Re: [asterisk-users] zaptel

[Louwrens Benadé]

 

 

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of gilbert
saunders
Sent: 23 April 2008 08:18 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] zaptel

 

hi i been installing asterisk on a new pc...i donwloaded
asterisk-1.6.0-beta7 ,  zaptel-1.4.10 and libpri-1.6.0-beta1 everthing
install and configure perfectly but zaptel gives me errors adn when i reboot
my system it says that no devices are configured...i have a sangoma a200
card

 

aren’t you forgetting about wanpipe?

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Re: [asterisk-users] Parsing incoming extension till first @

[Ali Jawad]

Thanks for the hint Patrick I appreciate it.

On Tue, Apr 22, 2008 at 3:02 PM, Rob Hillis <[EMAIL PROTECTED]> wrote:
> Using _. is going to result in warnings.  A much better practice is to
>  use _X.
>
>
>
>  Ali Jawad wrote:
>  > Thx again patrick it worked, I used
>  >
>  > [google-in]
>  > exten => _.,1,Set(dst=${CUT(EXTEN,@,1)})
>  > exten => _.,1,Dial(SIP/[EMAIL PROTECTED])
>  >
>  > while it should have been
>  >
>  > [google-in]
>  > exten => _.,1,Set(dst=${CUT(EXTEN,@,1)})
>  > exten => _.,2,Dial(SIP/[EMAIL PROTECTED])
>  >
>
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> >
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-- 
-- 
With Regards
Ali Jawad System Administrator
http://www.alijawad.org
Phone : +961-01-559031
Mobile : +961-03-041705





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[asterisk-users] zaptel

[gilbert saunders]

hi i been installing asterisk on a new pc...i donwloaded asterisk-1.6.0-beta7 , 
 zaptel-1.4.10 and libpri-1.6.0-beta1 everthing install and configure perfectly 
but zaptel gives me errors adn when i reboot my system it says that no devices 
are configured...i have a sangoma a200 card
   
-
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[asterisk-users] prepaid on the trunks

[Nhadie Ramos]

if i have this setup:

[sip users] -- [asterisk] --- [as5300] --- [pstn]

asterisk will talk to as5300 using sip. i will use as5300 as a trunk on the 
asterisk so sip users can call out to pstn.

what i would like to is do prepaid on those trunks, not on the sip users. sip 
users can call any other sip users . i want to do it that way coz i'm trying to 
build a multi-tenant pbx, and i will use the trunk as a unique identifier for 
each customer not their local extension.

e.g. customer A will have trunk 34587612 and will have extension 101 and 102,  
customer B will have a different trunk 87659043 but will also have the 
extension 101 and 102.

i want to create a billing system to monitor only the trunks and also to load 
amounts on those trunks. is this possible? will i be able to use app_prepaid 
for this?

thank you.
regards,
nhadie




   
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Re: [asterisk-users] AST-2008-006 - 3-way handshake in IAX2 incomplete

[Matt Watson]

I can;t imagine what headaches you'd have going from 1.4.11 to 1.4.19.1... that 
is a minor version upgrade... no real change in functionality thats 
basically 8 versions of bug fixes... if you just apply the IAX2 patch, you'll 
be fixing 1 out of probably a hundreds of bugs.  Going from 1.4.x to 1.6.x 
however... you'd run into some headaches probably... but if you are staying in 
the 1.4 series you shouldn;t have any problems... worst case is if its broke 
you just make install your 1.4.11 overtop of 1.4.19.1 to revert back.

--
Matt

From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Brian J. Murrell [EMAIL 
PROTECTED]
Sent: Tuesday, April 22, 2008 8:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AST-2008-006 - 3-way handshake in IAX2
incomplete

On Tue, 2008-04-22 at 17:58 -0500, Security Officer wrote:
> Asterisk Project Security Advisory - AST-2008-006

So given that I'm new to asterisk's svn and bug tracking tool, is it
sufficient then to apply the two patches (iax_dcallno_check-1.2.rev3.txt
and iax_dcallno_check.rev9.txt) listed in
http://bugs.digium.com/view.php?id=10078 to a 1.4.11ish release to
correct this vulnerability?  I really don't feel like buying into
any/all of the headaches that went into 1.4.11->1.4.20.  You know, "if
it ain't broke don't fix it", and my corollary, "if it is broke, only
fix what's broke, don't try to make it better".  :-)

Thanx,
b.


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Re: [asterisk-users] Parsing incoming extension till first @

[Rob Hillis]

Using _. is going to result in warnings.  A much better practice is to 
use _X.

Ali Jawad wrote:
> Thx again patrick it worked, I used
>
> [google-in]
> exten => _.,1,Set(dst=${CUT(EXTEN,@,1)})
> exten => _.,1,Dial(SIP/[EMAIL PROTECTED])
>
> while it should have been
>
> [google-in]
> exten => _.,1,Set(dst=${CUT(EXTEN,@,1)})
> exten => _.,2,Dial(SIP/[EMAIL PROTECTED])
>
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> !DSPAM:480e5326213018190740810!
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>   

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[asterisk-users] question about asterisk setup...

[Jorge L. Vazquez]

Hello guys.I'm new to asterisk, and I have a setup which * is running
behind a firewall with 3 softphones installed on different computers,
now the softphones can connect with no problem I can also make calls
outside my network to ppl with ekiga and gizmo accounts, but my question
is. can I receive calls with the setup I have from ppl with ekiga or
gizmo accounts, without the need for a service provider?... if so could
someone give me a hand getting this setup working?
 
Thanks
Jorge
 
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Re: [asterisk-users] AST-2008-006 - 3-way handshake in IAX2 incomplete

[Brian J. Murrell]

On Tue, 2008-04-22 at 20:34 -0400, Brian J. Murrell wrote:
> 
> So given that I'm new to asterisk's svn and bug tracking tool, is it
> sufficient then to apply the two patches (iax_dcallno_check-1.2.rev3.txt
> and iax_dcallno_check.rev9.txt)

Ahhh.  I see.  These must be two versions of the same patch.  One for
1.2 and one for 1.4 and 1.6?  Is that how it's working?  I must say that
the -1.2 patch appears to be closer to 1.4 than the "rev9" patch though.

In any case, I think it's pretty clear between both of these two
patches, what fixes to chan_iax2.c need to be made.

