[asterisk-users] Polycom getting DCHP address from wrong VLAN

[Thermal Wetland]

Hello,

I have been tearing my hair out on this issue for 2 days, any help
would be appreciated.

We have a normal network and a Cisco SGE2010P switch - a 48 port PoE switch

There are two VLANs, 1(data) & 50(VoIP).  When Polycoms are connected
to the switch with VLAN 50 hard coded in the config they grab a DHCP
address from VLAN 1, the PVID for the switch port.

The ports have membership in VLAN 1 as the PVID and VLAN 50 as tagged
traffic.  I know the VoIP DHCP server is working because if I change a
port to have a PVID of 50 any device gets the address from the VoIP
DHCP server.

I have tried the ports as 'general' and 'trunk' with no success.

Any help would be greatly appreciated, I don't have much hair left!

-- 
-Thermal

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Re: [asterisk-users] asterisk-users Digest, Vol 75, Issue 7

[Fazil Amaan]

Hi,


I cannot get asterisk to start again after the g729 install failed.


kindly advise what's the problem.

Thank's


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[asterisk-users] Radius client support

[Nikhil]

  Hi

   Will radius client in asterisk can use with third party radius 
servers instead of freeradius ?,if supports how do I  configure asterisk 
to make it work.

Thanks
Nikhil

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Re: [asterisk-users] Dahdi error

[Shaun Ruffell]

On 10/7/10 2:07 PM, Flavio Miranda wrote:
> asterisk:/etc/asterisk# /etc/init.d/dahdi start
> Loading DAHDI hardware modules:
> FATAL: Error inserting dahdi (/lib/modules/2.6.26-2-686/dahdi/dahdi.ko):
> Device or resource busy
> wct4xxp: done wcte12xp: done wct1xxp: done wcte11xp: done wctdm24xxp:
> done wcfxo: done wctdm: done wcb4xxp: error wctc4xxp: done xpp_usb: done
> Error: missing /dev/dahdi!

Best guess based on the information you provided:  zaptel was installed 
on this machine and is already loaded and registered major number 196. 
That would explain both the "Device or resource busy" error, and the 
fact that dahdi failed to load, yet most of the board drivers appear to 
have loaded (since the zaptel ones probably loaded up and the wcb4xxp 
driver did not load).

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[asterisk-users] RES: Alert-Info advice

[Rafael Prado Rocchi]

Hi, 
I use it on Linksys PAP2, I think you need to put brackets after Alert-info
on your code. 

Check mine:

exten => _1XXX,1,SIPAddHeader(Alert-Info: )




Atenciosamente,

Rafael Prado

PRACTIS - Comunicação & Tecnologia 
Av Aquidaban, 766 - Conj 51
CEP 13026-510, Campinas/SP - Brasil




-Mensagem original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Em nome de Ishfaq Malik
Enviada em: quarta-feira, 29 de setembro de 2010 11:13
Para: asterisk-users@lists.digium.com
Assunto: [asterisk-users] Alert-Info advice

Hi guys

I'm using asterisk 1.4 and going on to Snom phones. I'm trying to add a
sip header to make the Snom phone use a different ring tone on one
particular incoming number. I have added the following to the dial plan
of the incoming context

+--+--+---+--+--+---
--+
| id   | context  | exten | priority | app  | appdata
|
+--+--+---+--+--+---
--+
| 2656 | pack-01616601906 | s |4 | SIPAddHeader |
Alert-Info: alert-group | 
+--+--+---+--+--+---
--+

and in the config of the phone I have set Alert Group Ringer: to Ringer
5 but when a call comes in it's still the default ringer that rings.

I've also been lead to believe that I can set a URL for a custom
ringtone in the Alert-Info header but can't find the exact syntax
anywhere.

Does anyone have any experience of this?

Thanks

Ish
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Description: S/MIME cryptographic signature
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[asterisk-users] REINVITE with Auth Credentials has different SDP Codec

[Ujjval Karihaloo]

Hi I have a call from Service Provider (SP) to Asterisk to User

User sends a T38 REINVITE

Asterisk passes that to SP

SP challenges the INVITE

Asterisk sends INVITE with credentials but sends G711ulaw in the SDP instead of 
T38 udptl...

Obviously Fax fails..


Any ideas on how I can maintain the T38 SDP when SP challenges the mid-Call T38 
REINVITE?


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[asterisk-users] asterisk router

[steve casto]

  Looking for a router to connect to a 5/50 cable modem that works with 
Sip.  A Crisco RVS4000
installed now has real problems with Sip, one-way audio and throughput 
not up to the WAN speed.
No VPN needed, something affordable, $200-$350 US range. Every thing I 
looked at in that range had
some reported problem except   pfSense in a ATX box. Any recommendations 
or comments appreciated.
thanks
Steve Casto



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[asterisk-users] Asterisk 1.8.0 Release Candidate 3 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the third release candidate of
Asterisk 1.8.0. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

All interested users of Asterisk are encouraged to participate in the 1.8
testing process. Please report any issues found to the issue tracker,
https://issues.asterisk.org/. It is also very useful to see successful test
reports. Please post those to the asterisk-dev mailing list.

Asterisk 1.8 is the next major release series of Asterisk. It will be a Long
Term Support (LTS) release, similar to Asterisk 1.4. For more information about
support time lines for Asterisk releases, see the Asterisk versions page.

http://www.asterisk.org/asterisk-versions

With the availability of the Asterisk 1.8.0 release candidates, the binary
add-on modules for Asterisk produced by Digium have been updated with new
versions that are compatible with Asterisk 1.8. The availability of these
modules will assist with the testing of Asterisk 1.8.0 in a wider variety of
situations.

