoming and outgoing legs to raw PCM anyway.  In any case, if your 
provider supports Opus, you can offload the donkey work to them .

4. Does it work on ARM boxes (Raspberry, ...) ?
The only thing that would prevent any software from working on ARM / 
Raspberry Pi would be if it
contained any architecture-specific binary code without Source Code 
(which you could just about get away with, if you released it under LGPL 
plus exceptions or an Apache licence).  And I suspect if any such code 
existed, it would be rewritten in fairly short order anyway.


Also, it's Debian; and they really, really don't like binary blobs, only 
grudgingly banishing them to a special "non-free" section which is not 
even enabled by default.  And that package was in the main repository, 
suggesting full Source Code availability.  In any case, I see builds for 
armhf  (R.Pi 1 and 2)  and arm64  (R.Pi 3);  so even if there is some 
sneaky binary-only component, you will be able to get it to work.


[1] https://packages.debian.org/stretch/asterisk-opus

Regards



--

JM or AJS

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Re: [asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general

[Tzafrir Cohen]

On Wed, Mar 29, 2017 at 06:04:49PM +0200, Olivier wrote:

> Is there any relation between this external patch and the binary mentioned
> in [2]
> [2] http://blogs.digium.com/2016/09/30/opus-in-asterisk/
> 
> The later one mentions a binary-only distribution to comply with legal
> constraints.

No, it is not. This package is in Debian's main archive, which tells you
it is not based on any binary blob.

Opus is widely implemented in software, including free software
(Firefox, Chromium, Linphone, Jitsi and a host of others). See also
https://en.wikipedia.org/wiki/Opus_(audio_format)#Software

My understanding is that to Digium's best legal advice, there are still
patent issues with the Opus codec. Even though many others disagree (as
evident from above) and I also happen to disagree. But I certainly am
not the one who runs Digium. And the powers that be there probably
decided that whatever patent issues there are, have merit and need to be
mitigated.

-- 
   Tzafrir Cohen
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com

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Re: [asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general

[Olivier]

2017-03-29 17:28 GMT+02:00 Tzafrir Cohen :

> On Wed, Mar 29, 2017 at 05:18:18PM +0200, Olivier wrote:
> > Hello,
> >
> > After reading [1] (in french), I would be very happy if I could get
> answers
> > to:
> >
> > 1. Does this 13.7+20161113-3 package version has any relation with
> > asterisk's version it complements ? Current asterisk version in repo is
> > 13.14.0. Does this 13.7 complies with it ?
>
> The opus codec was used as an external patch. It looked ugly and thus a
> separate package was preffered.
>

Thank you very much for this informative answer.

Is there any relation between this external patch and the binary mentioned
in [2]
[2] http://blogs.digium.com/2016/09/30/opus-in-asterisk/

The later one mentions a binary-only distribution to comply with legal
constraints.
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Re: [asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general

[Tzafrir Cohen]

On Wed, Mar 29, 2017 at 05:18:18PM +0200, Olivier wrote:
> Hello,
> 
> After reading [1] (in french), I would be very happy if I could get answers
> to:
> 
> 1. Does this 13.7+20161113-3 package version has any relation with
> asterisk's version it complements ? Current asterisk version in repo is
> 13.14.0. Does this 13.7 complies with it ?

The opus codec was used as an external patch. It looked ugly and thus a
separate package was preffered.

Its version number is not directly related to Asterisk. It has
originally been split from the Debian packaging of Asterisk, and
starting from the same version number allowed easier upgrading. There is
no version number for the upstream code (the patch).

> 
> 2. From package description, is this package enough or not to allow
> transcoding with G711 ?
> For instance, in the following situation:
> SIP Phone < Opus > Asterisk < G111 ---> ITSP

Technically Asterisk codecs translate to/from (typically) linear and
Asterisk combines codecs to do whatever transcoding needed. So the codec
does not transcode directly to G.711. But Asterisk can transcode between
opus and G.711.

> 
> 3. Can you share here any personal field experience with this codec, for
> home worker use case ?
> Is there a better user experience with Opus than with G729 or G711 ?
> 
> 4. Does it work on ARM boxes (Raspberry, ...) ?

Should work just the same.

> 
> 
> [1] https://packages.debian.org/stretch/asterisk-opus


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Re: [asterisk-users] Questions... connecting Asterisk to the World

[A J Stiles]

On Saturday 14 May 2016, Stefan Becker wrote:
> Greetings,
> 
> asterisk list and community,
> 
> I have a problem in how our telefon switch (Siemens HiCOM)
> "talks" with my new configured Asterisk server (V.11.18.0)
> 
> without my Asterisks server in the middle
> 
>  <--> Siemens HiCOM <-ISDN-> NTBA <-...-> PBX Telekom
> 
> A phone connected to the switch requests an "Outgoing" line
> by dialing "0". The party is connected via ISDN to
> the carrier (deutsche Telekom) where the party preceeds
> to dial numbers... and the call is connected
> 
> What I can see while I am dialing is that with every
> digit I press it is being displayed on my phone.
> Further more, these digits are being processed by the
> carrier. The call goes through, rings, immediately on
> completion on the number or is rejected if busy.
> 
> 
> WITH my Asterisks server in the middle of the exchange...
> 
> A phone connected to the switch requests an "Outgoing" line
> by dialing "0". -->  Asterisks recieves incoming call on "s".
> The dialed digits are collected. The dial plan is
> executed accordingly but the "caller" recieves no
> more information about the dialed number. The number is
> not placed in the "dialed" numbers simple functions like
> "redial" do not work anymore.
> 
> Does anybody know what I am doing wrong here. Is there a
> way to teach asterisk to behave exactly as if it were the
> PBX (deutsche Telekom).
> So, as to say, act in a way that NO ONE will rightly know
> the differance between having asterisk taking over the
> function of the ISDN PBX.
> 
> What do I need?  A better dial plan to somehow better simulate
> the way the switch normaly behaves?
> Is hardware the problem?
> 
> 
> My ISDN card in the server is:
> "QuadBRI ISDN Digium Wildcard b410P"
> 
> Most everything else functionly works. incoming and outgoing calls
> from and to ISDN, VoIP and other equipment work fine.
> 
> Just that the phones and switch don't recieve the "collected"
> number sequence the was dialed.
> 
> Any help or ideas anyone might have would be greatly appreciated.


Your problem is that you are still thinking in terms of old-fashioned, clicky-
clicky mechanical telephone exchanges.  Instead of "dialling 0 to request an 
outside line", you need to let Asterisk accept all the digits and then 
determine for itself whether the call is going to be an inside or outside one.

- If the user dials 3 digits  (or however long your internal numbers are),  
treat it as an internal number.
- If the user dials 6 digits  (or however long numbers are on your local 
exchange),  treat it as an external, local number.
- If the user dials 11 digits starting with 0  (or however long a number is in 
your country, including the STD code),  treat it as an external, STD number.
- If the number dials 9 or more digits starting with 00, treat it as an 
external, IDD number.


It will make your dialplan a little more complicated; but if it is too simple, 
you won't be taking full advantage of the power of Asterisk.

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] Questions... connecting Asterisk to the World

[Steve Edwards]

On Sat, 14 May 2016, Stefan Becker wrote:


On Sat, 14 May 2016, Steve Edwards wrote:

I think you need to make the outbound dial a single 'transaction' 
either by using an extension pattern that includes the 0 like 
'055' to dial 555-555- or eliminate the 0 (and the idiom of 
'requesting an outgoing line') and detect an internal vs external call 
via extension pattern matching.


this is the dialplan that I use:

[ReceiveCallOut]
exten = s,1,Read(LOKAL1,5)
 same =   n,Dial(SIP/${LOKAL}@tt)
 same =   n,Hangup()


I would:

) Drop the 'dial 0' anachronism.

) Not use read().

) Use extension pattern matching.

For example, in the US, I would have something like (off the top of my head):

; external, local
exten = _nxx,1, verbose(1,[${EXTEN}@${CONTEXT})
same = n,   goto(dial-local,${EXTEN},1)
same = n,   hangup()

; external, domestic
exten = _nxxnxx,1,  verbose(1,[${EXTEN}@${CONTEXT})
same = n,   goto(${CONTEXT},1${EXTEN},1)
same = n,   hangup()

; external, domestic
exten = _1nxxnxx,1, verbose(1,[${EXTEN}@${CONTEXT})
same = n,   goto(dial-domestic,${EXTEN},1)
same = n,   hangup()

; international
exten = _011x.,1,   verbose(1,[${EXTEN}@${CONTEXT})
same = n,   goto(dial-international,${EXTEN},1)
same = n,   hangup()

; internal
exten = _[2-9]xxx,1,verbose(1,[${EXTEN}@${CONTEXT})
same = n,   goto(dial-internal,${EXTEN},1)
same = n,   hangup()


When the user dials "0", the HiCOM ISDN switch immediately
goes "online" to the outgoing ISDN Copper Cable - connected
to ... A)  B)

A) connected to the NTBA in the wall jack to the NTBA phone company...
  the dialing preceeds to continue "offline" no dailtones are heard.
  The call is completed and connects


This sounds weird and very foreign (strange and unfamiliar, not as being a 
characteristic of a different country) to me. So, as a caller, I would 
hear the '0' DTMF but no other tones? No feedback as I press keys?



B) connected to the Asterisk ISDN Card

  Asterisk server reacts by executing the above dial plan...


The dialplan does not reflect your intentions.


  CLI > "answered call from "" to "s"

  The user has an open "answered" line and the dialing are collected by
  listening to the DTMF tones.  The generated dial tones can be heard
  on the phone line.

Somehow the signaling on the line of the outgoing call is differant
when the cable is handeled by the PBX or by asterisk.

But why ?


Dialplan and channel configuration.

Can't asterisk be configured to handle a call exactly as the otherwise 
connected phone company's PBX would?


My guess is yes.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] Questions... connecting Asterisk to the World

[Stefan Becker]

On Sat, 14 May 2016, Steve Edwards wrote:

> I think you need to make the outbound dial a single 'transaction' either by
> using an extension pattern that includes the 0 like '055' to dial
> 555-555- or eliminate the 0 (and the idiom of 'requesting an outgoing
> line') and detect an internal vs external call via extension pattern matching.

this is the dialplan that I use:

[ReceiveCallOut]
 exten = s,1,Read(LOKAL1,5)
  same =   n,Dial(SIP/${LOKAL}@tt)
  same =   n,Hangup()

When the user dials "0", the HiCOM ISDN switch immediately
goes "online" to the outgoing ISDN Copper Cable - connected 
to ... A)  B)

A) connected to the NTBA in the wall jack to the NTBA phone company...
   the dialing preceeds to continue "offline" no dailtones are heard.
   The call is completed and connects
   
B) connected to the Asterisk ISDN Card

   Asterisk server reacts by executing the above dial plan...

   CLI > "answered call from "" to "s"

   The user has an open "answered" line and the dialing are collected by
   listening to the DTMF tones.  The generated dial tones can be heard
   on the phone line.


Somehow the signaling on the line of the outgoing call is differant
when the cable is handeled by the PBX or by asterisk.

But why ?

Can't asterisk be configured to handle a call exactly as the 
otherwise connected phone company's PBX would?


thanks for listening,

Stefan

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Re: [asterisk-users] Questions... connecting Asterisk to the World

[Steve Edwards]

On Sat, 14 May 2016, Stefan Becker wrote:

A phone connected to the switch requests an "Outgoing" line by dialing 
"0". --> Asterisks recieves incoming call on "s". The dialed digits are 
collected. The dial plan is executed accordingly but the "caller" 
recieves no more information about the dialed number. The number is not 
placed in the "dialed" numbers simple functions like "redial" do not 
work anymore.


This is not my area of expertise, but I'll throw my $0.02 in...

When you 'request an outgoing line' by dialing 0, that call leg is 
processed by Asterisk, thus, that is what the phone 'sees' as the dialed 
number and that's what the phone will send when 'redial' is pressed.


I think you need to make the outbound dial a single 'transaction' either 
by using an extension pattern that includes the 0 like '055' to 
dial 555-555- or eliminate the 0 (and the idiom of 'requesting an 
outgoing line') and detect an internal vs external call via extension 
pattern matching.


Does your internal extension numbering plan conflict with external 
national numbering plan?


Dialing a prefix digit seems so 1970s to me.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] Questions regarding ICE and STUN with Asterisk

[Kirill Marchuk]

Well, after a more specific research I came to 2 conclusions:

1) no need to specify "stunaddr" option in Asterisk configs in this 
case, as we know that host definitely has a public IP


2) as of the inclusion of all local IP-addresses as candidates, this is 
(apparently) done in "rtp_add_candidates_to_ice" function of 
res_rtp_asterisk.c, where it has a code:

/* Add all the local interface IP addresses */
..

And as fas as I can tell from basic ICE overview [1], this should NOT 
prevent proper session functioning, as long as candidate pairs 
(local/remote) are checked for connectivity first.


Still, I would think it to be useful, to have an option to EXCLUDE local 
IP-addresses from using as local candidates.


What does the community think on this ?

Thanks

Kirill Marchuk

[1] https://trac.pjsip.org/repos/wiki/Using_Standalone_ICE

18.03.2016 14:37, Kirill Marchuk пишет:

Hi everyone

 I would like to get some help and clarification from the experienced 
ones, upon the following:


- we're using Asterisk 13.7.0, that is deployed on a host, that has a 
public IP *and* a couple of gray IPs (192.168.x.x & 10.10.x.x)


- we're using WebRTC web-page (jsSIP) as a client

Which is the proper setup of ICE/STUN related config (on the Asterisk 
and on the client) for the most reliable work in most cases ?


For example, now we're trying to use our own STUN server (from 
Debian's "stund" package), whose documentation says "You have to have 
2 different public IPs on the same server in order to run STUN server"


Is it really so? and what are the implications of using it with only 
one IP (which is possible, at least it runs seemingly well without that)


On the client side, we've configured jsSIP.UA to use our own STUN 
server via "pcConfig" object


On Asterisk, we have icesupport=yes both in sip.conf and rtp.conf. 
We've also enabled stunaddr=stun.l.google.com:19302 in rtp.conf.


Is it proper solution for this case ?

When I inspect SIP packets, I see that there are ICE candidates in 
both offers and answers. BUT: SDP bodies in the packets from server to 
client contain "gray" IPs of the Asterisk host:


a=ice-ufrag:636c49c84158d2b45840291c6724c0f9
a=ice-pwd:6b012c01092ec01275964eaa55a8784b
a=candidate:H904cc6da 1 UDP 2130706431 144.76.x.y 51604 typ host
a=candidate:Ha0a0202 1 UDP 2130706431 10.10.2.2 51604 typ host
a=candidate:S904cc6da 1 UDP 1694498815 144.76.x.y 51604 typ srflx 
raddr 144.76.x.y rport 51604

a=candidate:H904cc6da 2 UDP 2130706430 144.76.x.y 51605 typ host
a=candidate:Ha0a0202 2 UDP 2130706430 10.10.2.2 51605 typ host
a=candidate:S904cc6da 2 UDP 1694498814 144.76.x.y 51605 typ srflx 
raddr 144.76.x.y rport 51605


I am afraid it might be a potential problem, when a client will have 
his private IP in similar subnets. Or am I wrong here ?


So far we are not experiencing any issues, but this seems to be 
alarming..


Can this behaviour (namely, which IP addresses does Asterisk include 
into SDPs body) be configured somehow ?


Many thanks for any help with this question..

Kirill Marchuk



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Re: [asterisk-users] Questions on musiconhold.conf custom mode

[Olivier]

2014-10-27 11:50 GMT+01:00 Thorsten Göllner :
>
> Am 27.10.2014 08:54, schrieb Olivier:
>> 2014-10-25 19:33 GMT+02:00 Thorsten Göllner :
>>> Am 25.10.2014 00:09, schrieb Olivier:
 Hello,

 I need to play some musiconhold content starting at a random duration
 from the start.

 Thanks to mode=custom option and either madplay or mpg123 programs, I
 could successfully get what I was after on a Debian Wheezy system.

 Now I realized sox version on my target system (Debian Squeeze) cannot
 convert to MP3 format.
 So I'm looking after workarounds.

 0. I've read many  mpg123 or madplay examples. All of them are
 clutered with option converting MP3 input file into an appropriate
 format that Asterisk requires for music on hold.
 What is the name of this appropriate format ? sln ? wav ?