Cheers,
b.



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Re: [asterisk-users] AST-2008-006 - 3-way handshake in IAX2 incomplete

[Brian J. Murrell]

On Tue, 2008-04-22 at 17:58 -0500, Security Officer wrote:
> Asterisk Project Security Advisory - AST-2008-006

So given that I'm new to asterisk's svn and bug tracking tool, is it
sufficient then to apply the two patches (iax_dcallno_check-1.2.rev3.txt
and iax_dcallno_check.rev9.txt) listed in
http://bugs.digium.com/view.php?id=10078 to a 1.4.11ish release to
correct this vulnerability?  I really don't feel like buying into
any/all of the headaches that went into 1.4.11->1.4.20.  You know, "if
it ain't broke don't fix it", and my corollary, "if it is broke, only
fix what's broke, don't try to make it better".  :-)

Thanx,
b.



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Re: [asterisk-users] need examples of asterisk and mysql integration

[andres]

On Tue, 2008-04-22 at 13:13 -0700, Eric Fort wrote:

> I'm presently working on a project to build a scheduling system
> accessible by both web and phone.  on the web side one can query what
> items are available when by using the time or the item as a key then
> reserve for an available time slot.  reservations may also be modified
> by the user that made them or an admin.  Where may I find examples of
> doing similar things with asterisk?  


don't forget ruby's adhearsion


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Re: [asterisk-users] features.conf Problem with DTMF_sequence

[Tilghman Lesher]

On Tuesday 22 April 2008 05:22, Sergey Shumeyko wrote:
> I have following problem with my Asterisk installation (version 1.6.0. beta
> 7.1). I want to assign start record conversation to #7 and stop record
> conversation to #8, but it isn't working (on previous Asterisk 1.2.17 it
> was working fine).  When I assign those functions to 7/8 (without #)
> correspondingly it also works fine, but it works only from caller side. I
> would appreciate very much if somebody can take a look at my configuration
> below and give me comments what I am doing wrong.

http://bugs.digium.com/view.php?id=12299

-- 
Tilghman

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[asterisk-users] Asterisk 1.2.28, 1.4.19.1, and 1.6.0-beta8 Released

[The Asterisk Development Team]

The Asterisk development team has released versions 1.2.28, 1.4.19.1, and
1.6.0-beta8.

All of these releases contain a security patch for the vulnerability described
in the AST-2008-006 security advisory.  1.6.0-beta8 is also a regular update to
the 1.6.0 series with a number of bug fixes over the previous beta release.

Early last year, we made some modifications to the IAX2 channel driver to combat
potential usage of IAX2 in traffic amplification attacks.  Unfortunately, our
fix was not complete and we were not notified of this until the original
reporter of the issue decided to release information on how to exploit it to the
public.

This issue affects all users of IAX2 that have allowed non-authenticated calls.
 For more information on the vulnerability, see the published security advisory.

 * http://downloads.digium.com/pub/security/AST-2008-006.pdf

All releases are available for download from the following location:

 * http://downloads.digium.com/pub/telephony/asterisk/

Thank you for your continued support of Asterisk!


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[asterisk-users] AST-2008-006 - 3-way handshake in IAX2 incomplete

[Security Officer]

   Asterisk Project Security Advisory - AST-2008-006

   ++
   |  Product   | Asterisk  |
   |+---|
   |  Summary   | 3-way handshake in IAX2 incomplete|
   |+---|
   | Nature of Advisory | Remote amplification attack   |
   |+---|
   |   Susceptibility   | Remote unauthenticated sessions   |
   |+---|
   |  Severity  | Critical  |
   |+---|
   |   Exploits Known   | Yes   |
   |+---|
   |Reported On | April 18, 2008|
   |+---|
   |Reported By | Joel R. Voss aka. Javantea < jvoss AT altsci DOT  |
   || com > |
   |+---|
   | Posted On  | April 22, 2008|
   |+---|
   |  Last Updated On   | April 22, 2008|
   |+---|
   |  Advisory Contact  | Tilghman Lesher < tlesher AT digium DOT com > |
   |+---|
   |  CVE Name  | CVE-2008-1897 |
   ++

   ++
   | Description | Javantea originally reported an issue in IAX2, whereby   |
   | | an attacker could send a spoofed IAX2 NEW message, and   |
   | | Asterisk would start sending early audio to the target   |
   | | address, without ever receiving an initial response. |
   | | That original vulnerability was addressed in June 2007,  |
   | | by requiring a response to the initial NEW message   |
   | | before starting to send any audio.   |
   | |  |
   | | Javantea subsequently found that we were doing   |
   | | insufficent verification of the ACK response and that|
   | | the ACK response could be spoofed, just like the initial |
   | | NEW message. We have addressed this failure with two |
   | | changes. First, we have started to require that the ACK  |
   | | response contains the unique source call number that we  |
   | | send in our reply to the NEW message. Any ACK response   |
   | | that does not contain the source call number that we |
   | | have created will be silently thrown away. Second, we|
   | | have made the generation of our source call number a |
   | | little more difficult to predict, by randomly selecting  |
   | | a source call number, instead of allocating them |
   | | sequentially.|
   ++

   ++
   | Workaround | Disable remote unauthenticated IAX2 sessions, by  |
   || disallowing guest access. |
   ++

   ++
   | Resolution | Upgrade your Asterisk installation to revision 114561 or  |
   || later, or install one of the releases shown below.|
   ++

   ++
   | Commentary | We would like to thank Javantea for notifying us of this  |
   || problem; however, we note that he posted exploit code |
   || prior to that notification, which is considered   |
   || irresponsible behavior in the whitehat security industry. |
   || In the future, advance notice of any such release would   |
   |   

Re: [asterisk-users] multiple users collisions

[Cyril SCETBON]

Issue resolved.

I was using a mobile phone (with speaker on) and my home phone. When I 
was pressing a key on my mobile phone, my home phone was catching the 
sound from the speaker.

Regards.