This release candidate contains fixes since the release candidate as reported by
the community. A sampling of the changes in this release candidate include:

  * Still build chan_sip even if res_crypto cannot be built (use, but not 
depend)
(Reported by a user on the mailing list. Patched by tilghman)

  * Get notifications for call files only when a file is closed, not when 
created
(Closes issue #17924. Reported by mkeuter. Patched by abeldeck)

  * Fixes to chan_gtalk to allow outbound DTMF support to work correctly. Gtalk
expects the DTMF to arrive on the RTP stream and not via jingle DTMF
signalling.
(Patched by dvossel. Tested by malcolmd)

  * Fixes to allow chan_gtalk to communicate with the Gmail web client.
(Patched by phsultan and dvossel)

  * Fix to GET DATA to allow audio to be streamed via an AGI.
(Closes issue #18001. Reported by jamicque. Patched by tilghman)

  * Resolve dnsmgr memory corruption in chan_iax2.
(Closes issue #17902. Reported by afried. Patched by russell, dvossel)

A short list of available features includes:

  * Secure RTP
  * IPv6 Support in the SIP channel driver
  * Connected Party Identification Support
  * Calendaring Integration
  * A new call logging system, Channel Event Logging (CEL)
  * Distributed Device State using Jabber/XMPP PubSub
  * Call Completion Supplementary Services support
  * Advice of Charge support
  * Much, much more!

A full list of new features can be found in the CHANGES file.

http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout

For a full list of changes in the current release candidate, please see the
ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc3

Thank you for your continued support of Asterisk!

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Re: [asterisk-users] integrate Intertel Axxess with Asterisk

[David Backeberg]

On Wed, Oct 6, 2010 at 5:00 PM, marvin horst  wrote:
> Has anyone successfully integrated Asterisk with an Inter-tel Axxess phone
> system via a SIP trunk using the IPRC card?

I have, believe it or not, integrated Asterisk with Inter-Tel.

However, not via SIP. Run the costs.

When I did, it was way cheaper to integrate asterisk with Inter-Tel
via PRI card than via SIP, especially when you figured price per
channel. I had a bunch of PRI cards on Inter-Tel talking to asterisk.
That was revision one.

Revision two, as we got bigger, I went to Cisco gear, like the 3845,
and plugged the PRIs from Inter-Tel into the Cisco gear, and used the
Cisco gear for the SIP conversion. This let asterisk talk straight SIP
and not worry about talking directly to the Inter-Tel.

We grew the Inter-Tel to as big as we could get it, offloaded as much
as we could, and eventually we couldn't fit our call center into it
anymore. Now we're Cisco for the call center.

I don't know whether what I used was called Inter-Tel Axxess. I always
just called it Inter-Tel.

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Re: [asterisk-users] ADA: DOA?

[Matt Riddell]

On 8/10/10 6:02 AM, Danny Nicholas wrote:
> FWIW, "open source" is only "truly dead" when you can't find anywhere to
> download the source.

It wasn't ever Open Source, and source was never provided.  I checked a 
while ago and that's never likely to happen, so yep (at least as of last 
time) it is a dead project.

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___

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[asterisk-users] Dahdi error

[Flavio Miranda]

Hi all,
 What hell hapen here?
asterisk:/etc/asterisk# /etc/init.d/dahdi startLoading DAHDI hardware 
modules:FATAL: Error inserting dahdi 
(/lib/modules/2.6.26-2-686/dahdi/dahdi.ko): Device or resource busy   wct4xxp: 
done   wcte12xp: done   wct1xxp: done   wcte11xp: done   wctdm24xxp: done   
wcfxo: done   wctdm: done   wcb4xxp: error   wctc4xxp: done   xpp_usb: 
doneError: missing /dev/dahdi!
When I   installed the board, everything was  going ok,but, suddenly, 
everything is going wrong
Att,

 

Flavio Roberto Miranda

MSN:flaviormira...@hotmail.com
Skype: flaviormiranda

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Re: [asterisk-users] Checking SIP Headers existence and content

[Jayson Baker]

Favorites?  voip-info.org should be your homepage.

On Thu, Oct 7, 2010 at 9:26 AM, Administrator TOOTAI wrote:

> Le 05/10/2010 05:13, VoIP Question a écrit :
> > Hello,
>
> Hi
>
> >
> > I would like to verify if a specific SIP header exists, and if yes,
> > extract the partial content from another header.
> >
> > 1. Is there a way to verify if a specific header exists?
> > 2. How do I extract data that is between the first : and the following
> > @? Specifically, The data looks like  > > and I would like to get only
> > the 1234567890
>
> Something like
>
> exten => s,1,Set(__DIALEDNUMBER=${SIP_HEADER(TO):5})
> exten => s,n,Set(__DIALEDNUMBER=${CUT(DIALEDNUMBER,@,1)})
> exten => s,n,GotoIf($["${DIALEDNUMBER:0:1}" != "+"]?numberIsOK)
> exten => s,n,Set(__DIALEDNUMBER=${CUT(DIALEDNUMBER,+,2)})
>
> Take a look here
>
> http://www.voip-info.org/wiki/view/Asterisk+func+sip_header
>
> voip-info.org should be in your favorites ;-)
>
> --
> Daniel
>
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Re: [asterisk-users] ADA: DOA?

[Ken D'Ambrosio]

On Thu, October 7, 2010 1:02 pm, Danny Nicholas wrote:

> FWIW, "open source" is only "truly dead" when you can't find anywhere to
> download the source.

I *totally* agree... if you can find me the source.  I have, at this
moment, at least, no reason to believe ADA is OSS -- indeed, looking at
it, I see no mention of the GPL (or, for that matter, any other license),
which in-and-of itself would be in direct violation of the GPL.  So I'm
thinking it's closed, closed, closed.  Crying shame.  If I'm wrong, and I
hope I am, please let me know.

In the meantime, if anyone gets it working -- correctly -- under 64-bit
Windows, please do let me know.

Thanks!

-Ken



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Re: [asterisk-users] ADA: DOA?

[Danny Nicholas]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, October 07, 2010 11:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ADA: DOA?