 1. Is there a player like mpg123, that can repeat content in
 appropriate format (see above)  to stdout but can read from anything
 different from MP3 ?

 2. Is there an option on Squeeze to convert audio files to MP3
 (reverse coversion works OK).

 3. Which options could I have for such custom MOH, if I was building
 on system without g729 transaltion capabilites ans with g729-only SIP
 trunks or phones ?

>>> Is the gsm-format an option for you? So you may convert your moh-File to
>>> gsm:
>>> sox YouWavFile.wav -r 8000 -c1 MohFile.gsm
>> Hi Thorsten,
>>
>> Yes gsm-format is an option for me but how can you play such gsm file as MOH 
>> ?
>>
>> If I'm not mistaken, both madplay or mpg123 would only play MP3 files
>> (I've not tested with other formats, yet).
>> I could successfully play a RAW file with cat but cat has no repeat
>> option, so I still have to find something else anyway.
>
> When your musiconhold.conf looks like that ...
>
>  cut -
> [general]
>
> [default]
> mode=files
> directory=moh
>
> [your_moh_class]
> mode=files
> directory=/your/path/to/your/moh/files
>  cut -
>
Yes this is true but when you need your MOH to start randomly (ie not
start from the very start but from anywhere within your MOH file), you
need to switch to custom mode and customize application parameter.

In this specific case, I didn't find many options avoiding MP3 files.

> ... then you can put any supported file format into the specified
> directory. GSM is only one option. Asterisk will take the best (meaning
> "cheapest") file format availble in this directory.
>
>
>>
>>
>>> If you really need mp3 you have to compile sox with mp3-support by
>>> yourself OR maybe this is a solution on Debian:
>>> http://www.howtoinstall.co/en/debian/wheezy/main/libsox-fmt-mp3/
>> Yes, you're correct.
>> I'll suggest my customer a Wheezy upgrade.
>
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Re: [asterisk-users] Questions on musiconhold.conf custom mode

[Thorsten Göllner]

Am 27.10.2014 08:54, schrieb Olivier:
> 2014-10-25 19:33 GMT+02:00 Thorsten Göllner :
>> Am 25.10.2014 00:09, schrieb Olivier:
>>> Hello,
>>>
>>> I need to play some musiconhold content starting at a random duration
>>> from the start.
>>>
>>> Thanks to mode=custom option and either madplay or mpg123 programs, I
>>> could successfully get what I was after on a Debian Wheezy system.
>>>
>>> Now I realized sox version on my target system (Debian Squeeze) cannot
>>> convert to MP3 format.
>>> So I'm looking after workarounds.
>>>
>>> 0. I've read many  mpg123 or madplay examples. All of them are
>>> clutered with option converting MP3 input file into an appropriate
>>> format that Asterisk requires for music on hold.
>>> What is the name of this appropriate format ? sln ? wav ?
>>>
>>> 1. Is there a player like mpg123, that can repeat content in
>>> appropriate format (see above)  to stdout but can read from anything
>>> different from MP3 ?
>>>
>>> 2. Is there an option on Squeeze to convert audio files to MP3
>>> (reverse coversion works OK).
>>>
>>> 3. Which options could I have for such custom MOH, if I was building
>>> on system without g729 transaltion capabilites ans with g729-only SIP
>>> trunks or phones ?
>>>
>> Is the gsm-format an option for you? So you may convert your moh-File to
>> gsm:
>> sox YouWavFile.wav -r 8000 -c1 MohFile.gsm
> Hi Thorsten,
>
> Yes gsm-format is an option for me but how can you play such gsm file as MOH ?
>
> If I'm not mistaken, both madplay or mpg123 would only play MP3 files
> (I've not tested with other formats, yet).
> I could successfully play a RAW file with cat but cat has no repeat
> option, so I still have to find something else anyway.

When your musiconhold.conf looks like that ...

 cut -
[general]

[default]
mode=files
directory=moh

[your_moh_class]
mode=files
directory=/your/path/to/your/moh/files
 cut -

... then you can put any supported file format into the specified
directory. GSM is only one option. Asterisk will take the best (meaning
"cheapest") file format availble in this directory.


>
>
>> If you really need mp3 you have to compile sox with mp3-support by
>> yourself OR maybe this is a solution on Debian:
>> http://www.howtoinstall.co/en/debian/wheezy/main/libsox-fmt-mp3/
> Yes, you're correct.
> I'll suggest my customer a Wheezy upgrade.

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Re: [asterisk-users] Questions on musiconhold.conf custom mode

[Olivier]

2014-10-25 19:33 GMT+02:00 Thorsten Göllner :
>
> Am 25.10.2014 00:09, schrieb Olivier:
>> Hello,
>>
>> I need to play some musiconhold content starting at a random duration
>> from the start.
>>
>> Thanks to mode=custom option and either madplay or mpg123 programs, I
>> could successfully get what I was after on a Debian Wheezy system.
>>
>> Now I realized sox version on my target system (Debian Squeeze) cannot
>> convert to MP3 format.
>> So I'm looking after workarounds.
>>
>> 0. I've read many  mpg123 or madplay examples. All of them are
>> clutered with option converting MP3 input file into an appropriate
>> format that Asterisk requires for music on hold.
>> What is the name of this appropriate format ? sln ? wav ?
>>
>> 1. Is there a player like mpg123, that can repeat content in
>> appropriate format (see above)  to stdout but can read from anything
>> different from MP3 ?
>>
>> 2. Is there an option on Squeeze to convert audio files to MP3
>> (reverse coversion works OK).
>>
>> 3. Which options could I have for such custom MOH, if I was building
>> on system without g729 transaltion capabilites ans with g729-only SIP
>> trunks or phones ?
>>
>
> Is the gsm-format an option for you? So you may convert your moh-File to
> gsm:
> sox YouWavFile.wav -r 8000 -c1 MohFile.gsm

Hi Thorsten,

Yes gsm-format is an option for me but how can you play such gsm file as MOH ?

If I'm not mistaken, both madplay or mpg123 would only play MP3 files
(I've not tested with other formats, yet).
I could successfully play a RAW file with cat but cat has no repeat
option, so I still have to find something else anyway.


>
> If you really need mp3 you have to compile sox with mp3-support by
> yourself OR maybe this is a solution on Debian:
> http://www.howtoinstall.co/en/debian/wheezy/main/libsox-fmt-mp3/

Yes, you're correct.
I'll suggest my customer a Wheezy upgrade.


>
>
> -Thorsten-
>
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Re: [asterisk-users] Questions on musiconhold.conf custom mode

[Thorsten Göllner]

Am 25.10.2014 00:09, schrieb Olivier:
> Hello,
>
> I need to play some musiconhold content starting at a random duration
> from the start.
>
> Thanks to mode=custom option and either madplay or mpg123 programs, I
> could successfully get what I was after on a Debian Wheezy system.
>
> Now I realized sox version on my target system (Debian Squeeze) cannot
> convert to MP3 format.
> So I'm looking after workarounds.
>
> 0. I've read many  mpg123 or madplay examples. All of them are
> clutered with option converting MP3 input file into an appropriate
> format that Asterisk requires for music on hold.
> What is the name of this appropriate format ? sln ? wav ?
>
> 1. Is there a player like mpg123, that can repeat content in
> appropriate format (see above)  to stdout but can read from anything
> different from MP3 ?
>
> 2. Is there an option on Squeeze to convert audio files to MP3
> (reverse coversion works OK).
>
> 3. Which options could I have for such custom MOH, if I was building
> on system without g729 transaltion capabilites ans with g729-only SIP
> trunks or phones ?
>

Is the gsm-format an option for you? So you may convert your moh-File to
gsm:
sox YouWavFile.wav -r 8000 -c1 MohFile.gsm

If you really need mp3 you have to compile sox with mp3-support by
yourself OR maybe this is a solution on Debian:
http://www.howtoinstall.co/en/debian/wheezy/main/libsox-fmt-mp3/


-Thorsten-

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Re: [asterisk-users] Questions about chan_dahdi, PRI, MWI (and Q.SIG)

[Richard Mudgett]

On Fri, Jun 28, 2013 at 3:59 AM, Jens Bürger  wrote:

> Hello everyone,
>
> My setup:
> Debian squeeze
> Asterisk 1.8, DAHDI, libpri, compiled from source
> TE110P, attached to a Deutsche Telekom Octopus E Modell 300/800
>
> I'm trying to get MWI for Voicemail working. In the same server I have
> also got an Eicon DIVA PRI card for testing purposes (it is integrated via
> CAPI and the chan-capi channel driver into my Asterisk). MWI works just
> fine there.
>
> I read through chan_dahdi.conf and have some questions:
>
> 1. The documentation of
> mwi_mailboxes
> says: "You can give a comma separated list of up to 8 mailboxes per span."
>
> Is this constraint really existing? How am I supposed to use the MWI
> feature in even a semi-professional environment? My PBX is used in a
> non-commercial project, but I have connected about 50 phones to my PBX,
> though, and it is interconnected with two further PBXs which would also
> need voicemail with MWI...
>

This option is intended for BRI spans communicating to a group of ISDN
phones connected to that span and the Asterisk server is handling voicemail
itself.


> 2. How can I set the MWI origin number?
>
> 3. Are there any debug possibilities for MWI?
>

Other than the limited support for sending MWI messages to ISDN phones on
BRI lines there is no other ISDN MWI support.

Richard
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Re: [asterisk-users] Questions about sRTP

[Matthew Jordan]

On Thu, Jun 20, 2013 at 5:10 PM, Mike Diehl  wrote:

>
>
> On Thu, Jun 20, 2013 at 2:05 PM, Joshua Colp  wrote:
>
>> Mike Diehl wrote:
>>
>>> Hi all,
>>>
>>> I'm getting ready to setup SIP/TLS and SRTP.  But I have a few
>>> questions.  The first one is that I was reading an article at:
>>>
>>> https://supportforums.cisco.com/docs/DOC-15381
>>>
>>> That indicated that Asterisk doesn't support TLS as an OPTIONAL
>>> transport.  It's either all or nothing.  Specifically, this is what it
>>> said:
>>>
>>
>> Your statement is incorrect. Asterisk supports TLS as an optional
>> signaling transport (although if you do SDES SRTP without it then someone
>> can snoop on your keys and ultimately decrypt your media).
>>
>> What it does not support is optional *SRTP*. If a device requests SRTP
>> and it's not possible, the call will fail.
>>
>>
> So then, is it safe to say that Asterisk will ALLOW a secure phone call,
> but the client hast to REQUEST it?
>
> I understand that requesting SRTP without SIP/TLS is evil; I just
> misunderstood what I was reading.
>
> I'm also thinking that the AGI script I use to route calls can check if
> either leg of a call comes from or goes to port 5061 and play a sound file
> to indicate that the cal is 'secure.'  Does this seem reasonable?
>
>
You can query a channel using the CHANNEL function (
https://wiki.asterisk.org/wiki/display/AST/Function_CHANNEL) to see if the
channel currently supports secure communication, and you can request that
the outbound channel be made secure using the same function.

An example of doing this is on the wiki:

https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Specifics

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Re: [asterisk-users] Questions about sRTP

[Mike Diehl]

On Thu, Jun 20, 2013 at 2:05 PM, Joshua Colp  wrote:

> Mike Diehl wrote:
>
>> Hi all,
>>
>> I'm getting ready to setup SIP/TLS and SRTP.  But I have a few
>> questions.  The first one is that I was reading an article at:
>>
>> https://supportforums.cisco.com/docs/DOC-15381
>>
>> That indicated that Asterisk doesn't support TLS as an OPTIONAL
>> transport.  It's either all or nothing.  Specifically, this is what it
>> said:
>>
>
> Your statement is incorrect. Asterisk supports TLS as an optional
> signaling transport (although if you do SDES SRTP without it then someone
> can snoop on your keys and ultimately decrypt your media).
>
> What it does not support is optional *SRTP*. If a device requests SRTP and
> it's not possible, the call will fail.
>
>
So then, is it safe to say that Asterisk will ALLOW a secure phone call,
but the client hast to REQUEST it?

I understand that requesting SRTP without SIP/TLS is evil; I just
misunderstood what I was reading.

I'm also thinking that the AGI script I use to route calls can check if
either leg of a call comes from or goes to port 5061 and play a sound file
to indicate that the cal is 'secure.'  Does this seem reasonable?

Thanks,

Mike.
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Re: [asterisk-users] Questions about sRTP

[Joshua Colp]

Mike Diehl wrote:

Hi all,

I'm getting ready to setup SIP/TLS and SRTP.  But I have a few
questions.  The first one is that I was reading an article at:

https://supportforums.cisco.com/docs/DOC-15381

That indicated that Asterisk doesn't support TLS as an OPTIONAL
transport.  It's either all or nothing.  Specifically, this is what it said:


Your statement is incorrect. Asterisk supports TLS as an optional 
signaling transport (although if you do SDES SRTP without it then 
someone can snoop on your keys and ultimately decrypt your media).


What it does not support is optional *SRTP*. If a device requests SRTP 
and it's not possible, the call will fail.


--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org

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Re: [asterisk-users] Questions about extension.conf

[Ron Wheeler]

Excellent.
It appears that Getting Started has a lot more stuff in it than the 
documentation for 1.8.


Very helpful.

Ron

On 29/11/2012 12:31 PM, David M. Lee wrote:


On Nov 29, 2012, at 11:18 AM, Ron Wheeler wrote:


That is a good answer.
Thanks.
Any reason why it is not documented?


It's documented on the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Contexts,+Extensions,+and+Priorities


Ron


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www.asterisk.org 





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Re: [asterisk-users] Questions about extension.conf

[David M. Lee]

On Nov 29, 2012, at 11:18 AM, Ron Wheeler wrote:

> That is a good answer.
> Thanks.
> Any reason why it is not documented?

It's documented on the Asterisk wiki:
  
https://wiki.asterisk.org/wiki/display/AST/Contexts,+Extensions,+and+Priorities

> Ron

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Re: [asterisk-users] Questions about extension.conf

[Eric Wieling]

The Wiki is (always) out of date.  You might consider taking a look at 
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#DialplanBasics_id262049
 which is likely less out of data.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler
Sent: Thursday, November 29, 2012 12:17 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Questions about extension.conf

On 29/11/2012 11:47 AM, Salman Zafar wrote:


It is self explanatory, for example:

exten =>  _X.,1, Noop("Let say we have allowed all numbers i.e. _X 
means and . specifies any range")
same => n,NoOp("Here we have skipped mentioning dial-pattern again and 
thats it")


Hope I have answered your question. 


Not for me.
What part of those lines and comments discusses "same"?

What is the syntax for a "same" line? what does it mean to use "same" rather 
than "exten"?






On Thu, Nov 29, 2012 at 8:40 AM, Shitian Long  
wrote:


Hello  

I have been reading the sample extension.conf

;###


[outbound-freenum2]
; This is the handler which performs the dialing logic. It is 
called
; from the [outbound-freenum] context
;
exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)})
; make sure the suffix is all digits as well
same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != 
"${SUFFIX}"]?fn-CONGESTION,1)

; filter out bad characters per the README-SERIOUSLY.best-practices.txt 
document
same => n,Set(TIMEOUT(absolute)=10800)
same => 
n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) ; perform our 
lookup with freenum.org
same => n,GotoIf($["${isnresult}" != ""]?from)
same => n,Set(DIALSTATUS=CONGESTION)
same => n,Goto(fn-CONGESTION,1)
same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial)   
; check if we set the FREENUMDOMAIN global variable in [global]
same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) 
;if we did set it, then we'll use it for our outbound dialing domain
same => n(dial),Dial(SIP/${isnresult},40)
same => n,Goto(fn-${DIALSTATUS},1)

exten => fn-BUSY,1,Busy()

exten => _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
same => n,Congestion()

;##


According to 
"http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf";

"Syntax for defining a context: keywords exten, include, 
ignorepat and switch." "same is not mentioned in this wiki." 

There is a part of dial plan from sample extension.conf above. 
My Question is  how "same =>" key word works . 

Thanks



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-- 
Regards 


**
Muhammad Salman
***


 

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Re: [asterisk-users] Questions about extension.conf

[Ron Wheeler]

That is a good answer.
Thanks.
Any reason why it is not documented?