Moshe Brevda wrote:
> Logs?
> 
> On Thu, Apr 17, 2008 at 11:47 PM, Cyril SCETBON <[EMAIL PROTECTED] 
> > wrote:
> 
> Hi,
> 
> My dialplan works fine with one user (asking for the sharp key to be
> pressed to continue, and others), but when 2 users are calling at the
> same time if one press key # the two users are jumping to the next step.
> 
> Anyidea ?
> 
> FYI, I'm using Asterisk 1.4.10. 
> 
> --
> Cyril SCETBON
> 
> 
> 
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> 
> 
> 
> -- 
> Moshe Brevda, CTO
> ipconnect, ltd.
> 26 Strauss St., Jerusalem, Israel
> W. 1.800.800.456 (+9722.569.5295)
> M. +97254.666.1367
> 
> 
> 
> 
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-- 
Cyril SCETBON



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[asterisk-users] Can't transfer call

[Cyril SCETBON]

Hi guys,

I receiving call through a gateway without any problem but I can't 
transfer the call. Asterisk is complaining about not being able to 
translate a path and getting 403 error from gateway.

Here is my sip configuration :

[412345679] 
context=accueil
host=192.168.19.10
username=412345679
type=peer
insecure=very

[sipout]
type=peer
host=192.168.19.10

in extension.conf I'm trying to transfer the call :

exten => _*,1,Dial(SIP/sipout/612345678)

but asterisk does not agree :-(

Audio is at 192.168.19.10 port 19322
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.19.1:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.19.10:5060;branch=z9hG4bK2fa34bcb;rport
From: "412345678" ;tag=as123c4a09
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 22 Apr 2008 20:35:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 3385 3385 IN IP4 192.168.19.10
s=session
c=IN IP4 192.168.19.10
t=0 0
m=audio 19322 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
 -- Called sipout/612345678
[Apr 22 22:35:23] WARNING[3431]: channel.c:3337 
ast_channel_make_compatible: No path to translate from 
SIP/sipout-081909d0(4) to SIP/412345679-081885e8(8)
www*CLI>
<--- SIP read from 192.168.19.1:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.19.10:5060;branch=z9hG4bK2fa34bcb;rport
From: "412345678" ;tag=as123c4a09
To: ;tag=46E30A54-C9A
Date: Tue, 22 Apr 2008 20:35:20 GMT
Call-ID: [EMAIL PROTECTED]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0


<->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.19.1:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.19.10:5060;branch=z9hG4bK2fa34bcb;rport
From: "412345678" ;tag=as123c4a09
To: ;tag=46E30A54-C9A
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[Apr 22 22:35:23] WARNING[3392]: chan_sip.c:11995 
handle_response_invite: Received response: "Forbidden" from '"412345678" 
;tag=as123c4a09'
 -- SIP/sipout-081909d0 is circuit-busy
   == Everyone is busy/congested at this time (1:0/1/0)
Really destroying SIP dialog 
'[EMAIL PROTECTED]' Method: INVITE
www*CLI>
<--- SIP read from 192.168.19.1:55654 --->
BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP  192.168.19.1:5060;branch=z9hG4bK1895912ED
From: ;tag=46E29598-2618
To: ;tag=as73c6a52d
Date: Tue, 22 Apr 2008 20:34:51 GMT
Call-ID: [EMAIL PROTECTED]
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 12
Timestamp: 1208896525
CSeq: 102 BYE
Reason: Q.850;cause=16
Content-Length: 0


<->
--- (12 headers 0 lines) ---
Sending to 192.168.19.1 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.19.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
192.168.19.1:5060;branch=z9hG4bK1895912ED;received=192.168.19.1
From: ;tag=46E29598-2618
To: ;tag=as73c6a52d
Call-ID: [EMAIL PROTECTED]
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: 
Content-Length: 0


<>
Really destroying SIP dialog 
'[EMAIL PROTECTED]' Method: BYE
Executing last minute cleanups
   == Destroying musiconhold processes


Regards
-- 
Cyril SCETBON



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Re: [asterisk-users] need examples of asterisk and mysql integration

[Michiel van Baak]

On 13:13, Tue 22 Apr 08, Eric Fort wrote:
> I'm presently working on a project to build a scheduling system
> accessible by both web and phone.  on the web side one can query what
> items are available when by using the time or the item as a key then
> reserve for an available time slot.  reservations may also be modified
> by the user that made them or an admin.  Where may I find examples of
> doing similar things with asterisk?  all I've been able to find thus
> far is examples of how to store call detail records and voicemail
> using a database.

Hi,

With AGI you can write logic in whatever language fits you best.
if you like perl, use the asterisk-perl stuff.
if you like php, you can use the native mysql stuff
if you like bash, you can use the mysql commandline client
etc etc
You can even use the MYSQL dialplan functions. Not sure where they are,
I think in -addons.

-- 

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

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


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Re: [asterisk-users] OT: Linksys devices send incorrect REGISTER

[CSB]

>I would suspect it's an Asterisk issue and not a Linksys issue. We use 
>a non-Asterisk registrar with 1000's of Linksys devices and don't have 
>that problem.
>
>If you are starting to get a lot of registration traffic it would be a 
>good time to look at a way at moving it off Asterisk. Asterisk is great 
>for the media and feature side of the PBX but there are better 
>solutions for signalling and registrations such as OpenSER.

Sounds good since we run OpenSER for other stuff already. But how does it
work? When two OpenSER-registered UACs want to call each other through
Asterisk how does that happen? When a call comes into an IVR how does
Asterisk know where to contact the relevant UAC if it's not registered with
it?

Thanks 

Cameron



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Re: [asterisk-users] Can I roll my own E911?

[Michael Graves]

On Tue, 22 Apr 2008 13:17:43 -0400, Alex Balashov wrote:

>> I my case I bought a single port SIP-GSM gateway that takes a SIM card
>> from my cellular provider. I then made the gateway a peer to my
>> Asterisk. Voila...a GSM trunk. Add dialplan logic to route 411 and 911
>> calls to that trunk and away you go.
>> 
>
>Great approach!
>
>My question would be - is this actually compliant with the FCC E911 
>regulations applicable to VoIP providers?

I could not even guess about such things. I provide the service to
myself for my home and home office.

For moral reasons I cannot do business with AT&T, who are the only
purveyor of POTS lines hereabouts. This was a significant part of my
motivation to find an alternative.

Michael

--
Michael Graves
mgravesmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
[EMAIL PROTECTED]



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Re: [asterisk-users] need examples of asterisk and mysql integration

[Mike Trest - On Travel]

Hi,

I suggest you look at writing a PERL  agi program to handle all of 
the MYSQL / DB
access and just pass variables between your CONTEXT/dialplan.   I have done
a lot of these things.  You can get PERL examples for DBI  and use one of
provided  agi scripts as a prototype.