 

 

2010/10/7 Paul Hayes 

On 06/10/10 20:25, Ken D'Ambrosio wrote:
> Hey, all.  While ADA can still be downloaded, that's about all that I see.
>   No development, no recent mention, and -- perhaps worst of all -- it
> appears not to work properly under 64-bit systems.  So, assuming Digium's
> abandoned it, are there any suggestions of alternatives?  Right now, I'm
> replacing a Shoretel system, and I'd *dearly* love to avoid the incredibly
> fat client they have; if there's something slender -- roughly in the same
> line as ADA -- I'd be very interested, even if it's not free.


One of the most appealing feature from ADA was its ability launch calls from
MSOffice documents (smart tag feature).
Other features (screen popups and click2call were quite easy to find
elsewhere or develop yourself)



>
> Thanks,
>
> -Ken
>
>

It would seem to be a dead project yes, I can't understand why Digium
bought Click2Dial, re-branded it ADA and then stopped doing anything
with it.

I even tried to ask a Digium employee at a VoIP show in the UK about a
year ago what was going on with it but they skirted the question and
tried selling Switchvox to me (which might actually, inadvertently
answer the question ;) ).

cheers,
Paul.


FWIW, "open source" is only "truly dead" when you can't find anywhere to
download the source.

 

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[asterisk-users] Voice drop out

[Gopalakrishnan A.N]

Hi,

I am facing some voice drop in inbound, outbound, and IVR. But while
checking the process of the CPU and memory utilization is very less.

Mem: 21304K used, 36500K free, 0K shrd, 1896K buff, 13228K cached

The voice drop is in systemic. I am not too sure what to check... all the
configuration and codec are set to proper...

Anything to do with tos in sip.conf or something else... I am using Asterisk
1.2 version.

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Gops
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Re: [asterisk-users] ADA: DOA?

[Olivier]

2010/10/7 Paul Hayes 

> On 06/10/10 20:25, Ken D'Ambrosio wrote:
> > Hey, all.  While ADA can still be downloaded, that's about all that I
> see.
> >   No development, no recent mention, and -- perhaps worst of all -- it
> > appears not to work properly under 64-bit systems.  So, assuming Digium's
> > abandoned it, are there any suggestions of alternatives?  Right now, I'm
> > replacing a Shoretel system, and I'd *dearly* love to avoid the
> incredibly
> > fat client they have; if there's something slender -- roughly in the same
> > line as ADA -- I'd be very interested, even if it's not free.
>

One of the most appealing feature from ADA was its ability launch calls from
MSOffice documents (smart tag feature).
Other features (screen popups and click2call were quite easy to find
elsewhere or develop yourself)


>
> > Thanks,
> >
> > -Ken
> >
> >
>
> It would seem to be a dead project yes, I can't understand why Digium
> bought Click2Dial, re-branded it ADA and then stopped doing anything
> with it.
>
> I even tried to ask a Digium employee at a VoIP show in the UK about a
> year ago what was going on with it but they skirted the question and
> tried selling Switchvox to me (which might actually, inadvertently
> answer the question ;) ).
>
> cheers,
> Paul.
>
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Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

[Daniel Tryba]

On Thu, Oct 07, 2010 at 02:57:27PM +0200, Jonas Kellens wrote:
> nat=yes is set as a global parameter and also in the realtime MySQL 
> sip_buddies database I have for every peer nat=yes.
> 
> I then find it very strange that when placing these Snom phones in my 
> environment (for configuration) work very well, and then when I hook 
> them up at the site there is trouble with nat. I'm also behind nat here...

I have never seen this problem before with Snom M3 and different routers
(Linux/Cisco or stupid SpeedTouches) without any
connection NAT helpers for SIP enabled. I'd say you should try the
difference values for nat to see if one works with the NAT gateway or
use STUN like suggested elsewhere.

-- 

   Daniel Tryba

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Re: [asterisk-users] AMI getting related channels in Ringing state

[Daniel Tryba]

On Wed, Oct 06, 2010 at 01:56:55PM +0200, Daniel Tryba wrote:
> Issuing the AMI Status command results in a list of active channels. But
> how to figure out which channels are related before the call is
> answered? 

Anybody?

My workaround for this problem is setting a persistent variable in the 
incoming channel:
Set(__xuniqueid=${RAND(10,99)}-${CALLERID(num)}-${EXTEN})
and match related channels this on the same xuniqueid variable.
To bad * doesn't set such a variable itself, I now have to make sure any
originating call executes this function.

-- 

   Daniel Tryba

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Re: [asterisk-users] RTP Read too short

[Kevin P. Fleming]

On 10/07/2010 10:36 AM, Bryant Zimmerman wrote:
> Hi All
> 
> In the console I am seeing warring rtp.c:1635 ast_rtp_read: RTP Read too
> short
> 
> I get these all of the time things seem to be working fine but I am
> trying to figure out if there is a way to resolve these Warnings.
> I am running asterisk 1.6.2.13

The way to resolve them is to have whatever device is sending your
system invalid RTP packets stop doing so.

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Re: [asterisk-users] SIP authentication - Thoughts please

[Steve Davies]

On 7 October 2010 10:10, Stefan Schmidt  wrote:
> Am 07.10.10 10:52, schrieb Steve Davies:
>> Hi,
>>
> 
>
> Hello,
>
> i just want to say something about point 4 which comes to my mind about
> security.
>
>>
>> 4) I am not sure whether it is worth dropping through and testing auth
>> against other peers if there is no username match. Can auth ever
>> succeed under those circumstances (password matches, but not
>> username?)
>
> If you use UDP its very easy to fake the source ip of a call so do you
> really want to open a door to an attacker by authenticate only by ip and
> passwort which can match to any peer with the same ip adress? To
> bruteforce this would be much easier than to bruteforce against sending
> IP, right username and right password.