Ron

On 29/11/2012 11:52 AM, Mikhail Lischuk wrote:


Shitian Long wrote 29.11.2012 18:40:

There is a part of dial plan from sample extension.conf above. My 
Question is  how "same =>" key word works .

Thanks
  


"same" is used for complex templates, if you don't want to copy 
previous line or afraid you can make a typo.


exten => _1XXNXXX,1,Answer

same => n,HangUp

is the substitution for:

exten => _1XXNXXX,1,Answer

exten => _1XXNXXX,n,HangUp

Also,  it makes grepping the particular exten in a file a lot easier, 
and if you want to change some template for exten which has 50 lines, 
you don't have to edit all 50 of them.


--
With Best Regards
Mikhail Lischuk  

  



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Re: [asterisk-users] Questions about extension.conf

[Ron Wheeler]

On 29/11/2012 11:47 AM, Salman Zafar wrote:

It is self explanatory, for example:

exten =>  _X.,1, Noop("Let say we have allowed all numbers i.e. _X 
means and . specifies any range")
same => n,NoOp("Here we have skipped mentioning dial-pattern again and 
thats it")



Hope I have answered your question.

Not for me.
What part of those lines and comments discusses "same"?

What is the syntax for a "same" line? what does it mean to use "same" 
rather than "exten"?






On Thu, Nov 29, 2012 at 8:40 AM, Shitian Long > wrote:


Hello

I have been reading the sample extension.conf

;###


[outbound-freenum2]
; This is the handler which performs the dialing logic. It is called
; from the [outbound-freenum] context
;
exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)})  ;
make sure the suffix is all digits as well
same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" !=
"${SUFFIX}"]?fn-CONGESTION,1)
  ; filter out bad characters per the
README-SERIOUSLY.best-practices.txt document
same => n,Set(TIMEOUT(absolute)=10800)
same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org
)}) ; perform our lookup with freenum.org

same => n,GotoIf($["${isnresult}" != ""]?from)
same => n,Set(DIALSTATUS=CONGESTION)
same => n,Goto(fn-CONGESTION,1)
same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial)
  ; check if we set the FREENUMDOMAIN global variable in [global]

same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ;
   if we did set it, then we'll use it for our outbound dialing domain
same => n(dial),Dial(SIP/${isnresult},40)
same => n,Goto(fn-${DIALSTATUS},1)

exten => fn-BUSY,1,Busy()

exten => _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
same => n,Congestion()

;##


According to
"http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf";

"Syntax for defining a context: keywords *exten*, *include*,
*ignorepat* and *switch*." "same is not mentioned in this wiki."

There is a part of dial plan from sample extension.conf above. My
Question is  how "same =>" key word works .

Thanks



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**
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Re: [asterisk-users] Questions about extension.conf

[Mikhail Lischuk]

 

Shitian Long wrote 29.11.2012 18:40: 

> There is a part of dial
plan from sample extension.conf above. My Question is how "same =>" key
word works . 
> 
> Thanks

"same" is used for complex templates, if you
don't want to copy previous line or afraid you can make a typo. 

exten
=> _1XXNXXX,1,Answer 

same => n,HangUp 

is the substitution for:


exten => _1XXNXXX,1,Answer 

exten => _1XXNXXX,n,HangUp 

Also, it
makes grepping the particular exten in a file a lot easier, and if you
want to change some template for exten which has 50 lines, you don't
have to edit all 50 of them. 

-- 
With Best Regards
Mikhail Lischuk

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Re: [asterisk-users] Questions about extension.conf

[Danny Nicholas]

As I understand it, same => is a way to shorthand your list of the other
keywords. In the example you posted, you save 4 keystrokes for each line you
enter; not a lot of savings for this short example, but put it in a 1000+
line dialplan and it's quite a time-saver.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shitian Long
Sent: Thursday, November 29, 2012 10:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Questions about extension.conf

 

Hello 

 

I have been reading the sample extension.conf

 

;###

 

 

[outbound-freenum2]

; This is the handler which performs the dialing logic. It is called

; from the [outbound-freenum] context

;

exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})

same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)});
make sure the suffix is all digits as well

same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" !=
"${SUFFIX}"]?fn-CONGESTION,1)

;
filter out bad characters per the README-SERIOUSLY.best-practices.txt
document

same => n,Set(TIMEOUT(absolute)=10800)

same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) ;
perform our lookup with freenum.org

same => n,GotoIf($["${isnresult}" != ""]?from)

same => n,Set(DIALSTATUS=CONGESTION)

same => n,Goto(fn-CONGESTION,1)

same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})

same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial)   ;
check if we set the FREENUMDOMAIN global variable in [global]

same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ;
if we did set it, then we'll use it for our outbound dialing domain

same => n(dial),Dial(SIP/${isnresult},40)

same => n,Goto(fn-${DIALSTATUS},1)

 

exten => fn-BUSY,1,Busy()

 

exten => _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})

same => n,Congestion()

 

;##

 

 

According to
"http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf";

 

"Syntax for defining a context: keywords exten, include, ignorepat and
switch." "same is not mentioned in this wiki." 

 

There is a part of dial plan from sample extension.conf above. My Question
is  how "same =>" key word works . 

 

Thanks

 

 

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Re: [asterisk-users] Questions on converting to ConfBridge

[Leif Madsen]

On 02/10/12 06:07 PM, Richard Kenner wrote:

I'm looking at what would be involved in converting from MeetMe to
ConfBridge and there seems to be a lot of missing administrative things,
but I hope I'm just missing it.

>

There also doesn't
seem to be a way to lock conferences or mute or kick out users from
the dialplan.

What am I missing?


You're missing the custom DTMF based menus in confbridge.conf, which 
allows you to set menus separately for admins and users of the 
conference bridge. This menu allows you to control kicking, muting, etc 
of users within the conference bridge.


No need to manipulate from the dialplan anymore.


--
Leif Madsen
http://www.oreilly.com/catalog/asterisk

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Re: [asterisk-users] Questions on converting to ConfBridge

[kenner]

> Why are you wanting to use CLI commands instead of AMI? The available 
> AMI actions for ConfBridge can do listing/locking/muting/kicking etc as 
> you want.

Because I can't easily manually do an AMI command, but instead have to
write code to do it.  It's important to me to be able to clean up things
from the command-line if something is stuck or broken.

> As for dialplan applications to do the various things - what are you 
> trying to achieve using them?

And IVR application that people can call into and manipulate people in
conference rooms.  Note that this depends on dialplan commands *and*
having a number index for them.  It's unclear how I'd do this with
confbridge.

Here's the dialplan I'm using.

exten => 210/_[12]XX,1,NoOp ; Valid if internal.
exten => 210,s,Gosub(Authenticate,s,1()); Else authenticate.
 same => n,Mset(C=conference&ha/room&digits/2&digits/0,E=adacore/not-exist)
 same => n,Mset(STATS_INC(conf_mgr)=1,__G=conf_op) ; Count the usage.
 same => n(r),Macro(Get-Speech,${G},${EFN}adacore/conf_mgr,2,10,100,w)
 same => n,GotoIf(${S_T}?${S_T},1:r); Retry or do action.

exten => _[lLK]20Z,1,GotoIf(${MEETME_INFO(parties,20${EXTEN:-1})}?:err)
exten => _L20Z,n,Mset(V=lock,T=locked,E=is&adacore/already-locked)
exten => _l20Z,s,Mset(V=unlock,T=unlocked,E=is&adacore/already-unlocked)
exten => _K20Z,s,Mset(V=terminate,T=terminated) ; To terminate.
exten => 
_[lLK]20Z,n,Set(CFN=adacore/you-want-to&adacore/${V}&${C}&digits/${EXTEN:-1}&)
exten => _[lLK]20Z,n,Gosub(Is-That-Correct,s,1) ; ... and see if correct.
exten => _[lLK]20Z,n,GotoIf($[${GOSUB_RETVAL}=2]?210,r) ; Retry it not.
exten => _L20Z,n,GotoIf(${MEETME_INFO(lock,20${EXTEN:-1})}?err) ; Bad status.
exten => _l20Z,s,GotoIf(${MEETME_INFO(lock,20${EXTEN:-1})}?:err) ; Likewise.
exten => _K20Z,s,NoOp   ; No test needed for termination.
exten => _[lLK]20Z,n,MeetMeAdmin(${EXTEN:-3},${EXTEN:0:1}) ; Perform op.
exten => _[lLK]20Z,n,Set(EFN=${C}&digits/${EXTEN:-1}&is&now&adacore/${T}&)
exten => _[lLK]20Z,n,Goto(210,r); Ask for another operation.

exten => _[lLK]20Z,n(err),Set(EFN=im-sorry&${C}&digits/${EXTEN:-1}&${E}&)
exten => _[lLK]20Z,n,Goto(210,r); See if another operation is wanted.

exten => _s20Z,1,Goto(s20${EXTEN:-1}${MEETME_INFO(parties,${EXTEN:-3})},1)
exten => _s20Z.,1,Playback(${C}&digits/${EXTEN:3:1}) ; Say that conference ...
exten => _s20Z0,n,Playback(adacore/not-exist)   ; ... doesn't exist,
exten => _s20Z1,s,Swift(has one participant); ... or has one person,
exten => _s20Z.,s,Swift(has ${EXTEN:4} participants) ; ... or more.
exten => _s20Z.,n,ExecIf(${MEETME_INFO(lock,${EXTEN:1})}?Swift(and is locked)
exten => _s20Z.,n,Set(M=$[CEIL(MEETME_INFO(activity,${EXTEN:1:3})/60)])
exten => _s20Z[1-9]!,n,Swift(and has been active for ${M} minutes)
exten => _s20Z.,s,NoOp  ; In other cases, do nothing.
exten => _s20Z.,n,Goto(210,r)   ; Go back for another operation.

exten => _j20Z,1,Set(CFN=you-wish-to-join&${C}&digits/${EXTEN:-1}&)
 same => n,Gosub(Is-That-Correct,s,1) ; See if correct.
 same => n,GotoIf($[${GOSUB_RETVAL}=2]?210,r) ; Retry it not.
 same => n,SpeechDestroy; Else free speech channel.
 same => n,Goto(${EXTEN:-3},1)  ; And go there.

exten => 
_[pP]20Z,1,GotoIf($[MEETME_INFO(parties,${EXTEN:1})=0]?s20${EXTEN:-1}0,1)
 same => n,Swift(participants in)   ; Say the header and ...
 same => n,Playback(${C}&digits/${EXTEN:3:1})   ; ... conference number.
 same => n,ExecIf($[x${G:0:3}=xtmp]?System(rm -f ${GRAMS}/${G}.gram))
 same => n,Set(__G=tmp/r${RAND(1,9)})   ; Grammar filename part.
 same => n,AGI(conflist.php,${EXTEN:1},${GRAMS}/${G}.gram,${EXTEN:0:1})
 same => n,Goto(210,r)  ; And go back.

exten => _m20ZXX.,1,Mset(Q=adacore/unmute,OP=adacore/unmuted)
exten => _M20ZXX.,s,Mset(Q=adacore/mute,OP=adacore/muted)
exten => _k20ZXX.,s,Mset(Q=adacore/remove,OP=removed)
exten => _[Mmk]20ZXX.,n,Playback(adacore/you-want-to&${Q}) ; Start question.
 same => n,Swift(${EXTEN:6}); Say who ...
 same => n,Gosub(Is-That-Correct,s,1)   ; ... and see if correct.
 same => n,GotoIf($[${GOSUB_RETVAL}=2]?210,r)   ; Retry it not.
 same => n,Set(U=${IF($[${EXTEN:4:1}=0]?${EXTEN:5:1}:${EXTEN:4:2})})
 same => n,MeetMeAdmin(${EXTEN:1:3},${EXTEN:0:1},${U}) ; Do operation.
 same => n,Swift(${EXTEN:6}); Say name ...
 same => n,Playback(${OP})  ; ... and what we did.
 same => n,Goto(210,r)  ; Go back for another try.

exten => What,1,Set(EFN=adacore/confop_what&) ; Say what options are available.
 same => n,Goto(210,r)  ; And go back and prompt again.
exten => Done,1,Playback(vm-goodbye); Here to hangup.

Here's the grammar:

#ABNF 1.0 UTF-8;
language en-US;
mode voice;
tag-format ;
root $conf_op;

$Operation = lock {out = "L";}
| unlock {out = "l";}
| (end | kill | terminate) {out

Re: [asterisk-users] Questions on converting to ConfBridge

[Joshua Colp]

Hola,

Richard Kenner wrote:


I'm looking at what would be involved in converting from MeetMe to
ConfBridge and there seems to be a lot of missing administrative things,
but I hope I'm just missing it.  We all know about the missing realtime
linkage.  That's a major nuisance, but can be worked around.


I think you are only the second or third person to really even mention 
realtime in the context of ConfBridge in the places I see.



More serious is that the CLI command to display users in a ConfBridge
don't show the caller ID information, so it becomes very hard to
have web applications that show who's in a conference.  There also doesn't
seem to be a way to lock conferences or mute or kick out users from
the dialplan.  And the CLI command needs a channel, not a user index,
making scripting via the dialplan that much harder.


Why are you wanting to use CLI commands instead of AMI? The available 
AMI actions for ConfBridge can do listing/locking/muting/kicking etc as 
you want.


As for dialplan applications to do the various things - what are you 
trying to achieve using them?


Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org

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Re: [asterisk-users] Questions on hardware or software-based echo cancellation

[Kevin P. Fleming]

On 01/13/2012 02:12 AM, Olivier wrote:

2012/1/12, Kevin P. Fleming:

On 01/12/2012 06:39 AM, Olivier wrote:

Hi,

I'm having some questions related to echo cancellation configuration
on a Digium board enabled systems (B410P, TE420, TE420B, ) for
cases when a hardware ech canceller is present or not.

I read in TEXXX manual that when setting echocancel=yes in
chan_dahdi.conf on a VPMOCT64-equiped system, 128ms hardware echo
cancellation was enabled.

1. I'm correct thinking that it is then impossible to switch from
hardware to software echo can without removing the VPMOCT64 module
itself ?
2. Does the same also apply to HA8 and its VPMOCT032 module ?


With DAHDI 2.6 (and possibly 2.5), it is possible to override the
configuration and apply a software echo canceller to a channel even if
it has a hardware one. With prior versions, yes, the echo cancellation
module would have to be physically removed (or disabled using a
parameter to the kernel module).


Then, maybe a line mentioning that in the next User Manual edition
would be perfect.


Sure, but you have to understand that the user manuals for our board 
products are typically only updated when the board itself gets changed; 
we don't actually deliver DAHDI with the boards, and 
installation/configuration instructions for DAHDI are primarily included 
in the manual for user convenience. Users should be aware that they 
could easily be out of date, and not include all options that are 
currently offered (although if the user manual's instructions become 
incorrect, we'll update the manual). Regardless, I'll mention this to 
the people who manage those products. Thanks!


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

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Re: [asterisk-users] Questions on hardware or software-based echo cancellation

[Olivier]

2012/1/12, Kevin P. Fleming :
> On 01/12/2012 06:39 AM, Olivier wrote:
>> Hi,
>>
>> I'm having some questions related to echo cancellation configuration
>> on a Digium board enabled systems (B410P, TE420, TE420B, ) for
>> cases when a hardware ech canceller is present or not.
>>
>> I read in TEXXX manual that when setting echocancel=yes in
>> chan_dahdi.conf on a VPMOCT64-equiped system, 128ms hardware echo
>> cancellation was enabled.
>>
>> 1. I'm correct thinking that it is then impossible to switch from
>> hardware to software echo can without removing the VPMOCT64 module
>> itself ?
>> 2. Does the same also apply to HA8 and its VPMOCT032 module ?
>
> With DAHDI 2.6 (and possibly 2.5), it is possible to override the
> configuration and apply a software echo canceller to a channel even if
> it has a hardware one. With prior versions, yes, the echo cancellation
> module would have to be physically removed (or disabled using a
> parameter to the kernel module).

Then, maybe a line mentioning that in the next User Manual edition
would be perfect.
Thanks for replying.

>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
>
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Re: [asterisk-users] Questions on hardware or software-based echo cancellation

[Kevin P. Fleming]

On 01/12/2012 06:39 AM, Olivier wrote:

Hi,

I'm having some questions related to echo cancellation configuration
on a Digium board enabled systems (B410P, TE420, TE420B, ) for
cases when a hardware ech canceller is present or not.