..mike..

At 04:13 PM 4/22/2008, you wrote:
>I'm presently working on a project to build a scheduling system
>accessible by both web and phone.  on the web side one can query what
>items are available when by using the time or the item as a key then
>reserve for an available time slot.  reservations may also be modified
>by the user that made them or an admin.  Where may I find examples of
>doing similar things with asterisk?  all I've been able to find thus
>far is examples of how to store call detail records and voicemail
>using a database.
>
>Thanks in advance,
>
>Eric
>
>P.S.
>
>Has anyone already built an asterisk/web based scheduling system like this?
>
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Re: [asterisk-users] Can I roll my own E911?

[Mike Trest - On Travel]

>

Yes,  if you have signed waivers, you may operate without fear of 
FCC.  Just be sure to have
physical paper in a file somewhere for each client in the event of an audit.

And, this will also satisfy you legal advisors to avoid liability in 
lawsuit by a consumer towards
you for any crazy reason if they think that you provided inadequate 
E911 service and failed
to advise the consumer regards proper use and expectations..

>But if you do get those waivers, it is compliant in principle, on that
>basis, right?
>
>--
>Alex Balashov
>Evariste Systems
>Web: http://www.evaristesys.com/
>Tel: (+1) (678) 954-0670
>Direct : (+1) (678) 954-0671
>Mobile : (+1) (706) 338-8599
>
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Re: [asterisk-users] need examples of asterisk and mysql integration

[linuxian iandsd]

vicidial has something similar in a way that you can schedule a callback ...
maybe you can download it & get a closer look at the code, it connects to
asterisk thru asterisk-perl module.

On Tue, Apr 22, 2008 at 8:13 PM, Eric Fort <[EMAIL PROTECTED]> wrote:

> I'm presently working on a project to build a scheduling system
> accessible by both web and phone.  on the web side one can query what
> items are available when by using the time or the item as a key then
> reserve for an available time slot.  reservations may also be modified
> by the user that made them or an admin.  Where may I find examples of
> doing similar things with asterisk?  all I've been able to find thus
> far is examples of how to store call detail records and voicemail
> using a database.
>
> Thanks in advance,
>
> Eric
>
> P.S.
>
> Has anyone already built an asterisk/web based scheduling system like
> this?
>
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[asterisk-users] need examples of asterisk and mysql integration

[Eric Fort]

I'm presently working on a project to build a scheduling system
accessible by both web and phone.  on the web side one can query what
items are available when by using the time or the item as a key then
reserve for an available time slot.  reservations may also be modified
by the user that made them or an admin.  Where may I find examples of
doing similar things with asterisk?  all I've been able to find thus
far is examples of how to store call detail records and voicemail
using a database.

Thanks in advance,

Eric

P.S.

Has anyone already built an asterisk/web based scheduling system like this?

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[asterisk-users] OT - How to configure TAPI client-server ?

[Olivier]

Hi,

ActivaTSP provides a way for TAPI compliant applications to access an
Asterisk telephony server.

On one hand, ActivaTSP uses AMI to dialog with Asterisk.
On the other hand,  ActivaTSP in integrated to TAPI middleware so that TAPI
applications (Outlook, ACT!, ...) gain access to Asterisk.

My question is mostly about TAPI installation and configuration (that's why
it's a bit off-topic in an Asterisk User mailing list).

How do you configure an Outlook/XP client, so that it can use a TAPI Server
Provider on a W2003 server ?
More precisely, is the following correct ?

- $ tcmsetup /c tapisername (from client's DOS)
- enabling Microsoft Windows Remote Service Provider in Config panel/Modems
- selecting remote TSP as Dialing option when calling a contact within
Outlook

In my case, I can't see any remote TSP (only standard and default ones such
H323, IPCONF, Unimodem ...) in Outlook's Dialing option dialog box.

Cheers
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Re: [asterisk-users] Can I roll my own E911?

[Jon Pounder]

Quoting Alex Balashov <[EMAIL PROTECTED]>:

FCC / CRTC or not, I think the reality is voip and cellular has just  
made a mess of the 911 system, and trying to make 911 fit all  
situations is just not possible. If you can't speak and tell the  
operator your actual address, essentially you are SOL, you also have  
to take it on faith you get connected to the right 911 operations  
center - how many cities have "main st" for example and would assume  
you are local if that is all you could tell them.

Right now I wouldn't dream of a purely voip system for just that  
reason, always at least one real line for emergencies.






> Mike Trest - On Travel wrote:
>> At 01:17 PM 4/22/2008, you wrote:
>>> My question would be - is this actually compliant with the FCC E911
>>> regulations applicable to VoIP providers?
>>
>> IMHO and EXPERIENCE before FCC, this arrangement is NOT compliant
>>
>> Reason:  multiple subscribers using the same number
>>neither number nor name nor location can be
>>tied to a specific subscriber at the time of use.
>>
>> You will need waivers on file from each client that acknowledges
>> the E911 service is verbal contact only and will not have a
>> fixed location or subscriber name associated with the number that
>> is seen by the E911 service provider.
>
> But if you do get those waivers, it is compliant in principle, on that
> basis, right?
>
> --
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
>
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>



Jon Pounder

_/_/_/  _/_/  _/   _/_/_/  _/_/  _/_/_/_/
 _/_/_/  _/  _/ _/_/_/  _/  _/_/
_/_/  _/_/  _/ _/_/  _/_/  _/
_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.inline.net
www.ihtml.com
www.ihtmlmerchant.com
www.opayc.com



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Re: [asterisk-users] Can I roll my own E911?

[Alex Balashov]

Mike Trest - On Travel wrote:
> At 01:17 PM 4/22/2008, you wrote:
>> My question would be - is this actually compliant with the FCC E911
>> regulations applicable to VoIP providers?
> 
> IMHO and EXPERIENCE before FCC, this arrangement is NOT compliant
> 
> Reason:  multiple subscribers using the same number
>neither number nor name nor location can be
>tied to a specific subscriber at the time of use.
> 
> You will need waivers on file from each client that acknowledges
> the E911 service is verbal contact only and will not have a
> fixed location or subscriber name associated with the number that
> is seen by the E911 service provider.