I was not clear. By option 4) I intended that you test the password
against other peers with a matching IP address. I am not sure whether
the username is included in the SIP password hash, so do not know
whether there is even any point in doing so. As far as I can tell, in
the EXISTING sip stack, digest username is not used to determine which
peer to authenticate with, it just uses the first peer with a matching
IP.

> Have you tried to use different ports to register? i think this could help.

AFAIK, Asterisk will only operate on one port, and the remote end is a
major ITSP who will not be wanting to listen to me making odd requests
:)

Thanks for the feedback!

Regards,
Steve

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Re: [asterisk-users] ADA: DOA?

[Paul Hayes]

On 06/10/10 20:25, Ken D'Ambrosio wrote:
> Hey, all.  While ADA can still be downloaded, that's about all that I see.
>   No development, no recent mention, and -- perhaps worst of all -- it
> appears not to work properly under 64-bit systems.  So, assuming Digium's
> abandoned it, are there any suggestions of alternatives?  Right now, I'm
> replacing a Shoretel system, and I'd *dearly* love to avoid the incredibly
> fat client they have; if there's something slender -- roughly in the same
> line as ADA -- I'd be very interested, even if it's not free.
>
> Thanks,
>
> -Ken
>
>

It would seem to be a dead project yes, I can't understand why Digium 
bought Click2Dial, re-branded it ADA and then stopped doing anything 
with it.

I even tried to ask a Digium employee at a VoIP show in the UK about a 
year ago what was going on with it but they skirted the question and 
tried selling Switchvox to me (which might actually, inadvertently 
answer the question ;) ).

cheers,
Paul.

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[asterisk-users] RTP Read too short

[Bryant Zimmerman]

Hi All 

In the console I am seeing warring rtp.c:1635 ast_rtp_read: RTP Read too 
short

I get these all of the time things seem to be working fine but I am trying 
to figure out if there is a way to resolve these Warnings.
I am running asterisk 1.6.2.13

Any direction is appreciated.

Thanks
Bryant

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[asterisk-users] How to change features.conf's atxfer dialing tone ?

[Olivier]

Hi,

I'm facing the following request :
"When someone is starting an assisted transfer using Asterisk's features
codes, he will ear a prompt saying "Transfer" and then a dialing inviting
him to dial the number he tries to reach.
This tone volume is qualified as a bit too load."

Is it possible to change that and have a more delicate volume ?

A quick look inside features.conf doesn't show evidence.

Regards
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Re: [asterisk-users] Checking SIP Headers existence and content

[Administrator TOOTAI]

Le 05/10/2010 05:13, VoIP Question a écrit :
> Hello,

Hi

>
> I would like to verify if a specific SIP header exists, and if yes, 
> extract the partial content from another header.
>
> 1. Is there a way to verify if a specific header exists?
> 2. How do I extract data that is between the first : and the following 
> @? Specifically, The data looks like  > and I would like to get only 
> the 1234567890

Something like

exten => s,1,Set(__DIALEDNUMBER=${SIP_HEADER(TO):5})
exten => s,n,Set(__DIALEDNUMBER=${CUT(DIALEDNUMBER,@,1)})
exten => s,n,GotoIf($["${DIALEDNUMBER:0:1}" != "+"]?numberIsOK)
exten => s,n,Set(__DIALEDNUMBER=${CUT(DIALEDNUMBER,+,2)})

Take a look here

http://www.voip-info.org/wiki/view/Asterisk+func+sip_header

voip-info.org should be in your favorites ;-)

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[asterisk-users] convert g729A-g729B and vice-versa

[Harel Cohen]

Hi all.
Is there a free, or at least non-expensive, solution that can convert g729A 
<-->g729B (with VAD)? The no-support for g729B on Asterisk gives me a BIG 
headache…
Thanks,
Harel


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Re: [asterisk-users] Alert-Info advice

[Rizwan Hisham]

I use the following syntax for sipura i think, and it works fine for me.

exten=> s,1010,SipAddHeader(Alert-Info: )
exten=> s,1020,SipAddHeader(Alert-Info: )
exten=> s,1030,SipAddHeader(Alert-Info: )
exten=> s,1040,SipAddHeader(Alert-Info: )
exten=> s,1050,SipAddHeader(Alert-Info: )

On Wed, Sep 29, 2010 at 12:38 PM, Philipp von Klitzing <
klitz...@pool.informatik.rwth-aachen.de> wrote:

> Hi!
>
> > Just out of interest, have you ever got this working?
>
> Yes, sure.
>
> >  Mine just isn't but I'm starting to think that my mp3 to 8000Hz Mono
> > 16 bit wav files is a bit dodgy
>
> Very well possible. Also look at the individual "identity x"
> configuration and consider to select "Custom ringtone", then enter the
> URL for the wav file in question. That is a good and easy way to test if
> that specifc wav file actually works - also check the http log and/or the
> snom log.
>
> Philipp
>
>
> --
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Re: [asterisk-users] Polycom: full caller ID?

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio
Sent: Thursday, October 07, 2010 9:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom: full caller ID?

Hi, all.  When I get calls on my SoundPoints, I only see the number -- is
there a way to get the alpha portion of the CID, as well?

Thanks!

-Ken

I'm guessing that you could set fullname in users.conf and correct this.
CallerID is a hit-or-miss proposition as documented elsewhere in this list.
Some TELCO's are very good (201-555-1212 Harry's wig Emporium) and others
not so (201-555-1212 Unknown caller) or (201-555-1212 Cell phone).  You can
address this internally for your most frequent callers.  You flip a coin on
"cold" calls.


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Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

[Jonas Kellens]

On 10/07/2010 04:18 PM, Philipp von Klitzing wrote:
> Hi!
>
>
>> I'm having difficulty with registering a SIP account in a Snom 320 IP-
>> phone.
>>  
> Do a SIP trace on your SNOM phone, and you will most probably see that
> the 401 reply of Asterisk does not arrive on the phone. Then check your
> STUN/ICE settings on the phone in combination with the nat= settings in
> sip.conf on Asterisk.
>
> BTW: Are you happy with firmware 8.4.18? I still stick to 7.3.30 for the
> 3xx models.
>
> Philipp
>

Hello,

I don't really need the version 8 firmware, but I install it because it 
is the most recent...