I read in TEXXX manual that when setting echocancel=yes in
chan_dahdi.conf on a VPMOCT64-equiped system, 128ms hardware echo
cancellation was enabled.

1. I'm correct thinking that it is then impossible to switch from
hardware to software echo can without removing the VPMOCT64 module
itself ?
2. Does the same also apply to HA8 and its VPMOCT032 module ?


With DAHDI 2.6 (and possibly 2.5), it is possible to override the 
configuration and apply a software echo canceller to a channel even if 
it has a hardware one. With prior versions, yes, the echo cancellation 
module would have to be physically removed (or disabled using a 
parameter to the kernel module).


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

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Re: [asterisk-users] Questions on IAX client

[Steve Edwards]

On Sun, 23 Oct 2011, asterisk asterisk wrote:

I used to use Zoiper IAX to connect to my asterisk server from remote 
site. On asterisk CLI, I can see my zoiper client registered and stay on 
line. HOwever, I don't know why now I can't call this client. It always 
show up as "Unable to create channel IAX2 (Cause 20 Unknown)


If you enable IAX debugging you may get some clues.

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

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Re: [asterisk-users] Questions on Dahdi

[Dale Noll]

On 10/05/2011 07:28 PM, asterisk asterisk wrote:

I have naive question. I do not have any hardware on my asterisk host.
All I have are either SIP trunk for DID or hardware ATA which bridges
the asterisk to PSTN. Do I need Dahdi install? Do i have ztdummy for
timing issue? I encounter problem in this when I try to install Dahdi
latest but I found it is not running, Instead it runs when service
starts but I can't find its status when I type in service dahdi status.

I am using Asterisk 1.8.7 on centos 5.7 32 bit.

CK


If you have no hardware in the server for PSTN connectivity, then you do 
not need any card drivers.  If you do not use MeetMe, you would not need 
any kernel modules either, but my impression is that it doesn't hurt in 
install at least the dahdi.ko module.


When 'service dahdi start' is run, it loads the dahdi.ko module as well 
as any modules in the /etc/dahdi/modules config file.  If you have no 
hardware, it is safe to comment out all the modules.


If you have no card modules loading, 'service dahdi status' returns 
nothing.  If you use the command 'lsmod | grep dahdi' and you see a line 
similar to the one below the kernel module is loaded.



dahdi 186420  0


If you see that, you are good to go.

Dale

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Re: [asterisk-users] Questions about FMFM with linked servers

[Faisal Hanif]

Did you tried to execute Set(CALLERID(num)=you-required-callerid)?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovey Forman
Sent: Friday, July 29, 2011 1:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Questions about FMFM with linked servers

 

All;

 

In a linked server environment, running Asterisk 1.6 I am noticing that when
a call is placed from server A to server B (via 4 digit extension) and
server B ext has a FMFM to call their mobile, the mobile phone shows the
default caller ID setting on server B instead of the actual caller id of the
person who initiated the call on server A.

 

This scenario, of course, works in the event a call in placed via the PSTN
into Server A (or B) and rings the FMFM extension. In this case, the mobile
phones sees the correct (initial) caller ID on the mobile.

 

Thanks!

 

--Dovey

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Re: [asterisk-users] Questions About Fax for Asterisk

[Steve Totaro]

On Sat, May 8, 2010 at 2:27 PM, David Backeberg wrote:

> On Sat, May 8, 2010 at 7:21 AM, Steve Totaro
>  wrote:
> > Maybe there is a simple setting somewhere, but "RTFM" from Digium tech
> > support when the FM offers no suggestion on how to possibly tweak
> settings
> > for better success.
>
> Do you want to dump some samples from your dialplan?
>
> I know I personally had better success with fax over sip when I put a:
> Playback(silence/1)
> infront of
> ReceiveFax()
>
>
I will post Monday, I have had exten=whatever,1,Answer(3) since day one, I
would guess that does just about the same thing as Playback(silence/1).

Thanks,
Steve Totaro
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Re: [asterisk-users] Questions About Fax for Asterisk

[David Backeberg]

On Sat, May 8, 2010 at 7:21 AM, Steve Totaro
 wrote:
> Maybe there is a simple setting somewhere, but "RTFM" from Digium tech
> support when the FM offers no suggestion on how to possibly tweak settings
> for better success.

Do you want to dump some samples from your dialplan?

I know I personally had better success with fax over sip when I put a:
Playback(silence/1)
infront of
ReceiveFax()

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Re: [asterisk-users] Questions About Fax for Asterisk

[Steve Totaro]

On Fri, May 7, 2010 at 9:56 PM, Steve Underwood  wrote:

> On 05/08/2010 08:15 AM, Steve Totaro wrote:
> >
> >
> > On Fri, May 7, 2010 at 2:01 PM, Martin  > > wrote:
> >
> > On Thu, May 6, 2010 at 3:11 PM, Steve Totaro
> >  > > wrote:
> > > Yes, I purchased licenses for Fax for Asterisk and yes I called
> > tech support
> > > and had the WORST experience I have ever had with any technical
> > support
> > > call.
> > >
> > > I am running Asterisk 1.6.2.6 and:
> > >
> > > FAX For Asterisk Components:
> > > Applications: 1.6.2.0_1.2.0
> > > voipgw01Digium FAX Driver: 1.6.2.0_1.2.0 (optimized for c3_2_32)
> > >
> > > The guy was arrogant and absolutely a jerk and I don't like to
> > call people
> > > names, but call it as I see it.  This has not been my experience
> > the five or
> > > six times I have had to call Digium over the years, but it has
> > been many
> > > years since my last call so I have no idea what the general
> > support staff is
> > > like.
> > >
> > > I could not get any questions answered by the tech that took
> > hours to call
> > > me back to tell me to read the readme.  That would be all well
> > and good if I
> > > didn't pay money.
> > >
> > > He could not explain Digium's math as far as faxing and failed
> > to offer to
> > > get back to me with any kind of answer.
> > >
> > > Maybe someone on the list can make sense of this Enron style of
> > accounting:
> > >
> > > voipgw01*CLI> fax show stats
> > > voipgw01*CLI>
> > > FAX Statistics:
> > > ---
> > >
> > > Current Sessions : 1
> > > Transmit Attempts: 0
> > > Receive Attempts : 336
> > > Completed FAXes  : 320
> > > Failed FAXes : 57
> > >
> > > Digium G.711
> > > Licensed Channels: 4
> > > Max Concurrent   : 1
> > > Success  : 0
> > > Switched to T.38 : 0
> > > Canceled : 0
> > > No FAX   : 1
> > > Partial  : 0
> > > Negotiation Failed   : 0
> > > Train Failure: 3
> > > Protocol Error   : 0
> > > IO Partial   : 0
> > > IO Fail  : 0
> > > voipgw01*CLI>
> > > Digium T.38
> > > Licensed Channels: 4
> > > Max Concurrent   : 4
> > > Success  : 175
> > > Canceled : 0
> > > No FAX   : 6
> > > Partial  : 19
> > > Negotiation Failed   : 0
> > > Train Failure: 83
> > > Protocol Error   : 33
> > > IO Partial   : 0
> > > IO Fail  : 0
> > >
> > > Thanks,
> > > Steve Totaro
> >
> >
> > wow definitely the acc"counting" engine is broken ...
> >
> > I can only make sense of this
> >
> > > Receive Attempts : 336
> > > Completed FAXes  : 320
> > > Failed FAXes : 57
> >
> > 1) your receive app was called 336 times but the fax hanged up before
> > negotiating
> > 2) you had 320 of this completed (partially or fully)
> > 3) but 57 out of 320 failed to transmit entirely
> >
> > 57/320=17.8% which is too high for a commercial product IHMO
> >
> > Martin
> >
> >
> > Considering that this is a direct cross connect from Leve3's cage to
> > my my cage in the same DC at an Equinix facility, 100Mb DIA w/EIPT
> > VoIP service, I would expect nearly 100% success.
> >
> > Considering the circuit was just turned up and there is no data except
> > Level3's phone traffic.  They are our carrier, RespOrg, origination
> > and termination, no 3rd parties, all on net.
> >
> > I could understand if it was a peaked out DIA circuit to some cut rate
> > VoIP provider, but not under "perfect" circumstances.
> >
> > Thanks,
> > Steve Totaro
> Were these all test calls made from a well defined source? It takes
> *two* correctly working FAX terminals to make a successful call. Its
> easy to get a high failure rate for silly reasons. In volume testing of
> spandsp and iaxmodem we had times where a high percentage of calls
> failed, which turned out to be just one rouge machine calling over and
> over again trying to achieve success. On the other hand, failures
> between known good FAX terminals should be far below 1%.
>
> Steve
>
>
These are not test calls.  These are real world calls from a real world.

Since this is Fax for Asterisk, volume is not really an issue, since I only
have four licenses on a brand new CentOS box (HP DL360 G6(or whatever is
currently shipping from HP).

Based on caller ID, it is not one caller inflating the numbers.  Generally,
a failed fax will succeed on a subsequent try.

Previously, we were terminating faxes to a quad port Digium PRI card,
everything from the OS to the hardware were from 2006.

Re: [asterisk-users] Questions About Fax for Asterisk

[Steve Underwood]

On 05/08/2010 08:15 AM, Steve Totaro wrote:
>
>
> On Fri, May 7, 2010 at 2:01 PM, Martin  > wrote:
>
> On Thu, May 6, 2010 at 3:11 PM, Steve Totaro
>  > wrote:
> > Yes, I purchased licenses for Fax for Asterisk and yes I called
> tech support
> > and had the WORST experience I have ever had with any technical
> support
> > call.
> >
> > I am running Asterisk 1.6.2.6 and:
> >
> > FAX For Asterisk Components:
> > Applications: 1.6.2.0_1.2.0
> > voipgw01Digium FAX Driver: 1.6.2.0_1.2.0 (optimized for c3_2_32)
> >
> > The guy was arrogant and absolutely a jerk and I don't like to
> call people
> > names, but call it as I see it.  This has not been my experience
> the five or
> > six times I have had to call Digium over the years, but it has
> been many
> > years since my last call so I have no idea what the general
> support staff is
> > like.
> >
> > I could not get any questions answered by the tech that took
> hours to call
> > me back to tell me to read the readme.  That would be all well
> and good if I
> > didn't pay money.
> >
> > He could not explain Digium's math as far as faxing and failed
> to offer to
> > get back to me with any kind of answer.
> >
> > Maybe someone on the list can make sense of this Enron style of
> accounting:
> >
> > voipgw01*CLI> fax show stats
> > voipgw01*CLI>
> > FAX Statistics:
> > ---
> >
> > Current Sessions : 1
> > Transmit Attempts: 0
> > Receive Attempts : 336
> > Completed FAXes  : 320
> > Failed FAXes : 57
> >
> > Digium G.711
> > Licensed Channels: 4
> > Max Concurrent   : 1
> > Success  : 0
> > Switched to T.38 : 0
> > Canceled : 0
> > No FAX   : 1
> > Partial  : 0
> > Negotiation Failed   : 0
> > Train Failure: 3
> > Protocol Error   : 0
> > IO Partial   : 0
> > IO Fail  : 0
> > voipgw01*CLI>
> > Digium T.38
> > Licensed Channels: 4
> > Max Concurrent   : 4
> > Success  : 175
> > Canceled : 0
> > No FAX   : 6
> > Partial  : 19
> > Negotiation Failed   : 0
> > Train Failure: 83
> > Protocol Error   : 33
> > IO Partial   : 0
> > IO Fail  : 0
> >
> > Thanks,
> > Steve Totaro
>
>
> wow definitely the acc"counting" engine is broken ...
>
> I can only make sense of this
>
> > Receive Attempts : 336
> > Completed FAXes  : 320
> > Failed FAXes : 57
>
> 1) your receive app was called 336 times but the fax hanged up before
> negotiating
> 2) you had 320 of this completed (partially or fully)
> 3) but 57 out of 320 failed to transmit entirely
>
> 57/320=17.8% which is too high for a commercial product IHMO
>
> Martin
>
>
> Considering that this is a direct cross connect from Leve3's cage to 
> my my cage in the same DC at an Equinix facility, 100Mb DIA w/EIPT 
> VoIP service, I would expect nearly 100% success.
>
> Considering the circuit was just turned up and there is no data except 
> Level3's phone traffic.  They are our carrier, RespOrg, origination 
> and termination, no 3rd parties, all on net.
>
> I could understand if it was a peaked out DIA circuit to some cut rate 
> VoIP provider, but not under "perfect" circumstances.
>
> Thanks,
> Steve Totaro
Were these all test calls made from a well defined source? It takes 
*two* correctly working FAX terminals to make a successful call. Its 
easy to get a high failure rate for silly reasons. In volume testing of 
spandsp and iaxmodem we had times where a high percentage of calls 
failed, which turned out to be just one rouge machine calling over and 
over again trying to achieve success. On the other hand, failures 
between known good FAX terminals should be far below 1%.

Steve


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Re: [asterisk-users] Questions About Fax for Asterisk

[Steve Totaro]

On Fri, May 7, 2010 at 2:01 PM, Martin  wrote:

> On Thu, May 6, 2010 at 3:11 PM, Steve Totaro
>  wrote:
> > Yes, I purchased licenses for Fax for Asterisk and yes I called tech
> support
> > and had the WORST experience I have ever had with any technical support
> > call.
> >
> > I am running Asterisk 1.6.2.6 and:
> >
> > FAX For Asterisk Components:
> > Applications: 1.6.2.0_1.2.0
> > voipgw01Digium FAX Driver: 1.6.2.0_1.2.0 (optimized for c3_2_32)
> >
> > The guy was arrogant and absolutely a jerk and I don't like to call
> people
> > names, but call it as I see it.  This has not been my experience the five
> or
> > six times I have had to call Digium over the years, but it has been many
> > years since my last call so I have no idea what the general support staff
> is
> > like.
> >
> > I could not get any questions answered by the tech that took hours to
> call
> > me back to tell me to read the readme.  That would be all well and good
> if I
> > didn't pay money.
> >
> > He could not explain Digium's math as far as faxing and failed to offer
> to
> > get back to me with any kind of answer.
> >
> > Maybe someone on the list can make sense of this Enron style of
> accounting:
> >
> > voipgw01*CLI> fax show stats
> > voipgw01*CLI>
> > FAX Statistics:
> > ---
> >
> > Current Sessions : 1
> > Transmit Attempts: 0
> > Receive Attempts : 336
> > Completed FAXes  : 320
> > Failed FAXes : 57
> >
> > Digium G.711
> > Licensed Channels: 4
> > Max Concurrent   : 1
> > Success  : 0
> > Switched to T.38 : 0
> > Canceled : 0
> > No FAX   : 1
> > Partial  : 0
> > Negotiation Failed   : 0
> > Train Failure: 3
> > Protocol Error   : 0
> > IO Partial   : 0
> > IO Fail  : 0
> > voipgw01*CLI>
> > Digium T.38
> > Licensed Channels: 4
> > Max Concurrent   : 4
> > Success  : 175
> > Canceled : 0
> > No FAX   : 6
> > Partial  : 19
> > Negotiation Failed   : 0
> > Train Failure: 83
> > Protocol Error   : 33
> > IO Partial   : 0
> > IO Fail  : 0
> >
> > Thanks,
> > Steve Totaro
>
>
> wow definitely the acc"counting" engine is broken ...
>
> I can only make sense of this
>
> > Receive Attempts : 336
> > Completed FAXes  : 320
> > Failed FAXes : 57
>
> 1) your receive app was called 336 times but the fax hanged up before
> negotiating
> 2) you had 320 of this completed (partially or fully)
> 3) but 57 out of 320 failed to transmit entirely
>
> 57/320=17.8% which is too high for a commercial product IHMO
>
> Martin
>
>
Considering that this is a direct cross connect from Leve3's cage to my my
cage in the same DC at an Equinix facility, 100Mb DIA w/EIPT VoIP service, I
would expect nearly 100% success.

Considering the circuit was just turned up and there is no data except
Level3's phone traffic.  They are our carrier, RespOrg, origination and
termination, no 3rd parties, all on net.