But if you do get those waivers, it is compliant in principle, on that 
basis, right?

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Can I roll my own E911?

[SIP]

Mike Trest - On Travel wrote:
> At 01:17 PM 4/22/2008, you wrote:
>   
>> My question would be - is this actually compliant with the FCC E911
>> regulations applicable to VoIP providers?
>> 
>
> IMHO and EXPERIENCE before FCC, this arrangement is NOT compliant
>
> Reason:  multiple subscribers using the same number
>neither number nor name nor location can be
>tied to a specific subscriber at the time of use.
>
> You will need waivers on file from each client that acknowledges
> the E911 service is verbal contact only and will not have a
> fixed location or subscriber name associated with the number that
> is seen by the E911 service provider.
>
>
>
>   
Technically, no. This might not suffice. FCC 05-116 sec37 states that a 
call must be routed via ANI or pseudo-ANI via the dedicated Wireline 
E911 Network (not wireless). Now, if this were interconnected to a 
single wireline point, it MIGHT suffice, as the same section specifies 
that you can use ANI or pseudo-ANI and need only provide appropriate 
callback information for the PSAP.

Sec.38, however, states that you can accomplish this interconnection 
either indirectly through a 3rd party, directly through the Wireline 
E911 network, OR through "any other solution that allows a provider to 
offer E911 service as described above."  That 'any other solution' MIGHT 
allow you to interconnect through a single point wireless IF you can 
provide individual callback info to the PSAP.

N.

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Re: [asterisk-users] WARNING: Remote host can't match request NOTIFY to call on Audiocodes MP-124 FXS

[Atis Lezdins]

On Tue, Apr 22, 2008 at 3:15 PM, Grey Man <[EMAIL PROTECTED]> wrote:
> For blind transfers Asterisk will send the call back to the dial plan
>  and into the TRANSFER (I think, could be a different name) context if
>  it exists. Within that context you can access the channel that was
>  answered on the original call using ${DIALEDPEERNUMBER}.
>
>  Note that this mechanism cannot be use for attended transfers as they
>  are not sent back to the dial plan for processing.

I apologize, but I don't have any problems with transfers. The
warnings I get in log appears there even without any calls going on.

Maybe You replied to wrong topic?

Regards,
Atos


-- 
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] Can I roll my own E911?

[Mike Trest - On Travel]

At 01:17 PM 4/22/2008, you wrote:
>My question would be - is this actually compliant with the FCC E911
>regulations applicable to VoIP providers?

IMHO and EXPERIENCE before FCC, this arrangement is NOT compliant

Reason:  multiple subscribers using the same number
   neither number nor name nor location can be
   tied to a specific subscriber at the time of use.

You will need waivers on file from each client that acknowledges
the E911 service is verbal contact only and will not have a
fixed location or subscriber name associated with the number that
is seen by the E911 service provider.





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Re: [asterisk-users] Can I roll my own E911?

[Alex Balashov]

Michael Graves wrote:
> On Tue, 22 Apr 2008 11:46:29 -0500, Doug wrote:
> 
>> At 09:09 4/22/2008, Michael Graves wrote:
>>> On Mon, 21 Apr 2008 23:17:40 -0400, Adam Moffett wrote:
>>>
 Assuming I only operate in one municipality (I do), and assuming I made
 some sort of connection to the emergency services center in this area,
 via SIP or a T1 or whatever, does asterisk have a way for me to send the
 E911 address data?
>>> Others have certainly offered good responses and advice on the matter.
>>> But I have a slighlty different approach. I use a GSM gateway to
>>> provide what are effectively "cellular trunks" that provide 911 (and
>>> 411 for that matter) through a cellular carrier.
>>>
>>> It seems an inexpensive way to ensure 911 service in what is otherwise
>>> a 100% VOIP situation.
>>>
>>> Michael
>> Hey Michael,
>>
>> Sounds ingenuous.  Please elaborate.
>>
>> Do you fake a cellphone, or do you forward through
>> a cellphone.
>>
>> How do you register each phone number? 
> 
> There can be several approaches to the mechanics of this. Some more
> elegant than others IMHO.
> 
> You could use chan_bluetooth to connect an old cell phone. You can get
> a "Dock-n-talk" to bridge from the celullar to fixed line world.
> 
> I my case I bought a single port SIP-GSM gateway that takes a SIM card
> from my cellular provider. I then made the gateway a peer to my
> Asterisk. Voila...a GSM trunk. Add dialplan logic to route 411 and 911
> calls to that trunk and away you go.
> 
> Such gateways are available from 1 to 32 ports. Some are GSM - FXO
> while others are GSM - SIP/H.323. Some are even GSM - Skype. Just
> Google "GSM gateway" and follow the bread crumbs.
> 
> I'm half way through an article on the project that will eventually fin
> its way to www.smallnetbuilder.com. I find that I can help offset the
> cost of the project by writing it up properly for a publisher.

Great approach!

My question would be - is this actually compliant with the FCC E911 
regulations applicable to VoIP providers?

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Can I roll my own E911?

[Michael Graves]

On Tue, 22 Apr 2008 11:46:29 -0500, Doug wrote:

>At 09:09 4/22/2008, Michael Graves wrote:
> >On Mon, 21 Apr 2008 23:17:40 -0400, Adam Moffett wrote:
> >
> >>Assuming I only operate in one municipality (I do), and assuming I made
> >>some sort of connection to the emergency services center in this area,
> >>via SIP or a T1 or whatever, does asterisk have a way for me to send the
> >>E911 address data?
> >
> >Others have certainly offered good responses and advice on the matter.
> >But I have a slighlty different approach. I use a GSM gateway to
> >provide what are effectively "cellular trunks" that provide 911 (and
> >411 for that matter) through a cellular carrier.
> >
> >It seems an inexpensive way to ensure 911 service in what is otherwise
> >a 100% VOIP situation.
> >
> >Michael
>
>Hey Michael,
>
>Sounds ingenuous.  Please elaborate.
>
>Do you fake a cellphone, or do you forward through
>a cellphone.
>
>How do you register each phone number? 

There can be several approaches to the mechanics of this. Some more
elegant than others IMHO.

You could use chan_bluetooth to connect an old cell phone. You can get
a "Dock-n-talk" to bridge from the celullar to fixed line world.