The SIP trace of the SNOM indeed does not show any answer to the 
REGISTER being sent (the SIP trace consists of only REGISTER-statements...)


I have no STUN/ICE settings filled in.

Is it possible I need STUN/ICE for the Snom 320 but not for the Zoiper 
softphone ???

The strange thing in this story is : Snom 320 receives no 200OK, but 
registration with the Zoiper softphone on a computer on the same LAN 
(connected to the PC port of the SNOM) works fine !!


Jonas.


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[asterisk-users] Polycom: full caller ID?

[Ken D'Ambrosio]

Hi, all.  When I get calls on my SoundPoints, I only see the number -- is
there a way to get the alpha portion of the CID, as well?

Thanks!

-Ken


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Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

[Philipp von Klitzing]

Hi!

> I'm having difficulty with registering a SIP account in a Snom 320 IP-
> phone.

Do a SIP trace on your SNOM phone, and you will most probably see that 
the 401 reply of Asterisk does not arrive on the phone. Then check your 
STUN/ICE settings on the phone in combination with the nat= settings in 
sip.conf on Asterisk.

BTW: Are you happy with firmware 8.4.18? I still stick to 7.3.30 for the 
3xx models.

Philipp


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Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

[Jonas Kellens]

On 10/07/2010 02:36 PM, Daniel Tryba wrote:
> On Thu, Oct 07, 2010 at 02:24:59PM +0200, Daniel Tryba wrote:
>
>>> It's the same account, the same password, but other agent.
>>>
>>> Can anyone help me with this please ?! I see no difference but there
>>> must be !!
>>>
>> The difference is the SNOM is using rport and Zoiper isn't. Is nat for
>> this client set to 'yes' or something else?
>>  

Even when I mark the option in the Zoiper softphone to use rport, the 
registration still goes fine... With or without rport... Registration 
succeeds. But meanwhile, the Snom is still not registering...


Jonas.

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Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

[Jonas Kellens]

On 10/07/2010 02:36 PM, Daniel Tryba wrote:
> On Thu, Oct 07, 2010 at 02:24:59PM +0200, Daniel Tryba wrote:
>
>>> It's the same account, the same password, but other agent.
>>>
>>> Can anyone help me with this please ?! I see no difference but there
>>> must be !!
>>>
>> The difference is the SNOM is using rport and Zoiper isn't. Is nat for
>> this client set to 'yes' or something else?
>>  
> Wanted to say more before sending:
> this might indicate a broken nat sip helper so the responses never get
> to the snom. You should check the inside traffic to the snom.
>

This is a REGISTER from the sip trace of the Snom 320 IP-phone :

REGISTER sip:sip.domain.tld SIP/2.0
Via: SIP/2.0/UDP 192.168.114.200:2049;branch=z9hG4bK-od9b7c0ekamp;rport
From: ;tag=vub2zjtmv2
To: 
Call-ID: 3c26701f88d8-6i37fwkca22u
CSeq: 46 REGISTER
Max-Forwards: 70
Contact: 
;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom320";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
 

User-Agent: snom320/8.4.18
Allow-Events: dialog
X-Real-IP: 192.168.114.200
Supported: path, gruu
Expires: 3600
Content-Length: 0


and indeed there is no ACK or 200 OK to this REGISTER...

Kind regards,

Jonas.

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Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

[Jonas Kellens]

On 10/07/2010 02:36 PM, Daniel Tryba wrote:
> On Thu, Oct 07, 2010 at 02:24:59PM +0200, Daniel Tryba wrote:
>
>>> It's the same account, the same password, but other agent.
>>>
>>> Can anyone help me with this please ?! I see no difference but there
>>> must be !!
>>>
>> The difference is the SNOM is using rport and Zoiper isn't. Is nat for
>> this client set to 'yes' or something else?
>>  
> Wanted to say more before sending:
> this might indicate a broken nat sip helper so the responses never get
> to the snom. You should check the inside traffic to the snom

And I wanted to add that it is possible to make outbound calls, but 
because the phone is not registered receiving calls is not possible...


Jonas.

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Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

[Jonas Kellens]

Hello !

Thank you very much for your quick answer !!

nat=yes is set as a global parameter and also in the realtime MySQL 
sip_buddies database I have for every peer nat=yes.


I then find it very strange that when placing these Snom phones in my 
environment (for configuration) work very well, and then when I hook 
them up at the site there is trouble with nat. I'm also behind nat here...


I do not find an rport-parameter in the snom's webinterface...


Jonas.


On 10/07/2010 02:24 PM, Daniel Tryba wrote:

On Thu, Oct 07, 2010 at 01:54:58PM +0200, Jonas Kellens wrote:
   

It's the same account, the same password, but other agent.

Can anyone help me with this please ?! I see no difference but there
must be !!
 

The difference is the SNOM is using rport and Zoiper isn't. Is nat for
this client set to 'yes' or something else?

   
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Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

[Daniel Tryba]

On Thu, Oct 07, 2010 at 02:24:59PM +0200, Daniel Tryba wrote:
> > It's the same account, the same password, but other agent.
> > 
> > Can anyone help me with this please ?! I see no difference but there 
> > must be !!
> 
> The difference is the SNOM is using rport and Zoiper isn't. Is nat for
> this client set to 'yes' or something else?

Wanted to say more before sending:
this might indicate a broken nat sip helper so the responses never get
to the snom. You should check the inside traffic to the snom.

-- 

   Daniel Tryba

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Re: [asterisk-users] Difference

[Rizwan Hisham]

Thanks for sharing all of your thoughts and information. If anyone knows a
good article about asterisk 1.8 then please let me know about it. I have
read the presentation by Kevin Fleming but more information is always good.