I could understand if it was a peaked out DIA circuit to some cut rate VoIP
provider, but not under "perfect" circumstances.

Thanks,
Steve Totaro
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Re: [asterisk-users] Questions About Fax for Asterisk

[Martin]

On Thu, May 6, 2010 at 3:11 PM, Steve Totaro
 wrote:
> Yes, I purchased licenses for Fax for Asterisk and yes I called tech support
> and had the WORST experience I have ever had with any technical support
> call.
>
> I am running Asterisk 1.6.2.6 and:
>
> FAX For Asterisk Components:
>     Applications: 1.6.2.0_1.2.0
> voipgw01Digium FAX Driver: 1.6.2.0_1.2.0 (optimized for c3_2_32)
>
> The guy was arrogant and absolutely a jerk and I don't like to call people
> names, but call it as I see it.  This has not been my experience the five or
> six times I have had to call Digium over the years, but it has been many
> years since my last call so I have no idea what the general support staff is
> like.
>
> I could not get any questions answered by the tech that took hours to call
> me back to tell me to read the readme.  That would be all well and good if I
> didn't pay money.
>
> He could not explain Digium's math as far as faxing and failed to offer to
> get back to me with any kind of answer.
>
> Maybe someone on the list can make sense of this Enron style of accounting:
>
> voipgw01*CLI> fax show stats
> voipgw01*CLI>
> FAX Statistics:
> ---
>
> Current Sessions : 1
> Transmit Attempts    : 0
> Receive Attempts : 336
> Completed FAXes  : 320
> Failed FAXes : 57
>
> Digium G.711
> Licensed Channels    : 4
> Max Concurrent   : 1
> Success  : 0
> Switched to T.38 : 0
> Canceled : 0
> No FAX   : 1
> Partial  : 0
> Negotiation Failed   : 0
> Train Failure    : 3
> Protocol Error   : 0
> IO Partial   : 0
> IO Fail  : 0
> voipgw01*CLI>
> Digium T.38
> Licensed Channels    : 4
> Max Concurrent   : 4
> Success  : 175
> Canceled : 0
> No FAX   : 6
> Partial  : 19
> Negotiation Failed   : 0
> Train Failure    : 83
> Protocol Error   : 33
> IO Partial   : 0
> IO Fail  : 0
>
> Thanks,
> Steve Totaro


wow definitely the acc"counting" engine is broken ...

I can only make sense of this

> Receive Attempts : 336
> Completed FAXes  : 320
> Failed FAXes : 57

1) your receive app was called 336 times but the fax hanged up before
negotiating
2) you had 320 of this completed (partially or fully)
3) but 57 out of 320 failed to transmit entirely

57/320=17.8% which is too high for a commercial product IHMO

Martin



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Re: [asterisk-users] Questions about asterisk and spa2102

[Steve Edwards]

On Fri, 29 Jan 2010, Kosa wrote:

> 1.- is it possible to use an spa2102 to make and revice calls from a 
> "normal" phone? I mean, I know I can use it to connect an analog to an 
> asterisk server, but I want to know if it can be used to connect 
> asterisk to the analog phoneline.

The 2102 is an FXS ("station") device. It connects to things like a 
telephone or a fax machine.

The 3102 is an FXS and FXO ("office") device. You can plug in a telephone 
and the wire coming out of the wall. I have the predecessor, the 3000. It 
has the neat feature that if it loses power it will "bridge" the FXS and 
FXO ports so the telephone can still be used. I don't know if current 
models still have this feature.

> 2.- I'm trying to unlock the spa2102 with no succes at the moment, any 
> links or hint will be very appreciated.

I did this many years ago with some PAP2s that were locked to Vonage. 
Definitely not worth the effort it took to configure my name servers to 
pretend they were Vonage's so they could resolve Vonage's names to my 
local IP addresses and setting up the Ethernet interface aliases to 
Vonage's IP addresses.

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

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Re: [asterisk-users] Questions about asterisk and spa2102

[John Novack]

Kosa wrote:
> Hi there! First mail on the list :)
>
> 1.- is it possible to use an spa2102 to make and revice calls from a "normal" 
> phone? I mean, I know I can use it to connect an analog to an asterisk 
> server, but I want to know if it can be used to connect asterisk to the 
> analog phoneline.
>   
In simple terms:

FXS ports provide battery to analog phones, provide ringing to analog 
phones when so instructed, and provide dialtone to analog phones, then 
forward the dialed number as data to a server.
FXO  ports expect to see battery from analog exchange lines, supply a 
loop closure to request service from an exchange, in some cases will 
pulse dial a string of digits, in all cases send a string of DTMF 
digits, and detect a ringing voltage from an exchange, forwarding 
received information as data to a server.

Some devices have both types of connections.

If you want an external device to do both, it will need both types of 
ports, one cannot be both.
> 2.-  I'm trying to unlock the spa2102 with no succes at the moment, any
> links or hint will be very appreciated.
>
>   
Why waste your ( valuable? ) time??
New unlocked similar devices are available from multiple sources.
Many of these are born locked to a specific service, and cannot be changed.


> I'm and absolute newbie on asterisk, btw.
>
>
> Thanx!
>
> Kosa
>
> - Un mundo mejor es posible -
>
>
>   

-- 
Dog is my co-pilot


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Re: [asterisk-users] Questions about static

[Noah Miller]

> We swapped PoE switches, phones, cable and switch ports multiple times.
> What do you mean by local interference? Cell phone? The person swears
> nothing is near the phone.

There are lots of things that can cause interference.  Radios,
elevators, bad electrical wiring, you name it.  Is the static still
there when you move the identical phone elsewhere?  If not, then the
static is most probably caused by some local interference where the
user is.


- Noah

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Re: [asterisk-users] Questions about static

[Don Kelly]

Could the static be in the user's hearing aid?

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovey Forman
Sent: Friday, November 27, 2009 12:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Questions about static

We swapped PoE switches, phones, cable and switch ports multiple times.
What do you mean by local interference? Cell phone? The person swears
nothing is near the phone.

Its very strange.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noah Miller
Sent: Friday, November 27, 2009 1:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Questions about static

> We have swapped out the phone multiple times for the user.
> Only one user.

Bad PoE port on the switch?

How about local interference that the user cannot control?  Does the
same phone experience static when moved elsewhere?

Do you have a power brick for the phone so you can try it as non-PoE?

Is the static consistent or intermittent?


- Noah

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Re: [asterisk-users] Questions about static

[Dovey Forman]

We swapped PoE switches, phones, cable and switch ports multiple times.
What do you mean by local interference? Cell phone? The person swears
nothing is near the phone.

Its very strange.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noah Miller
Sent: Friday, November 27, 2009 1:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Questions about static

> We have swapped out the phone multiple times for the user.
> Only one user.

Bad PoE port on the switch?

How about local interference that the user cannot control?  Does the
same phone experience static when moved elsewhere?

Do you have a power brick for the phone so you can try it as non-PoE?

Is the static consistent or intermittent?


- Noah

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Re: [asterisk-users] Questions about static

[Noah Miller]

> We have swapped out the phone multiple times for the user.
> Only one user.

Bad PoE port on the switch?

How about local interference that the user cannot control?  Does the
same phone experience static when moved elsewhere?

Do you have a power brick for the phone so you can try it as non-PoE?

Is the static consistent or intermittent?


- Noah

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Re: [asterisk-users] Questions about static

[Dovey Forman]

It’s a single user and we have swapped everything.

The phone is an Aastra 6731i and its PoE.



*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Michael Wyres
*Sent:* Wednesday, November 25, 2009 6:27 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Questions about static



Is it a single user?  Or every single phone?



If it’s a single user, and you can get hold of a UPS with power conditioning
on it, try plugging the various devices into it – there might be some dirty
power coming along.





*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dovey Forman
*Sent:* Thursday, 26 November 2009 07:08
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Questions about static



Using an Asterisk system running 1.2 with Aastra phones.

Would be a cause of static for inbound/outbound and ext to ext calls?



Its voip both in and out.



We swapped, phones, cordes, switches etc…..



Typically a reboot of the phone resolves the problem…person also swears
there is nothing on or near their desk to cause interference (microwave,
cell phone is purse).



Strange……



Thanks

--Dovey

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Re: [asterisk-users] Questions about static

[Dovey Forman]

We have swapped out the phone multiple times for the user.
Only one user.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cb
Sent: Wednesday, November 25, 2009 11:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Questions about static

On Nov 25, 2009, at 3:07 PM, Dovey Forman wrote:

> Would be a cause of static for inbound/outbound and ext to ext calls?
>
> Its voip both in and out.
>
> We swapped, phones, cordes, switches etc...
>
> Typically a reboot of the phone resolves the problem.person also
> swears there is nothing on or near their desk to cause interference
> (microwave, cell phone is purse).

Only one user? Did you check to see if it is a bad handset cord?

-chris




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Re: [asterisk-users] Questions about static

[cb]

On Nov 25, 2009, at 3:07 PM, Dovey Forman wrote:

> Would be a cause of static for inbound/outbound and ext to ext calls?
>
> Its voip both in and out.
>
> We swapped, phones, cordes, switches etc…..
>
> Typically a reboot of the phone resolves the problem…person also  
> swears there is nothing on or near their desk to cause interference  
> (microwave, cell phone is purse).

Only one user? Did you check to see if it is a bad handset cord?

-chris




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Re: [asterisk-users] Questions about static

[Michael Wyres]

Is it a single user?  Or every single phone?

If it's a single user, and you can get hold of a UPS with power conditioning on 
it, try plugging the various devices into it - there might be some dirty power 
coming along.


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovey Forman
Sent: Thursday, 26 November 2009 07:08
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Questions about static

Using an Asterisk system running 1.2 with Aastra phones.
Would be a cause of static for inbound/outbound and ext to ext calls?

Its voip both in and out.

We swapped, phones, cordes, switches etc.

Typically a reboot of the phone resolves the problem...person also swears there 
is nothing on or near their desk to cause interference (microwave, cell phone 
is purse).

Strange..

Thanks
--Dovey
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Confidentiality - This email and any attachments are intended for the named 
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all copies and do not use or disclose the contents.

Copyright - This email is subject to copyright and no part of it maybe 
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Re: [asterisk-users] Questions about Voicemail

[Dovey Forman]

Rob;

That would be great. You could send directly to me @ dovey.for...@idt.net
or respond to this list.

I appreciate it!

--Dovey

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert
Lister
Sent: Wednesday, November 25, 2009 11:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Questions about Voicemail

On Mon, 2009-11-23 at 14:46 -0500, Dovey Forman wrote:
> Regarding the email to multiple receipients, it is available on an
ad-hoc
> basis from the phone?
>
> IE; call into the voicemail system, enter x digit to send a voicemail to
> multiple users, record the message, then enter the destination
mailboxes,
> separated by  # sign...

You can enable an option in the voicemail that allows the prompt:

'To send a message to another user'...

sendvoicemail=yes ; Allow the user to compose and send a voicemail
  ; while  inside VoiceMailMain() [option 5 from
  ; mailbox's advanced menu]. If set to 'no',
  ; option 5 will not be listed.

This would enable the option from within the vm app, but you want to do
a dynamic list of mailboxes to deliver to, so by the time we get here,
I think it's going to be to late to to anything useful (since we already
called the voicemail app.)

You could write some dialplan magic with a while loop, so that the user
can dial a specific extn (maybe call it 'group message') and then it
will prompt for a mailbox number, followed by #, or just # to end.

Then it could build this list of mailboxes as a variable before calling
the voicemail app.

I can attempt to build an example if you are interested.

Rob



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Re: [asterisk-users] Questions about Voicemail

[Robert Lister]

On Mon, 2009-11-23 at 14:46 -0500, Dovey Forman wrote:
> Regarding the email to multiple receipients, it is available on an ad-hoc
> basis from the phone?
> 
> IE; call into the voicemail system, enter x digit to send a voicemail to
> multiple users, record the message, then enter the destination mailboxes,
> separated by  # sign...

You can enable an option in the voicemail that allows the prompt:

'To send a message to another user'...

sendvoicemail=yes ; Allow the user to compose and send a voicemail
  ; while  inside VoiceMailMain() [option 5 from
  ; mailbox's advanced menu]. If set to 'no', 
  ; option 5 will not be listed.

This would enable the option from within the vm app, but you want to do
a dynamic list of mailboxes to deliver to, so by the time we get here,
I think it's going to be to late to to anything useful (since we already
called the voicemail app.)

You could write some dialplan magic with a while loop, so that the user
can dial a specific extn (maybe call it 'group message') and then it
will prompt for a mailbox number, followed by #, or just # to end.

Then it could build this list of mailboxes as a variable before calling
the voicemail app.

I can attempt to build an example if you are interested.

Rob



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Re: [asterisk-users] Questions about Voicemail

[Dovey Forman]

Regarding the email to multiple receipients, it is available on an ad-hoc
basis from the phone?

IE; call into the voicemail system, enter x digit to send a voicemail to
multiple users, record the message, then enter the destination mailboxes,
separated by  # sign...

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert
Lister
Sent: Monday, November 23, 2009 1:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Questions about Voicemail

On Mon, 2009-11-23 at 10:37 -0500, Dovey Forman wrote:
> I am sorry if this is not the correct place to post a question.
>
> I am wondering if there is way in asterisk 1.2 to:
>
> 1.  Send a voicemail (from the phone) to multiple recipients.

Yes I believe so.

1. The voicemail app allows delivery to multiple destinations at once:

 - example :

exten => 100,1,VoiceMail(u101&102&103)

2. Create an e-mail alias/list and deliver the voicemail via e-mail to
that alias.


> 2.  Require (as an admin) for users 1st logging into their
> voicemail to change their greeting and/or password.

There is a user option forcegreetings:

forcegreetings = [yes|no]

Sets whether the user will be forced to record a new greeting
when logging in to the system for the first time. Default: no

Example:

forcegreetings = no


Not sure about the forced change PIN, but it should be easy enough to
write a little command wrapper around it and prompt for PIN via the
dialplan.


Rob





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Re: [asterisk-users] Questions about Voicemail

[Robert Lister]

On Mon, 2009-11-23 at 10:37 -0500, Dovey Forman wrote:
> I am sorry if this is not the correct place to post a question.
>
> I am wondering if there is way in asterisk 1.2 to:
>
> 1.  Send a voicemail (from the phone) to multiple recipients.

Yes I believe so.

1. The voicemail app allows delivery to multiple destinations at once:

 - example :

exten => 100,1,VoiceMail(u101&102&103)

2. Create an e-mail alias/list and deliver the voicemail via e-mail to
that alias.


> 2.  Require (as an admin) for users 1st logging into their
> voicemail to change their greeting and/or password.

There is a user option forcegreetings:

forcegreetings = [yes|no]

Sets whether the user will be forced to record a new greeting
when logging in to the system for the first time. Default: no

Example:

forcegreetings = no 


Not sure about the forced change PIN, but it should be easy enough to
write a little command wrapper around it and prompt for PIN via the
dialplan.


Rob





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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] 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?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
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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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Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters

[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.

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

[Olivier]

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.


> > 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


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.


> >
> > 2. How can specify groups in /etc/dahdi/genconf_parameters ?
> > I would like to group SPAN/1 and SPAN/2 into group 1 and SPAN/3 into
> group
> > 2.
> > I was unsuccessful with :
> >
> > group_lines 1
> > pri_termtype
> > SPAN/1  TE
> > SPAN/2  TE
> > group_lines 2
> > pri_termtype
> > SPAN/3  TE
> >
> > 3. After a dahdi_genconf commande, generated
> > /etc/asterisk/dahdi-channels.conf is like this :
> > ; Span 1: B4/0/1 "B4XXP (PCI) Card 0 Span 1" (MASTER) AMI/CCS
> > group=5,11
> > context=remote
> > switchtype = euroisdn
> > signalling = bri_cpe_ptmp
> > channel => 1-2
> > context = default
> > group = 63
> >
> > I can see 2 "group=" and "context=" lines. What is the difference between
> > them ?
> > Shall I care to have them both ?
>
> The second ones are not really needed. Unless you want to assume less of
> the configuration below.
>
> TODO: implement a [section] syntax there to care even less about such
> inclusion.
>
> --
>   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] Questions about Dahdi's /etc/dahdi/genconf_parameters

[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?

> 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.