I my case I bought a single port SIP-GSM gateway that takes a SIM card
from my cellular provider. I then made the gateway a peer to my
Asterisk. Voila...a GSM trunk. Add dialplan logic to route 411 and 911
calls to that trunk and away you go.

Such gateways are available from 1 to 32 ports. Some are GSM - FXO
while others are GSM - SIP/H.323. Some are even GSM - Skype. Just
Google "GSM gateway" and follow the bread crumbs.

I'm half way through an article on the project that will eventually fin
its way to www.smallnetbuilder.com. I find that I can help offset the
cost of the project by writing it up properly for a publisher.

Michael
--
Michael Graves
mgravesmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
[EMAIL PROTECTED]



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Re: [asterisk-users] Can I roll my own E911?

[Doug]

At 09:09 4/22/2008, Michael Graves wrote:
 >On Mon, 21 Apr 2008 23:17:40 -0400, Adam Moffett wrote:
 >
 >>Assuming I only operate in one municipality (I do), and assuming I made
 >>some sort of connection to the emergency services center in this area,
 >>via SIP or a T1 or whatever, does asterisk have a way for me to send the
 >>E911 address data?
 >
 >Others have certainly offered good responses and advice on the matter.
 >But I have a slighlty different approach. I use a GSM gateway to
 >provide what are effectively "cellular trunks" that provide 911 (and
 >411 for that matter) through a cellular carrier.
 >
 >It seems an inexpensive way to ensure 911 service in what is otherwise
 >a 100% VOIP situation.
 >
 >Michael

Hey Michael,

Sounds ingenuous.  Please elaborate.

Do you fake a cellphone, or do you forward through
a cellphone.

How do you register each phone number? 


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Re: [asterisk-users] Switch recommendation?

[Matt Watson]

I'm using Dell 3548P switches currently which I have powering 25 phone (mostly 
Aastra 9133i's, a couple 480i's a 57i + 560M)

So... basically I have a phone on about half my ports... my power utilization 
on the switch is:

console# show power inline

Unit  Power  Nominal Power   Consumed Power   Usage Threshold   Traps
 --- - -- --- -
 1 On  370 Watts 78 Watts (21%) 95 Disable


The unit will supply up to 370W of power, or 470W if you buy the additional 
power supply for it.


--
Matt

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan C. 
Bailey
Sent: Monday, April 21, 2008 9:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Switch recommendation?

We've been using D-Link DES-3028P and DES-3052P switches. They can supply full 
power to EACH port unlike the Linksys switches we've tried. They're also rock 
solid from our experience.


-Jon

- Original Message -
From: "Hilary Miller" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Monday, April 21, 2008 8:21:12 PM GMT -06:00 US/Canada Central
Subject: Re: [asterisk-users] Switch recommendation?

On Mon, Apr 21, 2008 at 5:54 PM, Sean Dennis <[EMAIL PROTECTED]> wrote:
>  The Cisco 3524 switch doesn't support 802.3af which is what your Linksys
>  phones are going to want.

Thank you for sharing Sean! When I saw them I felt a disturbance in
the force, and now I know why!

--
Just Hil

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Re: [asterisk-users] Conditional include=> ?

[Roderick A. Anderson]

Steve Davies wrote:
> 2008/4/22 Tzafrir Cohen <[EMAIL PROTECTED]>:
>> On Tue, Apr 22, 2008 at 09:55:37AM +0100, Steve Davies wrote:
>>  > Hi,
>>  >
>>  > Does anyone have a clever method of doing a conditional "include =>"
>>  > line in the dialplan?
>>  >
>>  > I want to include a bunch of standard contexts, but in the middle of
>>  > the bunch have one or more conditionally included, a bit like:
>>  >
>>  > [default]
>>  > include => start-here
>>  > include => then-here
>>  > if $[{COMPANY} = A]
>>  > include => company-A
>>  > endif
>>  > if $[{COMPANY} = B]
>>  > include => company-B
>>  > endif
>>  > if $[{COMPANY} = C]
>>  > include => company-C
>>  > endif
>>  > include => end-here
>>  >
>>  >
>>  > The list of conditional variables, and of static includes is
>>  > potentially large, so I do not want to create several separate sets of
>>  > contexts with lots of repetition as it is likely to get out-of-control
>>  > very quickly.
>>
>>  A different approach:
>>
>>  [company-base](!)
>>  ; common settings
>>
>>  [company-A](company-base)
>>  ; specific for company A
>>
>>  [company-B](company-base)
>>  ; specific for company B
>>
>>  [company-C](company-base)
>>  ; specific for company C
>>
>>
>>  Keep in mind you can also use:
>>
>>  [sub-template](!,base-template)
>>
>>  And:
>>
>>  [context](template1,template2)
>>
>>  But one limitation is that you can only add: no way to remove line added
>>  by a template your context uses.
>>
> 
> That sounds interesting, though I am not aware of that format - Guess
> I'll go look it up in the WiKi :)

It was a nice surprise I got while reading the downloaded version of 
"Asterisk: The Future of Telephony."  Not sure if it is in the Wiki as 
the PDF version works fine until my dead-tree version arrives.  I didn't 
even look.


Rod
-- 
> 
> Thanks,
> Steve
> 
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Re: [asterisk-users] DUNDi and SIP

[Bruce Reeves]

Try this,

[priv]
dbsecret=dundi/secret
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal
type=friend