Cheers

On Wed, Oct 6, 2010 at 10:28 AM, Miguel Molina wrote:

>  I find 1.6.2.13 version is stable for trunk call routing, and it should be
> too for basic call center use. The asterisk team has made some architectural
> improvements (moving to astobj2 a lot of internal structures, and much more
> you may not see from a user perspective) but given the several environment
> and different use cases, fear to upgrade or proven 1.4 stability for the
> job, the people usually don't upgrade or make it slowly with a lot of
> previous tests before making the jump.
>
> If you use FAX, I recommend you 1.6.2 or later. The app_fax module is far
> better than the ast-agx-addons for 1.4.
>
> The good old (now unsupported) 1.2 works for many people, ask Steve.
>
> So it's up to you.
>
> Cheers,
>
> --
> Ing. Miguel Molina
> Grupo de Tecnología
> Millenium Phone Center
>
> El 06/10/10 11:04, Zeeshan Zakaria escribió:
>
> For a production environment, 1.4 is the most stable, and it has everything
> one needs to setup a telecom platform. As per my understanding 1.6 never got
> the same recognition for stability as 1.4, plus it doesn't have any
> significant advantages over 1.4. The newer version 1.8 series might be my
> next jump once it'll be out of beta, but at this time it should not be used
> in a production environment. Many of us still use 1.4 in production and if
> you are just starting, this'll be your best choice.
>
> Zeeshan A Zakaria
>
> --
> www.ilovetovoip.com
>
> On 2010-10-06 11:54 AM, "Danny Nicholas"  wrote:
>
>   From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Be...
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan Hisham
> *Sent:* Wednesday, October 06, 2010 10:44 AM
>
>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Difference
>
>
>
>
>
> Is there any major architectural difference between 1.4 and 1.8?
>
> The dialplan uses the 1.6 nomenclature (delimiter in dialplan changes from
> , to |) and the AGI structure is enhanced.  If you don’t use AGI’s, a
> qualified “not really”.
>
> --
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>
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Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

[Daniel Tryba]

On Thu, Oct 07, 2010 at 01:54:58PM +0200, Jonas Kellens wrote:
> It's the same account, the same password, but other agent.
> 
> Can anyone help me with this please ?! I see no difference but there 
> must be !!

The difference is the SNOM is using rport and Zoiper isn't. Is nat for
this client set to 'yes' or something else?

-- 

   Daniel Tryba

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[asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

[Jonas Kellens]

Hello,

I'm having difficulty with registering a SIP account in a Snom 320 
IP-phone. This is what sip debug tells me :



[Oct  7 13:28:42] VERBOSE[20314] chan_sip.c: [Oct  7 13:28:42]
<--- SIP read from UDP:public_ip:58697 --->
REGISTER sip:sip.domain.tld SIP/2.0
Via: SIP/2.0/UDP 192.168.114.200:2048;branch=z9hG4bK-vj1xvbdnp4dw;rport
From: ;tag=sd2b3o74zc
To: 
Call-ID: 3c28a76e73cf-gp9nioi8zdci
CSeq: 12 REGISTER
Max-Forwards: 70
Contact: 
;reg-id=1;q=1.0;+sip.instance="";audio;mobility="fix

ed";duplex="full";description="snom320";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom320/8.4.18
Allow-Events: dialog
X-Real-IP: 192.168.114.200
Supported: path, gruu
Expires: 3600
Content-Length: 0


<->
[Oct  7 13:28:42] VERBOSE[20314] chan_sip.c: [Oct  7 13:28:42] --- (14 
headers 0 lines) ---
[Oct  7 13:28:42] VERBOSE[20314] chan_sip.c: [Oct  7 13:28:42] Sending 
to 192.168.114.200 : 2048 (no NAT)

[Oct  7 13:28:42] VERBOSE[20314] chan_sip.c: [Oct  7 13:28:42]
<--- Transmitting (NAT) to public_ip:58697 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
192.168.114.200:2048;branch=z9hG4bK-vj1xvbdnp4dw;received=public_ip;rport=58697

From: ;tag=sd2b3o74zc
To: ;tag=as6108a7e2
Call-ID: 3c28a76e73cf-gp9nioi8zdci
CSeq: 12 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="domain.tld", nonce="398aee1e"
Content-Length: 0


I would expect the Snom to try a second register, this time with some 
type of nonce. But there is just 1 REGISTER and 1 Unauthorized and 
that's it...


Other Snom phones with SIP-accounts go very well, but at this location 
the registration fails.


Another remark : when using a Zoiper softphone, the registration goes 
very well :



REGISTER sip:sip.domain.tld;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 
192.168.114.20:5060;branch=z9hG4bK-d8754z-fab4a5effbf90a05-1---d8754z-

Max-Forwards: 70
Contact: 


To: 
From: ;tag=db1a5018
Call-ID: NzBlZDMyN2U0YTEzZDk4Y2M2N2NmNzMxYTk4OWUxYTY.
CSeq: 1 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, 
SUBSCRIBE

User-Agent: Zoiper rev.7797
Allow-Events: presence, kpml
Content-Length: 0


<->
[Oct  7 13:46:52] VERBOSE[20314] chan_sip.c: [Oct  7 13:46:52] --- (13 
headers 0 lines) ---
[Oct  7 13:46:52] VERBOSE[20314] chan_sip.c: [Oct  7 13:46:52] Sending 
to 192.168.114.20 : 5060 (no NAT)

[Oct  7 13:46:52] VERBOSE[20314] chan_sip.c: [Oct  7 13:46:52]
<--- Transmitting (NAT) to public_ip:51363 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
192.168.114.20:5060;branch=z9hG4bK-d8754z-fab4a5effbf90a05-1---d8754z-;received=public_ip

From: ;tag=db1a5018
To: ;tag=as2fcfde3c
Call-ID: NzBlZDMyN2U0YTEzZDk4Y2M2N2NmNzMxYTk4OWUxYTY.
CSeq: 1 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="domain.tld", nonce="7833b268"
Content-Length: 0