> 
> 2. How can specify groups in /etc/dahdi/genconf_parameters ?
> I would like to group SPAN/1 and SPAN/2 into group 1 and SPAN/3 into group
> 2.
> I was unsuccessful with :
> 
> group_lines 1
> pri_termtype
> SPAN/1  TE
> SPAN/2  TE
> group_lines 2
> pri_termtype
> SPAN/3  TE
> 
> 3. After a dahdi_genconf commande, generated
> /etc/asterisk/dahdi-channels.conf is like this :
> ; Span 1: B4/0/1 "B4XXP (PCI) Card 0 Span 1" (MASTER) AMI/CCS
> group=5,11
> context=remote
> switchtype = euroisdn
> signalling = bri_cpe_ptmp
> channel => 1-2
> context = default
> group = 63
> 
> I can see 2 "group=" and "context=" lines. What is the difference between
> them ?
> Shall I care to have them both ?

The second ones are not really needed. Unless you want to assume less of
the configuration below.

TODO: implement a [section] syntax there to care even less about such
inclusion.

-- 
   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] Questions on X100P/X101P cards

[Wilton Helm]

>BTW, can someone explain to a libart major like me (;-)) where echo
comes on in a telephone conversation? I seem to recall it's due to the
length of the line between the CO and the local party, but I'm not
sure.

Yes, I'll tackle that.  It takes a finite amount of time for the electrical
signal originating in one phone to arrive at another phone over whatever
path it is taking.  If the path is copper, that time can be fairly small.
If the path is satellite, the time will have to exceed the 1/4 second round
trip to the bird.  If there are SIP packets involved, the time must be
larger than twice the packet size because of the time taken to collect the
data in the packet and then to serialize it at the other and after it
arrives.  If the path involves the internet, there is the path delay there
to be added in (ping will give you an idea of what that is, but it can often
be 50 - 200 ms).

All of this constitutes a delay.  It can be a bit annoying in its own right
because one person asks a question, and twice the delay time elapses before
they start hearing the answer.  However, if there are POTS analog circuits
involved anywhere, a second factor comes into play.  A POTS analog circuit
is two wires, which carry an electrical representation of sound.  Both sides
of the conversation are carried over the same wire. (its called a 2 wire
circuit.  There are also four wire circuits where each direction travels on
a separate pair of wires.  They don't have echo problems.  Digital circuits
also have separate paths for each direction, so are immune to echo)

The problem with a two wire circuit is how to separate the sound going in
both directions.  That is done by something called a 2 wire to 4 wire
converter, also commonly known as a hybrid.  It basically works by
subtracting out what it knows is being sent at the near end from what it
sees on the wire.  If that subtraction is perfect, only what came from the
other end is left and that is presented to the listener.  In the real world,
this isn't perfectly possible, but it can be done fairly well.

However, there is a side effect that comes with the transition from two wire
to four wire.  Some of the signal originating at one end of the wire gets to
the other end and is reflected back.  For an analogy, tie the end of a long
rope to a pipe, stretch it out and snap the other end.  You will see a wave
travel to the pipe and then come back.  If you were able to attach the rope
to the pipe with a suitable dashpot or something that would fully absorb the
wave, nothing would come back.  This reflection from the other end is the
cause of echo.  If the path is terminated in exactly the correct impedance,
there would be no echo.  However, for real circuits over the range of
frequencies that make up sound, that impedance is a complex quantity, and
cannot be exactly matched.

The bottom line is that any circuit with one or more 2 wire analog portions
is going to have some echo.  Since most of the circuits provided by a phone
company are POTS, they are two wire analog from the subscriber to the CO.
If the subscriber equipment is Asterisk, then a 2 wire to 4 wire conversion
and digitization takes place there.  Likewise virtually all telco links are
digital and a conversion takes place in the switch in the CO.  Then at the
other end the process is repeated.  That makes a total of 4 interfaces where
echo can originate in a typical phone call.  If part of the call is SIP, or
internet or satellite, the delay is large enough to guarantee it will be
noticeable.  Since there are several interfaces there can be several echoes.

Another example that illustrates the concept is a speaker phone.  If the
person on the other end is using a speakerphone, then some of what you say
comes out of the speakerphone, bounces off the walls of the room, gets
picked up in the mic and comes back to you.  Again, if the delay is very
large, it will be an echo by the time it gets back to you.  Speakerphones
(if they are full duplex--i.e. allow both parties to talk at once) have to
have echo cancellers to prevent this from happening.


>Is there a way to keep track of this issue, and overtime, to configure
it to answer a call by expecting such and such echo, and thus, avoid
starting sampling from scratch every time?

Unfortunately not.  If you've followed the discussion to this point, you
understand that the magnitude (loudness) of the echo depends on the
impedance mismatch which is unique to the circuitry at each end (for a
typical call) of the call.  The delay time is unique to the call path, which
is likely different for each call, and in the case of internet calls, can
vary within the call.  The echo canceller must constantly do pattern
matching to recognize changes and adjust for them.  Its job is to
subtracting out a signal of exactly the same amplitude as the echo, but of
the opposite polarity and delayed by exactly the path delay the echo is
travelling through.  Since there can easily be four or more ec

Re: [asterisk-users] Questions on X100P/X101P cards

[Vincent]

On Thu, 7 May 2009 13:40:20 +0300, Tzafrir Cohen
 wrote:
>Another thing: their "global-line-standard" should basically (if
>properly written) resolve http://bugs.digium.com/view.php?id=11057 . 
>Though I guess the new code will actually be in DAHDI, as Zaptel is
>frozen.

Ah yes, I seem to remember Zaptel had to change their name to DAHDI
for some reason.

Is there a mailing list to ask for more information about
Zaptel/DAHDI?
http://lists.digium.com/mailman/listinfo/


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Re: [asterisk-users] Questions on X100P/X101P cards

[Vincent]

On Thu, 07 May 2009 10:16:55 -0400, Jon Pounder 
wrote:
>yeah I agree with the above - I never really found echo to ever be a 
>problem, my only complaint was on some less than stellar cpu's I was 
>having dtmf recognition problems.

BTW, can someone explain to a libart major like me (;-)) where echo
comes on in a telephone conversation? I seem to recall it's due to the
length of the line between the CO and the local party, but I'm not
sure.

Is there a way to keep track of this issue, and overtime, to configure
it to answer a call by expecting such and such echo, and thus, avoid
starting sampling from scratch every time?

Thank you.


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Re: [asterisk-users] Questions on X100P/X101P cards

[Jon Pounder]

Jonathan Moore wrote:
> On Wed, May 6, 2009 at 10:53 PM, John Novack
>  wrote:
>   
>> Not sure how you would do that, as the X100 card is an FXO card, won't
>> provide either battery or dial tone to the cordless.
>> What you will want for that is an FXS card or ATA.
>> The X100 card will connect to a central office line, and with the later
>> software echo cancel works OK. Not nearly as bad as some have made it
>> out to be, though for US/Canada lines.  Not suitable for UK and others
>> 
>
>   
yeah I agree with the above - I never really found echo to ever be a 
problem, my only complaint was on some less than stellar cpu's I was 
having dtmf recognition problems.

> Ah, yes. Thanks for correcting me on that, I was getting some things
> mixed up in my head.
>
> -jonathan
>
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Re: [asterisk-users] Questions on X100P/X101P cards

[Jonathan Moore]

On Wed, May 6, 2009 at 10:53 PM, John Novack
 wrote:
> Not sure how you would do that, as the X100 card is an FXO card, won't
> provide either battery or dial tone to the cordless.
> What you will want for that is an FXS card or ATA.
> The X100 card will connect to a central office line, and with the later
> software echo cancel works OK. Not nearly as bad as some have made it
> out to be, though for US/Canada lines.  Not suitable for UK and others

Ah, yes. Thanks for correcting me on that, I was getting some things
mixed up in my head.

-jonathan

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Re: [asterisk-users] Questions on X100P/X101P cards

[Tzafrir Cohen]

On Thu, May 07, 2009 at 11:47:03AM +0200, Vincent wrote:

> X100P SE Setup Guide - Global Line Standards
> http://novavox.co.uk/support/x100p.html
> Richard Spencer supp...@novavox.co.uk

Another thing: their "global-line-standard" should basically (if
properly written) resolve http://bugs.digium.com/view.php?id=11057 . 
Though I guess the new code will actually be in DAHDI, as Zaptel is
frozen.

The driver already checks at init time the chip type and thus it is easy
to give different initialization / behaviour for si3034 and si3035.

-- 
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Re: [asterisk-users] Questions on X100P/X101P cards

[Tzafrir Cohen]

On Thu, May 07, 2009 at 11:47:03AM +0200, Vincent wrote:
> On Thu, 7 May 2009 09:32:19 +0300, Tzafrir Cohen
>  wrote:
> >Some X100P cards (e.g.: those that are based on SI3034, but not those
> >basedon SI3035) support "programmable" impedance settings. Sadly the
> >wcfxo driver does not support it.
> >
> >Fixing it should mostly be a matter of lifting some code from wctdm.c
> >and adapting it. Shouldn't be much of an issue. Anybody wants to try
> >that?
> >
> >(The cards I have at home are SI3035, sadly)
> >
> >A more interesting task would be to add support for some newer
> >(soft/"win"-) modems. Anybody wants to try that?
> >
> >The wcfxo driver needs some love and care. Don't expect Digium to do
> >that for you. They have more important stuff to do. Go and write your
> >own device drivers.
> 
> Thanks guys. So, provided the card has the right DAA chips to match
> the country in which it is used (FCC or CTR21), all it takes to use
> this hardware to handle a POTS line is patching Zaptel? IOW, the
> hardware itself is good enough for SOHO use?

The problem with X100P and "UK calelr ID" is, if I understand it
correctly, that the card does not detect polarity reversal.

But doesn't it provide you with the raw amperage and voltage of the
line? This should allow detecting polarity reversal. And once that is
done, the driver can send up a polarity reversal event, and no change in
Asterisk is required.

This is all within wcfxo.c and thus has no effect on the performance of
other DAHDI devices.

It sounds so simple that there must have been a good reason why
something more complicated has been required.

> 
> According to the following document, NovaVox (which no longer sells
> X100P cards) provides a Zaptel patch for cards sold by X100P.com to
> support non-FCC countries and UK CID:

Quoting later on:

| The DAA chip used in the X100P SE card is a Si3014/Si3034, which does
| support polarity reversal detection. However, because the X100P card 
| Zaptel driver does not include any polarity reversal detection code, 
| the X100P SE polarity detection feature cannot be used. An alternative 
| is to use a patch that uses a history buffer to store the CID value 
| written by Tony Hoyle. By the time the first ring arrives, the buffer 
| has a history of what was received immediately before so the CID 
| information can be extracted.

So, anybody wants to work on the code of wcfxo to add polarity
detection?

CPU is cheap (certainly so when referring to wcfxo, which will only
driver very few channels on a system), but this is still no excuse to
waste it.

Alternatively, some code to manually detect polarity reversal by
sampling amperage and voltage of the card may also help.

> 
> "This document describes how to configure an Open Source IP PBX with
> an X100P Special Edition (SE) FXO PCI card installed to support Caller
> ID received from a UK BT PSTN line. The configuration requires
> implementing a patch for Asterisk®/Zaptel that was originally written
> for the UK but has also been known to work in other countries.[...]
> 
> The Zaptel wcfxo driver has two user configurable modes of operation,
> FCC to support US line standards and CTR21 to support European line
> standards. The Silicon labs Si3012/Si3035 DAA chip used in the
> original Digium X100P card and low cost X100P clone cards only
> supports FCC mode. However, the Si3014/Si3034 DAA chip used on the
> X100P SE supports global line standards."

This is the first thing I mentioned. Should be a relatively trivial
change in the driver.

-- 
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Re: [asterisk-users] Questions on X100P/X101P cards

[Vincent]

On Thu, 7 May 2009 09:32:19 +0300, Tzafrir Cohen
 wrote:
>Some X100P cards (e.g.: those that are based on SI3034, but not those
>basedon SI3035) support "programmable" impedance settings. Sadly the
>wcfxo driver does not support it.
>
>Fixing it should mostly be a matter of lifting some code from wctdm.c
>and adapting it. Shouldn't be much of an issue. Anybody wants to try
>that?
>
>(The cards I have at home are SI3035, sadly)
>
>A more interesting task would be to add support for some newer
>(soft/"win"-) modems. Anybody wants to try that?
>
>The wcfxo driver needs some love and care. Don't expect Digium to do
>that for you. They have more important stuff to do. Go and write your
>own device drivers.

Thanks guys. So, provided the card has the right DAA chips to match
the country in which it is used (FCC or CTR21), all it takes to use
this hardware to handle a POTS line is patching Zaptel? IOW, the
hardware itself is good enough for SOHO use?

According to the following document, NovaVox (which no longer sells
X100P cards) provides a Zaptel patch for cards sold by X100P.com to
support non-FCC countries and UK CID:

"This document describes how to configure an Open Source IP PBX with
an X100P Special Edition (SE) FXO PCI card installed to support Caller
ID received from a UK BT PSTN line. The configuration requires
implementing a patch for Asterisk®/Zaptel that was originally written
for the UK but has also been known to work in other countries.[...]

The Zaptel wcfxo driver has two user configurable modes of operation,
FCC to support US line standards and CTR21 to support European line
standards. The Silicon labs Si3012/Si3035 DAA chip used in the
original Digium X100P card and low cost X100P clone cards only
supports FCC mode. However, the Si3014/Si3034 DAA chip used on the
X100P SE supports global line standards."

X100P SE Setup Guide - Global Line Standards
http://novavox.co.uk/support/x100p.html
Richard Spencer supp...@novavox.co.uk


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Re: [asterisk-users] Questions on X100P/X101P cards

[Tzafrir Cohen]

On Thu, May 07, 2009 at 07:24:21AM +0200, Massimo Nuvoli wrote:
> John Novack ha scritto:
> > 
> 
> > Not sure how you would do that, as the X100 card is an FXO card,
> > won't provide either battery or dial tone to the cordless. What you
> > will want for that is an FXS card or ATA. The X100 card will
> > connect to a central office line, and with the later software echo
> > cancel works OK. Not nearly as bad as some have made it out to be,
> > though for US/Canada lines.  Not suitable for UK and others
> 
> The problem is: analog line is a delicated environment where
> impedance, volts, and line quality are some of the critical components.
> 
> I found my X100 cards failing in production, no software component can
>  solve the "line impedance" or other physical things. I try but no way
> out.

Some X100P cards (e.g.: those that are based on SI3034, but not those
basedon SI3035) support "programmable" impedance settings. Sadly the
wcfxo driver does not support it.

Fixing it should mostly be a matter of lifting some code from wctdm.c
and adapting it. Shouldn't be much of an issue. Anybody wants to try
that?

(The cards I have at home are SI3035, sadly)

A more interesting task would be to add support for some newer
(soft/"win"-) modems. Anybody wants to try that?

The wcfxo driver needs some love and care. Don't expect Digium to do
that for you. They have more important stuff to do. Go and write your
own device drivers.

-- 
   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] Questions on X100P/X101P cards

[Massimo Nuvoli]

John Novack ha scritto:
> 

> Not sure how you would do that, as the X100 card is an FXO card,
> won't provide either battery or dial tone to the cordless. What you
> will want for that is an FXS card or ATA. The X100 card will
> connect to a central office line, and with the later software echo
> cancel works OK. Not nearly as bad as some have made it out to be,
> though for US/Canada lines.  Not suitable for UK and others

The problem is: analog line is a delicated environment where
impedance, volts, and line quality are some of the critical components.

I found my X100 cards failing in production, no software component can
 solve the "line impedance" or other physical things. I try but no way
out.

After i bougt a 'real' analog board, even the worst is much much much
better.