priv => dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial



On Tue, Apr 22, 2008 at 8:23 AM, Jeremy Mann <[EMAIL PROTECTED]> wrote:
> No.
>
>
>  -Original Message-
>  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
>
>
> Sent: Tuesday, April 22, 2008 6:00 AM
>  To: Asterisk Users Mailing List - Non-Commercial Discussion
>  Subject: Re: [asterisk-users] DUNDi and SIP
>
>  Jeremy,
>
>  Did you get this working?
>
>
>
>  On Thu, Apr 17, 2008 at 7:38 AM, Jeremy Mann <[EMAIL PROTECTED]> wrote:
>  > I have it working via IAX, when I try changing everything to SIP I can't 
> specify a username and an extension, so it becomes useless.
>  >
>  >
>  >
>  >  -Original Message-
>  >  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce 
> Reeves
>  >  Sent: Thursday, April 17, 2008 6:51 AM
>  >  To: Asterisk Users Mailing List - Non-Commercial Discussion
>  >  Subject: Re: [asterisk-users] DUNDi and SIP
>  >
>  >  Jeremy,
>  >
>  >  Here is a working sample to compare to. This is an IAX2 setup, but the
>  >  only difference is in the mapping change IAX2 to SIP. Notice the 4th
>  >  setting in the mapping? It defines to use the IAX2 peer "priv" with
>  >  the secret generated of the key defined in the peers section of
>  >  dundi.conf. When you look at the peer in iax.conf on the remote box,
>  >  there is no host entry and it uses dbsecret=dundi/secret, the
>  >
>  >  dundi.conf
>  >  priv => dundi-internal,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
>  >
>  >  [00:19:66:1C:78:D5] ; Dev Box
>  >  model = symmetric
>  >  host = 192.168.99.252
>  >  inkey = eus
>  >  outkey = eus
>  >  include = priv
>  >  permit = priv
>  >  qualify = yes
>  >
>  >
>  >  From iax.conf
>  >  [priv]
>  >  type=friend
>  >  dbsecret=dundi/secret
>  >  context=longdistance
>  >
>  >  Hope this helps, in your case Dundi will save you a world of work on
>  >  configuring that many systems, in fact if you structure Dundi like
>  >  spokes around a small number of master servers, the config gets real
>  >  easy.Let me know how it goes.
>  >
>  >  On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann <[EMAIL PROTECTED]> wrote:
>  >  >
>  >  >
>  >  >
>  >  >
>  >  > I'm a little confused with DUNDi and SIP as the backend channel type:
>  >  >
>  >  >
>  >  >
>  >  > Dundi.conf:
>  >  >
>  >  > [mappings]
>  >  >
>  >  > priv => dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
>  >  >
>  >  >
>  >  >
>  >  > Using the above, the dial string passed to the person on the other box 
> is
>  >  > SIP/[EMAIL PROTECTED]
>  >  >
>  >  >
>  >  >
>  >  > How can you use authentication, along with SIP, along with specifying
>  >  > extension?
>  >  >
>  >  >
>  >  >
>  >  > My sip.conf has a friend defined:
>  >  >
>  >  >
>  >  >
>  >  > [priv]
>  >  >
>  >  > host=dynamic
>  >  >
>  >  > secret=priv
>  >  >
>  >  > disallow=all
>  >  >
>  >  > allow=ulaw
>  >  >
>  >  > canreinvite=no
>  >  >
>  >  > nat=no
>  >  >
>  >  > context=from-internal\
>  >  >
>  >  > type=friend
>  >  >
>  >  >
>  >  >
>  >  > I need to specify the sip channel to use the priv peer, priv secret, and
>  >  > pass the extension.  I've tried defining my mapping as:
>  >  >
>  >  >
>  >  >
>  >  > Priv => dundi-priv-local,0,SIP,priv:[EMAIL 
> PROTECTED]/${NUMBER},nopartial
>  >  >
>  >  >
>  >  >
>  >  > But obviously the console on the far end complains that peer
>  >  > a.b.c.d/${NUMBER} cannot be found.
>  >  >
>  >  >
>  >  >
>  >  > Thanks for any insight into this.  I'd prefer not having to define a sip
>  >  > peer per box(I have 25 connected in my dundi cloud), nor would I like to
>  >  > enable anonymous SIP calls, as I have the ports open to the world for
>  >  > inbound sip from bandwidth.com
>  >  >
>  >  >
>  >  >
>  >  >
>  >  >  
>  >  >  This e-mail, facsimile, or letter and any files or attachments 
> transmitted
>  >  > with it contains information that is confidential and privileged. This
>  >  > information is intended only for the use of the individual(s) and
>  >  > entity(ies) to whom it is addressed. If you are the intended recipient,
>  >  > further disclosures are prohibited without proper authorization. If you 
> are
>  >  > not the intended recipient, any disclosure, copying, printing, or use of
>  >  > this information is strictly prohibited and possibly a violation of 
> federal
>  >  > or state law and regulations. If you have received this information in
>  >  > error, please notify Texas Health Management Group immediately at
>  >  > 1-817-310-4999. Texas Health Management Group, its subsidiaries, and
>  >  > affiliates hereby claim all applicable privileges related to this
>  >  > information.
>  >  >
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Re: [asterisk-users] Conditional include=> ?

[Steve Davies]

2008/4/22 Tzafrir Cohen <[EMAIL PROTECTED]>:
>
> On Tue, Apr 22, 2008 at 09:55:37AM +0100, Steve Davies wrote:
>  > Hi,
>  >
>  > Does anyone have a clever method of doing a conditional "include =>"
>  > line in the dialplan?
>  >
>  > I want to include a bunch of standard contexts, but in the middle of
>  > the bunch have one or more conditionally included, a bit like:
>  >
>  > [default]
>  > include => start-here
>  > include => then-here
>  > if $[{COMPANY} = A]
>  > include => company-A
>  > endif
>  > if $[{COMPANY} = B]
>  > include => company-B
>  > endif
>  > if $[{COMPANY} = C]
>  > include => company-C
>  > endif
>  > include => end-here
>  >
>  >
>  > The list of conditional variables, and of static includes is
>  > potentially large, so I do not want to create several separate sets of
>  > contexts with lots of repetition as it is likely to get out-of-control
>  > very quickly.
>
>  A different approach:
>
>  [company-base](!)
>  ; common settings
>
>  [company-A](company-base)
>  ; specific for company A
>
>  [company-B](company-base)
>  ; specific for company B
>
>  [company-C](company-base)
>  ; specific for company C
>
>
>  Keep in mind you can also use:
>
>  [sub-template](!,base-template)
>
>  And:
>
>  [context](template1,template2)
>
>  But one limitation is that you can only add: no way to remove line added
>  by a template your context uses.
>

That sounds interesting, though I am not aware of that format - Guess
I'll go look it up in the WiKi :)

Thanks,
Steve

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Re: [asterisk-users] Asterisk sends 486 Busy Here instead of 600 Busy Everywhere

[Jared Smith]

On Tue, 2008-04-22 at 18:11 +0530, Aadilkhan Maniyar wrote:
> Ideally Asterisk should be relaying the 600 response. What I fail to
> get is, why does Asterisk need to send 486 instead of 600.
> 

Asterisk is a back-to-back user agent (not a proxy!), and as such, it
doesn't always relay the exact same SIP responses back to the first leg
of the call. 
 