REGISTER sip:sip.domain.tld;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 
192.168.114.20:5060;branch=z9hG4bK-d8754z-fdd59e394f9c23b9-1---d8754z-

Max-Forwards: 70
Contact: 


To: 
From: ;tag=db1a5018
Call-ID: NzBlZDMyN2U0YTEzZDk4Y2M2N2NmNzMxYTk4OWUxYTY.
CSeq: 2 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, 
SUBSCRIBE

User-Agent: Zoiper rev.7797
Authorization: Digest 
username="test3",realm="domain.tld",nonce="7833b268",uri="sip:sip.domain.tld;transport=UDP",response="198f6262248fb11fe6cb55408a1cb8ce",algorithm=MD5

Allow-Events: presence, kpml
Content-Length: 0


SIP/2.0 200 OK
Via: SIP/2.0/UDP 
192.168.114.20:5060;branch=z9hG4bK-d8754z-fdd59e394f9c23b9-1---d8754z-;received=public_ip

From: ;tag=db1a5018
To: ;tag=as2fcfde3c
Call-ID: NzBlZDMyN2U0YTEzZDk4Y2M2N2NmNzMxYTk4OWUxYTY.
CSeq: 2 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: 
;expires=60

Date: Thu, 07 Oct 2010 11:46:52 GMT
Content-Length: 0


It's the same account, the same password, but other agent.

Can anyone help me with this please ?! I see no difference but there 
must be !!



Kind regards,
Jonas.
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Re: [asterisk-users] How to test BRI lines energy saving mode ?

[Andrew Thomas]

Well, to go slightly O/T:

If you read the issue tracker for 17270 - it appears to be a LibPri
'fault'.  So I would say that the main work would need to be in LibPri
.

Maybe someone who knows LibPri and DAHDI better can explain how the two
combine...

Cheers
Andy


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: 07 October 2010 11:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to test BRI lines energy saving mode ?




2010/10/7 Andrew Thomas 

The D-channel isn't actually 'dropped' - it is put in to a 'power-save'
state.

See http://www.isdn-test.de/ihhe12.htm#Heading37 and scroll down to
'Activation / Deactivation' for more information.

Anyway - this is a known 'problem' -
https://issues.asterisk.org/view.php?id=17270

As there is no fix for the above - then I doubt * will be able to
emulate the NT's function.




Thanks for these interesting links !

So this Activation/Desactivation feature seems to be missing in
Asterisk.
Would you say it should be implemented in libpri, in dahdi, or both ?


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you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
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are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
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Re: [asterisk-users] Problems compiling Asterisk on Debian

[Tzafrir Cohen]

On Mon, Sep 27, 2010 at 06:21:16PM +0200, Danny Dias wrote:
> Thanks Dean,
> 
> I've done it before, that's why i'm here asking :( take a look:
> 
> r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# apt-cache search
> linux-headers-$(uname -r)
> linux-headers-2.6.26-2-amd64 - Header files for Linux 2.6.26-2-amd64
> r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# sudo apt-get install
> linux-headers-$(uname -r)
> Reading package lists... Done
> Building dependency tree
> Reading state information... Done
> linux-headers-2.6.26-2-amd64 is already the newest version.
> 0 upgraded, 0 newly installed, 0 to remove and 0 not upgraded.
> r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# make
> echo "You do not appear to have the sources for the 2.6.26-2-amd64 kernel
> installed."
> You do not appear to have the sources for the 2.6.26-2-amd64 kernel
> installed.
> exit 1
> make: *** [modules] Error 1

This error basically means:

  no such file /lib/modules/`uname -r`/build/.config

'build' above is a symlink. If you have linux-headers-2.6.26-2-amd64 ,
it is supposed to point from /lib/modules/2.6.26-2-amd64/build to
/usr/src/linux-headers-2.6.26-2-amd64 .

Is that the case?

-- 
   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] How to test BRI lines energy saving mode ?

[Olivier]

2010/10/7 Andrew Thomas 

> The D-channel isn't actually 'dropped' - it is put in to a 'power-save'
> state.
>
> See http://www.isdn-test.de/ihhe12.htm#Heading37 and scroll down to
> 'Activation / Deactivation' for more information.
>
> Anyway - this is a known 'problem' -
> https://issues.asterisk.org/view.php?id=17270
>
> As there is no fix for the above - then I doubt * will be able to
> emulate the NT's function.
>


Thanks for these interesting links !

So this Activation/Desactivation feature seems to be missing in Asterisk.
Would you say it should be implemented in libpri, in dahdi, or both ?
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Re: [asterisk-users] SIP authentication - Thoughts please

[Stefan Schmidt]

Am 07.10.10 10:52, schrieb Steve Davies:
> Hi,
> 


Hello,

i just want to say something about point 4 which comes to my mind about
security.

> 
> 4) I am not sure whether it is worth dropping through and testing auth
> against other peers if there is no username match. Can auth ever
> succeed under those circumstances (password matches, but not
> username?)

If you use UDP its very easy to fake the source ip of a call so do you
really want to open a door to an attacker by authenticate only by ip and
passwort which can match to any peer with the same ip adress? To
bruteforce this would be much easier than to bruteforce against sending
IP, right username and right password.

Have you tried to use different ports to register? i think this could help.


> Regards,
> Steve
> 
best regards

Stefan

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[asterisk-users] SIP authentication - Thoughts please

[Steve Davies]

Hi,

We have a scenario where we need multiple discrete SIP trunks (peers)
from/to a single endpoint. Because the authentication system starts by
matching IP address, it only ever matches on one of the SIP peer
entries, and ignores the others. This is documented behaviour and as
such is "correct". I would like to propose an extension to how SIP
peers are authenticated in this case:

1) Initial INVITE arrives, scan the list of all matching peer IP addresses.
   If a peer with no password is found, proceed with that peer immediately.