I now, the cost is a problem, and there is NOT a single cheap analog
board from DIGIUM or SANGOMA or others

Bye.

begin:vcard
fn:Massimo Nuvoli
n:Nuvoli;Massimo
org:Progetto Archivio SRL
adr:;;Via Giustetto 75;Abbadia Alpina Pinerolo;TO;10060;Italia
email;internet:mass...@archivio.it
title:Amministratore Delegato
tel;work:0121303544
tel;fax:0121040601
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end:vcard



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Re: [asterisk-users] Questions on X100P/X101P cards

[John Novack]

Jonathan Moore wrote:
> On Wed, May 6, 2009 at 8:47 PM, ContactTel Business
>  wrote:
>   
>> I'd say in life you get what you pay for.. and sometime you even pay for
>> stuff that should be free..
>> 
>
> I have to agree.
>
> I have a few of these cards I started out with.  They were great for the 
> "wow, I finally got asterisk to do something" but worthless for actually 
> running in a system for any kind of real work.  That being said, I have every 
> intention of leaving one in the system as a quick way to get a cordless phone 
> in our work area (we have an old cordless telephone laying around... hook it 
> up, lets me call up front with no problems... that I can't live with).
>
>   
Not sure how you would do that, as the X100 card is an FXO card, won't 
provide either battery or dial tone to the cordless.
What you will want for that is an FXS card or ATA.
The X100 card will connect to a central office line, and with the later 
software echo cancel works OK. Not nearly as bad as some have made it 
out to be, though for US/Canada lines.  Not suitable for UK and others

John Novack

-- 
Dog is my co-pilot


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Re: [asterisk-users] Questions on X100P/X101P cards

[Jonathan Moore]

On Wed, May 6, 2009 at 8:47 PM, ContactTel Business
 wrote:
> I'd say in life you get what you pay for.. and sometime you even pay for
> stuff that should be free..

I have to agree.

I have a few of these cards I started out with.  They were great for
the "wow, I finally got asterisk to do something" but worthless for
actually running in a system for any kind of real work.  That being
said, I have every intention of leaving one in the system as a quick
way to get a cordless phone in our work area (we have an old cordless
telephone laying around... hook it up, lets me call up front with no
problems... that I can't live with).

I would not suggest using these cheap cards in production systems
where a little bit of bad service really matters.  After all, it's for
"the phone system" the one thing most people assume just always
works... as always... ymmv..

--
jon

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Re: [asterisk-users] Questions on X100P/X101P cards

[ContactTel Business]

I'd say in life you get what you pay for.. and sometime you even pay for
stuff that should be free..

These knockoff cards, can be built in-house for 20$ or less using an old
walkie talkie, a rope, some standard matches, and an old MCgyver Tv
episode..They do just that, echo the sound back to the other end.. no
seriously..

Used 3, all 3 where thrown out either a window, or another opening .. 

It's great for testing don't get me wrong, once PSTN echo starts to drive
you or the other party mad , or your cheap cap's start to go off specs,
you'll say "I should of bought 1 of the 50$ instead of 5 of the $10 ones"..
of course, same strategy goes with china knockoffs, you run 6 pair of shoes
for the price of 1 that lasts 6 times the other..

So basically for 10$ try it out..

Then when you want 2-3 ports, or do voice rec (spynx openmrc etc whatever)
you'll need quality

My 0.02


>>-Original Message-
>>From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
>>boun...@lists.digium.com] On Behalf Of Andrew Joakimsen
>>Sent: May-06-09 7:01 PM
>>To: Asterisk Users Mailing List - Non-Commercial Discussion
>>Subject: Re: [asterisk-users] Questions on X100P/X101P cards
>>
>>I use these cards and they work pretty well. FWIW when Digium sold
>>them they were also just winmodems with a resistor removed to change
>>the PCI device ID. Later on the Zaptel driver included the device ID
>>of the winmodem.
>>
>>I used to be able to get the winmodem itself for under $10, but I
>>think they are discontinued now. Ambient = Intel, FWIW. If you want
>>I'll dig out out and give you the details.
>>
>>If you need a large quantity I would try to find the winmodems that
>>are compatible.
>>
>>
>>On Wed, May 6, 2009 at 08:43, Vincent 
>>wrote:
>>> Hello,
>>>
>>>        I'm looking for a dirt cheap solution for SOHO use to handle
>>at most
>>> a couple of POTS lines, and I notice that X10?P cards go for $15 on
>>> eBay as opposed to $90 for an OpenVox card or over $200 for a
>>Sangoma.
>>>
>>> I have a couple of questions about those cheap FXO cards:
>>>
>>> 1. Are they all glorified softmodems, ie. none has an on-board CPU or
>>> DSP and outsources all processing to the computer's CPU?
>>>
>>> 2. Are they all bad, no matter what chipset is used (Intel, Motoral,
>>> Ambient)? If not, which offer good enough quality to handle a single
>>> POTS line?
>>>
>>> 3. Why are they often bad quality? Because the driver itself is badly
>>> written? Because PC's don't have enough speed to handle the tasks
>>> using their own CPU (hard to believe, but I don't know)?
>>>
>>> Thank you.
>>>
>>>
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Re: [asterisk-users] Questions on X100P/X101P cards

[Andrew Joakimsen]

I use these cards and they work pretty well. FWIW when Digium sold
them they were also just winmodems with a resistor removed to change
the PCI device ID. Later on the Zaptel driver included the device ID
of the winmodem.

I used to be able to get the winmodem itself for under $10, but I
think they are discontinued now. Ambient = Intel, FWIW. If you want
I'll dig out out and give you the details.

If you need a large quantity I would try to find the winmodems that
are compatible.


On Wed, May 6, 2009 at 08:43, Vincent  wrote:
> Hello,
>
>        I'm looking for a dirt cheap solution for SOHO use to handle at most
> a couple of POTS lines, and I notice that X10?P cards go for $15 on
> eBay as opposed to $90 for an OpenVox card or over $200 for a Sangoma.
>
> I have a couple of questions about those cheap FXO cards:
>
> 1. Are they all glorified softmodems, ie. none has an on-board CPU or
> DSP and outsources all processing to the computer's CPU?
>
> 2. Are they all bad, no matter what chipset is used (Intel, Motoral,
> Ambient)? If not, which offer good enough quality to handle a single
> POTS line?
>
> 3. Why are they often bad quality? Because the driver itself is badly
> written? Because PC's don't have enough speed to handle the tasks
> using their own CPU (hard to believe, but I don't know)?
>
> Thank you.
>
>
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Re: [asterisk-users] Questions on X100P/X101P cards

[Gordon Henderson]

On Wed, 6 May 2009, Vincent wrote:

> On Wed, 06 May 2009 14:02:20 +0100, "Alan Lord (News)"
>  wrote:
>> For a cheap backup to your VOIP service they do the job. I wouldn't use
>> them for a "proper" system though.
>
> Thanks for the feedback. I have two more questions:
> 1. Can the OSLEC echo canceller run OK on an 1.6GHz Intel Atom not
> doing much more than this and running Asterisk?

The OSLEC benchmark tell me it can run 14 concurrent instances on a 550MHz 
VIA C3 processor. On a 1.6GHz Atom, it tells me it can do 80 concurrent 
instances, so I think the overhead of OSLEC is the least of your problems 
there.

> 2. Is it good enough to handle a single FXO line for professional use?

I use OSLEC in my standard PBX products - Not with x100p cards though, but 
with Digium and OpenVox cards. However I'm in the UK - your lines may be 
different...

Gordon


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Re: [asterisk-users] Questions on X100P/X101P cards

[Vincent]

On Wed, 06 May 2009 14:02:20 +0100, "Alan Lord (News)"
 wrote:
>For a cheap backup to your VOIP service they do the job. I wouldn't use 
>them for a "proper" system though.

Thanks for the feedback. I have two more questions:
1. Can the OSLEC echo canceller run OK on an 1.6GHz Intel Atom not
doing much more than this and running Asterisk?
2. Is it good enough to handle a single FXO line for professional use?
3. Can you give me a pointer about which X100p you bought on eBay?
AFAIK, there are three chipsets : Intel, Motorola, and Ambient.

Using a $15 card over an $80 card is not insignificant because I could
then sell a small server for $99.

Thank you.


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Re: [asterisk-users] Questions on X100P/X101P cards

[Alan Lord (News)]

On 06/05/09 13:43, Vincent wrote:
> Hello,
>
>   I'm looking for a dirt cheap solution for SOHO use to handle at most
> a couple of POTS lines, and I notice that X10?P cards go for $15 on
> eBay as opposed to $90 for an OpenVox card or over $200 for a Sangoma.
>
> I have a couple of questions about those cheap FXO cards:
>
> 1. Are they all glorified softmodems, ie. none has an on-board CPU or
> DSP and outsources all processing to the computer's CPU?
>
> 2. Are they all bad, no matter what chipset is used (Intel, Motoral,
> Ambient)? If not, which offer good enough quality to handle a single
> POTS line?
>
> 3. Why are they often bad quality? Because the driver itself is badly
> written? Because PC's don't have enough speed to handle the tasks
> using their own CPU (hard to believe, but I don't know)?

Hi Vincent,

I bought a cheap "eBay" X100p card over a year ago. When I first tried 
it was appalling. I couldn't get rid of the echo and noise no matter what.

I then came across OSLEC (at the time a new Free Echo Canceller). A bit 
of hacking to get it to work and hey-presto! No more echo.

I have been using the same card ever since with no noticeable issues.

I think OSLEC is now the default EC for many distributions so I would 
have thought you will be fine although, of course, YMMV.

For a cheap backup to your VOIP service they do the job. I wouldn't use 
them for a "proper" system though.

HTH

Al


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Re: [asterisk-users] questions regarding DAHDI

[David fire]

the echo cancelation sintaxis has change in dahdi read the documents about
it.
David

2008/12/6 Jon Weisman <[EMAIL PROTECTED]>

>  Just moved over to DAHDI today and have a few questions.
>
> My environment is:
>
> Asterisk 1.4.22
> Using a TE410P
> 4 PRI's
>
> 1. Now when calls come in on those PRI's I get this message in the console:
>
> [Dec  6 19:42:18] WARNING[31557]: chan_dahdi.c:1481 dahdi_enable_ec: Unable
> to enable echo cancellation on channel 41 (No such device)
>
> How can I resolve this?
> 2. Do I need to be running echo cancellation on TDM circuits? (PRI's?)
>
> 3. I compiled DAHDI tools and I seem to have all the scripts, however I do
> not have or can not seem to find dahdi_tools. How can I get this? I believe
> this is the zttools replacement.
>
> 4. Is there a way for me to verify that I am receiving clock source on
> these PRI's from the telco? I need to verify this because it seems that if I
> use my internal clock source my HDLC errors seem to go away. (Still testing)
>
>
> Thanks in advance,
> Jon
>
>
>
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Re: [asterisk-users] questions running 2 asterisk under the same LAN

[Steve Edwards]

On Fri, 18 Apr 2008, Pete Kay wrote:

> I want to know if I am running two machines each with its own Asterisk on my
> LAN, show I change the port of one of the Asterisk to something  like 5061?

It is common to have multiple instances of Asterisk listening to the same 
port number on the same LAN.

> Otherwise, how does an external SIP client ( like IPkall.com) knows how to
> route DID call to Asterisk?   What is the solution for this kind of setup?

It depends on the provider and your networking resources.

Some (most?) providers "know" where to send the call based on 
registration. Each host can register as many accounts with as many 
providers as desired.

Port forwarding can also solve a lot of problems.

Switching to IAX (with registration) makes most of these issues moot.

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] questions running 2 asterisk under the same LAN

[linuxian iandsd]

you surely are using port forwarding right now, something like this:

wan(5061) <-you_router_here-->
lan(5061)<--->Asterisk_box_1(5061)

so you only need to add this :

wan(5062) <-you_router_here-->
lan(5061)<--->Asterisk_box_2(5061)

just tell your provider that second account goes to port 5062.

another solution would be to add a second connection to internet & use it
for second box as internet is cheap these days.



On Thu, Apr 17, 2008 at 4:03 PM, Pete Kay <[EMAIL PROTECTED]> wrote:

> Hi,
>
> I want to know if I am running two machines each with its own Asterisk on
> my LAN, show I change the port of one of the Asterisk to something  like
> 5061?
> Otherwise, how does an external SIP client ( like IPkall.com) knows how to
> route DID call to Asterisk?   What is the solution for this kind of setup?
>
> Thank you very much in advance for your inputs.
>
> Regards,
> Pete
>
>
>
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Re: [asterisk-users] questions running 2 asterisk under the same LAN

[Steve Totaro]

On Thu, Apr 17, 2008 at 12:03 PM, Pete Kay <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I want to know if I am running two machines each with its own Asterisk on my
> LAN, show I change the port of one of the Asterisk to something  like 5061?
> Otherwise, how does an external SIP client ( like IPkall.com) knows how to
> route DID call to Asterisk?   What is the solution for this kind of setup?
>
> Thank you very much in advance for your inputs.
>
> Regards,
> Pete

Maybe if you draw a clearer picture of what you have/need.

I guess you only have one public IP and everything is behind a NAT router?

If you have or can get additional IPs, it would make port forwarding
rules on your router easier (if it supports multiple IPs) than
changing ports on the Asterisk boxen, that would probably be my
approach if possible, from what I can gather from your description.

Thanks,
Steve Totaro

This link might be of help.
http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions

Thanks,
Steve Totaro

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Re: [Asterisk-Users] Questions from a working doctors' office installation

[Luki]

Michael,


Is memory leak still as much an issue with 1.2.7 versus 1.2.5?  In other words,
is it worth it to upgrade a working, memory-leaking 1.2.5 to 1.2.7 or 1.2.8
just to potentially encounter other bugs in the new versions?  Have other
people been satisfied with the new versions so far?  I have Polycom 501s and
301s.  Call transfers are prone to crashing the system, getting sent to the
wrong phone, etc.


Huh... interesting... I had (and actually still do) have 1.2.5 version
perfectly; it's been >60 days since the last restart so I figure I
would have noticed memory leaks until now. This system is in a small
real estate office with 15 extensions but with hundreds of calls a
day, plenty of transfers. However, it's SIP only, no hardware, no IAX.
Perhaps the memory leaks are specific to certain hardware or protocol
or activity. Anyway, I'm not going to argue there are no memory leaks
-- if you have them, try an upgrade :).


Is there some sort of rollback function?  I'm considering having a second PBX
box for the upgraded version, then keeping the working production system as a
backup.


Yes. Here's what I do. I symlink the executable asterisk ->
asterisk-1.2.5 and directory modules -> modules-1.2.5. When I want to
switch versions, I change the symlinks for those two keeping
everything else the same. No problems going back and forth, at least
not between 1.2.x versions. When you build asterisk, don't do a "make
install" but simply copy the executable to asterisk-VERSION and all
.so files from the build directory to modules-VERSION -- i.e. cp -a
`find -name '*.so'` /usr/lib/asterisk/modules-VERSION/. I run this in
a chrooted environment, but you don't have to.


My PSTN providers are voipjet (out) and Axvoice (in).  Sometimes we have
dropped calls incoming, or busy lines outgoing.  Anyone else using good service
providers they can recommend?


That's something to the -biz list, probably but you may contact me off
list if you need suggestions.

--Luki
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Re: [Asterisk-Users] Questions on ANI

[Li Mark]

May I know if you could send me some coding on the *.conf, so that I
can follow the idea that you suggest??

ML

2006/5/2, Lacy Moore - Aspendora <[EMAIL PROTECTED]>:


If the incoming call is from a PRI with DIDs, you could just simply map the
caller ID name to the company name (company name being the name of the
company being called, this is opposite of normal caller id name) based on
the DID.  Or, you could also play a sound file containing the company name.
I'm looking at doing that with queues.  We have multiple businesses all
sharing the same office/employees/etc, and I plan to have a queue for each
company.  When the receptionist picks up the phone, she will hear an
announcement with the name of the company that the person called.




On 5/2/06, Li Mark <[EMAIL PROTECTED]> wrote:
>


I set up the Asterisk for my company which is a business center, I will
assign a specific telephone number to my client that uses my serivces.  All
of their incoming calls will be first picked up by the receiptionist, can I
disply the company name instead of the called number on my receptionist's
telephone display, so that she can answer the call with the right identity
at once...

Regards,
ML
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--
Lacy Moore
Aspendora, Inc.
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Re: [Asterisk-Users] Questions on ANI

[Lacy Moore - Aspendora]

If the incoming call is from a PRI with DIDs, you could just simply map the caller ID name to the company name (company name being the name of the company being called, this is opposite of normal caller id name) based on the DID.  Or, you could also play a sound file containing the company name.  I'm looking at doing that with queues.  We have multiple businesses all sharing the same office/employees/etc, and I plan to have a queue for each company.  When the receptionist picks up the phone, she will hear an announcement with the name of the company that the person called.  

 
 
On 5/2/06, Li Mark <[EMAIL PROTECTED]> wrote:


I set up the Asterisk for my company which is a business center, I will assign a specific telephone number to my client that uses my serivces.  All of their incoming calls will be first picked up by the receiptionist, can I disply the company name instead of the called number on my receptionist's telephone display, so that she can answer the call with the right identity at once... 

 
Regards,
ML___--Bandwidth and Colocation provided by 
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreAspendora, Inc. 
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Re: [Asterisk-Users] Questions on call recording and conference.

[Dinesh Nair]

On 03/31/06 08:24 Wai Wu said the following:

In Asterisk, what happens to the files when both legs of the call hangs
up?   Is there a way to create a conference room on the flight? i.e.
without pre-defining the conference ID in meetme.conf.


look at the 'd' option to MeetMe.

--
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[EMAIL PROTECTED](0 0)http://www.alphaque.com/
+==oOO--(_)--OOo==+
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|   for b in clients employers associates relatives neighbours pets; do   |
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Re: [Asterisk-Users] Questions about real-time voicemail, foreign languages and voicemail folders...

[Carlos Chavez]

On Sun, 2005-07-03 at 12:31 -0400, Leo Burd wrote:
> Hello there,
> 
> I'm trying to configure my voicemail system and I have a couple of 
> questions:
> 
> * Is real-time voicemail already working?  If so, where is it that I 
> should specify the database name, user and password?  Where can I get 
> more information about the different options that exist and the 
> different files that need to be changed?
> 
> * Is it possible to setup the voicemail interface to speak in Spanish?
> 
> * I've heard it is possible to create folders within mailboxes.  Is 
> there any documentation written about that?  In fact, would anyone 
> recommend a good reference about comedian mail?
> 
> Thanks in advance,
> 
Everything you want is possible.  Please read the Asterisk
documentation at www.voip-info.org to set up your system.  

-- 
Telecomunicaciones Abiertas de Mexico
Carlos Chavez
Director de Tecnologia
+52-55-91169161 Ext. 2001


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Re: [Asterisk-Users] Questions about real-time voicemail, foreign languages and voicemail folders...

[Matthew Boehm]

> * Is real-time voicemail already working?

Yes, and has been working for quite some time. Since December I believe.


>  If so, where is it that I
> should specify the database name, user and password?

Read docs/README.extconfig


> Where can I get 
> more information about the different options that exist and the
> different files that need to be changed?

No files need to be changed, except extconfig.conf


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Re: [Asterisk-Users] Questions about contexts

[Rich Adamson]

> I'm trying to clarify contexts and their uses. I do have a good
> general understanding of them. My question is about "undeclared"
> and "non-existant" contexts.
> 
> If I have a block somewhere (in sip.conf, for example), and it
> has no "context=thiscontext" field, does it just automatically
> use the "default" context? Or is this settable? (I see there is
> an entry for context in the [general] block. I'm guessing this
> can be used to set a global context for that file. But if this
> is omitted, what context does the file get? Is "default" a
> hardcoded fallback?)

Let's see if this helps...  Many of asterisk config definitions
use a "context=" to refer to other sections of config files. In
most cases (but not all) asterisk will _assume_ the use of a 
default context if you don't specify one. The assumed context is
usually "context=default".

In extensions.conf (as an example), if you don't have any [xx] 
context defined, then all definitions starting from the top of the
file are assumed to be in the [default] context. Same with the
sip.conf file.  That's one reason why those that are interested in
security _don't_ use anything that assumes a default context.

So in sip.conf (as an example), after the [general] section but
before any other defintions, you should have something like
context=bogon-calls to catch any random sip connect attempts. Then
in extensions.conf, have a section like [bogon-calls] that does
something with those sip connect attempts that don't specifically
match other definitions in sip.conf. The extensions.conf entry
might be something like this:
[bogon-calls]   
 exten => _X.,1,Dial(SIP/3000,15,r)
 exten => _X.,2,Congestion
where you forcibly send unauthorized sip call attempts to a well
defined destination (sip/3000).

Sooner or later asterisk systems that are exposed to the internet
will be hit with random sip and iax connect attempts. Deal with
them now and define all contexts instead of relying on 'default'.

To over simplify this, anytime you see a 'context=' think in terms
of that functioning just like a 'goto' programming statement.

> Someone suggested that I could define a "non-existant" context,
> being one that isn't actually declared, and set that in my
> "context=" line, and calls so directed wouldn't go anywhere.
> Would it be possible to do this with an empty definition (are
> they equivalent?)

Yes. A 'context=nowhere' with [nowhere] defined and nothing in it
will be exactly the same thing as missing the [nowhere]. They 
function the same.
 
> Lastly, I'm trying to get a good handle on "include" contexts.
> If I have:
> [contexta]
> ; commands omitted
> [contextb]
> ; commands omitted
> 
> and I add:
> [contextc]
> include => contexta
> include => contextb
> ; commands omitted
> 
> Then "contexta" and "contextb" are available to "contextc"'s
> dialers, right?

Yes, and that exact approach is also very useful to "force" the
order in which definitions are executed. So, if you used the above
example in extensions.conf, when the statement context=contexc is
seen, it _will_ execute what's in contexta _before_ it attempts
contextb. The approach might apply if you had something like
exten=>_1800XXX. in contexta and exten=1800 in contextb. (Maybe
bad example, but you get the picture, I'm sure.)

> Do I have to make sure there are no overlapping extensions in
> all three contexts?

No, see my bad example above. The bad example assumes you have
a valid extensions 1800, but you allow your users to also dial
1-800-555-1212. Which do you want to check for _first_ in your
dialplan?


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Re: [Asterisk-Users] Questions about contexts

[Moises Silva]

i guess in sip.conf, and iax.conf you can put a parameter
defaultcontext=somecontext in the [general] context, so all the users
and /or peers that does not have a defined context will get there.

about the includes, yes, all the extensions in context 'a' and 'b' are
available to 'c'. You can put overlapping extensions, but the first
match will be used. So, if you first include => a, and later include
=> b, 'a' will take precedence. Note that overlapping extensions
inside the same context, lets say

[overlapping]
exten => _XX,blah
exten => _.,blah

it doesnt matter that the XX extension appears firts, the matchin will
be the more generic one., so if you want XX first, you have to
separate the extensions and include them by include => xxcontext
include => theother

so the xx extension will be matched first

best regards

best regards

On 6/14/05, Jerry <[EMAIL PROTECTED]> wrote:
> 
> I'm trying to clarify contexts and their uses. I do have a good
> general understanding of them. My question is about "undeclared"
> and "non-existant" contexts.
> 
> If I have a block somewhere (in sip.conf, for example), and it
> has no "context=thiscontext" field, does it just automatically
> use the "default" context? Or is this settable? (I see there is
> an entry for context in the [general] block. I'm guessing this
> can be used to set a global context for that file. But if this
> is omitted, what context does the file get? Is "default" a
> hardcoded fallback?)
> 
> Someone suggested that I could define a "non-existant" context,
> being one that isn't actually declared, and set that in my
> "context=" line, and calls so directed wouldn't go anywhere.
> Would it be possible to do this with an empty definition (are
> they equivalent?)
> 
> Lastly, I'm trying to get a good handle on "include" contexts.
> If I have:
> [contexta]
> ; commands omitted
> [contextb]
> ; commands omitted
> 
> and I add:
> [contextc]
> include => contexta
> include => contextb
> ; commands omitted
> 
> Then "contexta" and "contextb" are available to "contextc"'s
> dialers, right?
> Do I have to make sure there are no overlapping extensions in
> all three contexts?
> 
> Thanks,
> J.
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Re: [Asterisk-Users] Questions about a 7960 and images

[Henry Devito]

You can get around this by creating the SIPDefault with only the first line: 
image_version: "P0S3-07-3-00"
and a blank SIP

Email me off list if you need help
- Original Message - 
From: "Gregory Wiktor - ADCom Corp." <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial 
Discussion" 
Sent: Friday, April 22, 2005 7:26 PM
Subject: RE: [Asterisk-Users] Questions about a 7960 and images

Hello Ron,
This link seems to indicate that there may be other issues with
upgrading that does require a midstream upgrade... They say connect it
to a callmanager, but someone with cco managed to do it in a 2 step
upgrade...
http://www.vovida.org/pipermail/vocal/2002-August/006404.html
Content-Transfer-Encoding: quoted-printable
Hi there,
Cisco 7960 phone with firmware P003AM30 has a bug that cannot load a new
=
load file that is more than 393216 bytes in size (384K).  Hence you will
=
see your firmware being downloaded again and and again via your TFTP.
You will need to upgrade the phone to P00303010102 first before you will
=
be able to load the latest firmware into the phone.=20
Hope this helps!
David
~~~
Regards,
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ron
Wellsted
Sent: Friday, April 22, 2005 5:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Questions about a 7960 and images
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Gregory Wiktor - ADCom Corp. wrote:
Hello All,
I was wondering how everyone got along with cisco 7960's. I just
picked one up and I am having problems locating an image.  I called
cisco, but they will not sell to end users...  Does anyone know a
place where it can be purchased in the US?
It has stock firmware, and the skinny seems to crash asterisk oddly...
Also, does it require a download to run?  For example, can I configure

it, then bring it to another office and just plug in and go, or must
it tftp from the head server?
Regards,
Greg
You could try CDW or Insight, see <http://www.voip-info.org/wiki-Cisco>
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961  Gossiptel:9309811
N 52.567623, W 2.137621
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Re: [Asterisk-Users] Questions about a 7960 and images

[Ron Wellsted]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Gregory Wiktor - ADCom Corp. wrote:
> Hello All,
> I was wondering how everyone got along with cisco 7960's. I just picked
> one up and I am having problems locating an image.  I called cisco, but
> they will not sell to end users...  Does anyone know a place where it
> can be purchased in the US?
> 
> It has stock firmware, and the skinny seems to crash asterisk oddly...
> 
> Also, does it require a download to run?  For example, can I configure
> it, then bring it to another office and just plug in and go, or must it
> tftp from the head server?
> 
> Regards,
> Greg

You could try CDW or Insight, see 

- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961  Gossiptel:9309811
N 52.567623, W 2.137621
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org

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RE: [Asterisk-Users] Questions about a 7960 and images

[Gregory Wiktor - ADCom Corp.]

Hello Ron,

This link seems to indicate that there may be other issues with
upgrading that does require a midstream upgrade... They say connect it
to a callmanager, but someone with cco managed to do it in a 2 step
upgrade...

http://www.vovida.org/pipermail/vocal/2002-August/006404.html 


Content-Transfer-Encoding: quoted-printable

Hi there,

Cisco 7960 phone with firmware P003AM30 has a bug that cannot load a new
=
load file that is more than 393216 bytes in size (384K).  Hence you will
=
see your firmware being downloaded again and and again via your TFTP.

You will need to upgrade the phone to P00303010102 first before you will
=
be able to load the latest firmware into the phone.=20

Hope this helps!
David
~~~

Regards,
Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ron
Wellsted
Sent: Friday, April 22, 2005 5:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Questions about a 7960 and images

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Gregory Wiktor - ADCom Corp. wrote:
> Hello All,
> I was wondering how everyone got along with cisco 7960's. I just 
> picked one up and I am having problems locating an image.  I called 
> cisco, but they will not sell to end users...  Does anyone know a 
> place where it can be purchased in the US?
> 
> It has stock firmware, and the skinny seems to crash asterisk oddly...
> 
> Also, does it require a download to run?  For example, can I configure

> it, then bring it to another office and just plug in and go, or must 
> it tftp from the head server?
> 
> Regards,
> Greg

You could try CDW or Insight, see <http://www.voip-info.org/wiki-Cisco>

- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961  Gossiptel:9309811
N 52.567623, W 2.137621
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Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org

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RE: [Asterisk-Users] Questions about VOIP phones

[Howard Lowndes]

On Wed, 2004-12-22 at 10:22, Martin Geldenhuys wrote:
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Kobus
> Wolvaardt
> Sent: Wednesday, December 22, 2004 8:00 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Questions about VOIP phones
> 
> hi,
> 
> > Is it sure, that you need on both offices * ???
> > Maybe you are cheaper to install * only at one site.
> > Maybe you add a 1-800 number instead of both sides PSTN
> 
> I dont know what 1-800 means or if we have an equivalent in South Africa,  
> but we have gov protected telecom monopoly (makes most options expensive).
> 
> 
> > I would try to avoid to use any plain phones at all, since you limit for  
> > these phones some functions. While it is cheaper I would try to avoid  
> > it. It might be interesting to use 2 to 4 FXS to use them into an old  
> > PBX, till this PBX faces out.
> 
> No old PBX, I'm setting it up from scratch, which is nice. Unfortunately I  
> know nothing of these things and will have to learn really fast.
> 
> 
> >
> > I believe you only need one * and just a gateway at the other place.
> >
> I need phone lines at both offices and PBX at both offices.
> 
> Also: We dont have a static IP (Broadband in SA sucks! 512k capped to 3  
> gig with dynamic IP) will that be a problem hooking the two offices up?  
> How much voices can be carried on 512k (assuming it always runs at 512k)
> 
Gee that sucks.  I thought we were badly of in AU.  As for dyn IP you
can use dyndns.com or dyndns.org as your name servers to overcome that
problem.

You don't say whether it is 512/512 or 512/128.  I get good performance
on the first but only moderate performance on the latter.

> Have the same problem and living in south Africa jhb, you can get a static
> dns name for free to link the two ser4vers.
> 
> 
> I assume the hookup if two *'s are used will be with AIX.
> 
> Thanks,
> Kobus
-- 
Howard.
LANNet Computing Associates;
Your Linux people <http://www.lannetlinux.com>
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."


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RE: [Asterisk-Users] Questions about VOIP phones

[Martin Geldenhuys]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kobus
Wolvaardt
Sent: Wednesday, December 22, 2004 8:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Questions about VOIP phones

hi,

> Is it sure, that you need on both offices * ???
> Maybe you are cheaper to install * only at one site.
> Maybe you add a 1-800 number instead of both sides PSTN

I dont know what 1-800 means or if we have an equivalent in South Africa,  
but we have gov protected telecom monopoly (makes most options expensive).


> I would try to avoid to use any plain phones at all, since you limit for  
> these phones some functions. While it is cheaper I would try to avoid  
> it. It might be interesting to use 2 to 4 FXS to use them into an old  
> PBX, till this PBX faces out.

No old PBX, I'm setting it up from scratch, which is nice. Unfortunately I  
know nothing of these things and will have to learn really fast.


>
> I believe you only need one * and just a gateway at the other place.
>
I need phone lines at both offices and PBX at both offices.

Also: We dont have a static IP (Broadband in SA sucks! 512k capped to 3  
gig with dynamic IP) will that be a problem hooking the two offices up?  
How much voices can be carried on 512k (assuming it always runs at 512k)

Have the same problem and living in south Africa jhb, you can get a static
dns name for free to link the two ser4vers.


I assume the hookup if two *'s are used will be with AIX.

Thanks,
Kobus

-- 
Produksiebestuurder
Die Wêreld media
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Re: [Asterisk-Users] Questions about VOIP phones

[Kobus Wolvaardt]

hi,
Is it sure, that you need on both offices * ???
Maybe you are cheaper to install * only at one site.
Maybe you add a 1-800 number instead of both sides PSTN
I dont know what 1-800 means or if we have an equivalent in South Africa,  
but we have gov protected telecom monopoly (makes most options expensive).


I would try to avoid to use any plain phones at all, since you limit for  
these phones some functions. While it is cheaper I would try to avoid  
it. It might be interesting to use 2 to 4 FXS to use them into an old  
PBX, till this PBX faces out.
No old PBX, I'm setting it up from scratch, which is nice. Unfortunately I  
know nothing of these things and will have to learn really fast.


I believe you only need one * and just a gateway at the other place.
I need phone lines at both offices and PBX at both offices.
Also: We dont have a static IP (Broadband in SA sucks! 512k capped to 3  
gig with dynamic IP) will that be a problem hooking the two offices up?  
How much voices can be carried on 512k (assuming it always runs at 512k)

I assume the hookup if two *'s are used will be with AIX.
Thanks,
Kobus
--
Produksiebestuurder
Die Wêreld media
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