2) ...otherwise, a password is required, so send 407 challenge

3) INVITE arrives with Basic-Auth.
   Scan for /all/ matching peers based on IP address. If one of the
matching peers has a section name matching the Basic-Auth username,
use it to proceed.

4) I am not sure whether it is worth dropping through and testing auth
against other peers if there is no username match. Can auth ever
succeed under those circumstances (password matches, but not
username?)

Thanks for any feedback.

Regards,
Steve

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[asterisk-users] Fw: asterisk > cisco gateway > westell > isdx

[Damian Turburville]

Anyone?



- Forwarded Message 


Hi,
I am hoping someone can help me with a problem I am having.
I am trying to setup a connection from an Elastix 2 server to a Siemens isdx 
PBX. The setup is as follows

Elastix 2
*sip trunk*
Cisco 2621XM router with 2 E1 voice interfaces
*QSIG*
Westell IQ2000 protocol convertor
*DPNSS*
Siemens ISDX

So the Elastix box has a SIP trunk to the cisco router which then talks QSIG to 
the Westell which converts it to DPNSS to talk to the ISDX.
I have managed to make a call from a phone on Elastix to a phone on the ISDX 
but 
it drops after about 3 seconds, every time. Would anyone have any idea why this 
is? 

Here is the setup I have on Elastix and the Cisco router

Elastix SIP trunk PEER  details
type=friend
qualify=no
nat=no
insecure=very
host=10.132.41.13
dtmfmode=rfc2833
context=from-internal
canreinvite=yes
disallow=all
allow=ulaw

relevant Cisco config
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
boot-start-marker
boot-end-marker
!
no network-clock-participate slot 1
no network-clock-participate wic 0
ip cef
!
!
no ip domain lookup
isdn switch-type primary-qsig
voice-card 1
!
!
voice rtp send-recv
!
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
 sip
  bind control source-interface FastEthernet0/0
  bind media source-interface  FastEthernet0/0
!
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8 bytes 40
!
!
controller E1 1/0
!
controller E1 1/1
 pri-group timeslots 1-31
!
!
class-map match-all FAX
  description Match T.38 Fax
 match access-group name FAX
!
interface FastEthernet0/0
 ip address 10.132.41.13 255.255.255.0
 no ip mroute-cache
 speed 100
 full-duplex
!
interface FastEthernet0/1
 no ip address
 duplex auto
 speed auto
!
interface Serial1/1:15
 no ip address
 encapsulation hdlc
 no logging event link-status
 isdn switch-type primary-qsig
 isdn overlap-receiving
 isdn incoming-voice voice
 no cdp enable
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 10.132.41.1
!
ip http server
!
control-plane
!
voice-port  1/1:15
!
!
dial-peer voice 786 voip
 huntstop
 destination-pattern 56...
 progress_ind setup enable 3
 voice-class codec 1
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 2 pots
 description QSIG Link to PABX via Westell
 destination-pattern 2
 progress_ind setup enable 3
 progress_ind alert enable 8
 direct-inward-dial
 port 1/1:15
 forward-digits all
!
dial-peer voice 3 pots
 description Featurenet 7xx routing
 huntstop
 destination-pattern 7T
 progress_ind setup enable 3
 progress_ind alert enable 8
 incoming called-number 786
 no digit-strip
 direct-inward-dial
 forward-digits all
!
gateway
!
sip-ua
 registrar ipv4:10.133.40.50 expires 3600
 sip-server  ipv4:10.133.40.50

Any help would be greatly appreciated
Thanks,
DT


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Re: [asterisk-users] How to test BRI lines energy saving mode ?

[Andrew Thomas]

The D-channel isn't actually 'dropped' - it is put in to a 'power-save'
state.

See http://www.isdn-test.de/ihhe12.htm#Heading37 and scroll down to
'Activation / Deactivation' for more information.

Anyway - this is a known 'problem' -
https://issues.asterisk.org/view.php?id=17270

As there is no fix for the above - then I doubt * will be able to
emulate the NT's function.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle Giese
Sent: 07 October 2010 01:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to test BRI lines energy saving mode ?


Olivier wrote:
> Hello,
>
> If my understanding is correct, these days it seems that many ISDN BRI

> lines are configured in energy saving mode in which signalling 
> D-channel is "dropped" until a new call comes in.
>
> Is it possible to replicate this behaviour with Asterisk (when 
> Asterisk is in NT mode and is seen as a public ISDN by another PBX, 
> for instance) ? If not, would you it would be a useful addition to 
> Asterisk ?
>
> Regards
>
>
Energy saving???  I don't think so. 

If the D channel is down, how would I make an outgoing phone call? 
Something in this mode or your explanation just does not sound right...

Lyle Giese
LCR Computer Services, Inc.


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 If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments. 

Registered in England. No. 27459085.



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Re: [asterisk-users] How to test BRI lines energy saving mode ?

[Olivier]

2010/10/7 Lyle Giese 

> Olivier wrote:
> > Hello,
> >
> > If my understanding is correct, these days it seems that many ISDN BRI
> > lines are configured in energy saving mode in which signalling
> > D-channel is "dropped" until a new call comes in.
> >
> > Is it possible to replicate this behaviour with Asterisk (when
> > Asterisk is in NT mode and is seen as a public ISDN by another PBX,
> > for instance) ?
> > If not, would you it would be a useful addition to Asterisk ?
> >
> > Regards
> >
> >
> Energy saving???  I don't think so.
>
> If the D channel is down, how would I make an outgoing phone call?
> Something in this mode or your explanation just does not sound right...
>

I'm far from fluent on ISDN signalling but I've read several times telco use
some energy saving mode.
For instance here:
https://issues.asterisk.org/view.php?id=14031

Maybe the words "D-channel is "dropped"" are misleading in this context but
the fact remains if telco are using some kind "energy saving mode", how can
we reproduce it ?

Cheers



> Lyle Giese
> LCR Computer Services, Inc.
>
>
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