> 
> Is there any configuration that needs to be done in order to achieve
> this or is this a default behavior of Asterisk.

Right now, I don't know of any configuration that can be done with
Asterisk that can achieve that behavior.  Somebody smarter than I am
will have to answer the question of how hard it would be to change the
Asterisk source code to do that (and whether or not that's the right
thing for Asterisk to do).  

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Cisco 7961 + 7914, speeddials, BLF & Asterisk 1.4?

[Michiel van Baak]

On 16:12, Tue 22 Apr 08, Patrick wrote:
> Hi,
> 
> Does anyone have any experience with a Cisco 7961 + 7914 Operator
> Console setup and speeddials/BLF on Asterisk 1.4? Would appreciate
> feedback if this works reliably. I have a 7961 on skinny registering on
> an 1.4.19 box with chan_sccp and speeddials work fine so that part seems
> ok. I have no experience with the BLF part.
> 
> >From googling around it seems that BLF on the 7914 only works with
> skinny. Is that the case? If so, anyone know if chan_skinny as part of
> Asterisk 1.4 will do or will I need to use chan_sccp? Any other
> requirements or patches needed?
> 
> Your feedback is most appreciated.

chan_skinny in 1.4 is very limited. It does not support hints/speeddials
on the phones.
The 1.6 version has this, and should work fine in your setup.

I'm using chan_skinny at home with both an 7905 and 7960 and
hints/speeddials just work.

I dont know how good chan_sccp is these days. I used it in the 1.0 time
but went with chan_skinny a long time ago (I only use it at home, and
there I run trunk)

-- 

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

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


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Re: [asterisk-users] Conditional include=> ?

[Steve Davies]

2008/4/22 Philipp Kempgen <[EMAIL PROTECTED]>:
> Steve Davies schrieb:
>
>
>
>  > Does anyone have a clever method of doing a conditional "include =>"
>  > line in the dialplan?
>  >
>  > I want to include a bunch of standard contexts, but in the middle of
>  > the bunch have one or more conditionally included, a bit like:
>  >
>  > [default]
>  > include => start-here
>  > include => then-here
>  > if $[{COMPANY} = A]
>  > include => company-A
>  > endif
>  > if $[{COMPANY} = B]
>  > include => company-B
>  > endif
>  > if $[{COMPANY} = C]
>  > include => company-C
>  > endif
>  > include => end-here
>  >
>  >
>  > The list of conditional variables, and of static includes is
>  > potentially large, so I do not want to create several separate sets of
>  > contexts with lots of repetition as it is likely to get out-of-control
>  > very quickly.
>
>  I suppose #ifdefs and a preprocessor like cpp is not what you're
>  looking for? You didn't make clear at what stage you want these
>  includes to be replaced / evaluated. I guess you want Asterisk to
>  do it.
>
>  Regards,
>   Philipp Kempgen

I would be looking for a runtime evaluation based on a variable. I'll
take a look at Tzafrir's suggestion and try to undesrstand it :)

Steve

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Re: [asterisk-users] lots of warnings from translate.c

[Francesco Castellano]

I tried calls with a couple of other IP phones, but not extensively,
and these warnings did not happen. Anyway, also with the Siemens IP
phone, the warnings happen only sometimes.

At the moment, I'm not succeeded in correlating the warnings to a
specific user-agent.

Thanks,
Francesco

On Tue, Apr 22, 2008 at 12:50 PM, David Boyd <[EMAIL PROTECTED]> wrote:
>
> On Tue, 2008-04-22 at 12:28 +0200, Francesco Castellano wrote:
>  > We have a couple of servers with asterisk 1.4.19 and zaptel 1.4.10,
>  > acting as gateways from SIP to ISDN PRI interfaces. Each has one
>  > Digium TE420 (with hardware echo cancellation) and one TC400B for
>  > transcoding, in order to handle 60/90 contemporary calls in peak
>  > hours.
>  >
>  > In my logs there are hundreds of thousand warnigs per day like these:
>  >
>  > transcode.c: no samples for lintoulaw
>  > transcode.c: zapg729toalaw did not update samples ###
>  >
>  > Could you suggest me what are the possible causes for that? Are they
>  > signs of bad audio quality? Any ideas for resolving these issues?
>  >
>  > In addition I can say that we are using a quite large jitter buffer in
>  > zapata.conf:
>  >
>  > jitterbuffers=16 (=> 0.32s)
>  >
>  > Moreover, it uses the fixed implementation, because when I tried the
>  > adaptive one I experienced one-way audio.
>  > Finally I have to note that, using a Siemens IP phone (G.729 no
>  > AnnexB) in conditions of no load on servers, I could replicate
>  > non-deterministically (sigh!) each of these problems, with a very
>  > noisy audio, and a annoying period of silence during the first seconds
>  > of call.
>  >
>  > Regards,
>  > Francesco
>  >
>  > PS. Previous versions of asterisk and zaptel presented an identical 
> situation.
>  >
>  Have you tried additional types of phones and if so can you produce the
>  same non-deterministic problems?
>
>
>  Dave
>
>
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[asterisk-users] Cisco 7961 + 7914, speeddials, BLF & Asterisk 1.4?

[Patrick]

Hi,

Does anyone have any experience with a Cisco 7961 + 7914 Operator
Console setup and speeddials/BLF on Asterisk 1.4? Would appreciate
feedback if this works reliably. I have a 7961 on skinny registering on
an 1.4.19 box with chan_sccp and speeddials work fine so that part seems
ok. I have no experience with the BLF part.

>From googling around it seems that BLF on the 7914 only works with
skinny. Is that the case? If so, anyone know if chan_skinny as part of
Asterisk 1.4 will do or will I need to use chan_sccp? Any other
requirements or patches needed?

Your feedback is most appreciated.

Regards,
Patrick


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Re: [asterisk-users] Can I roll my own E911?

[Michael Graves]

On Mon, 21 Apr 2008 23:17:40 -0400, Adam Moffett wrote:

>Assuming I only operate in one municipality (I do), and assuming I made 
>some sort of connection to the emergency services center in this area, 
>via SIP or a T1 or whatever, does asterisk have a way for me to send the 
>E911 address data?

Others have certainly offered good responses and advice on the matter.
But I have a slighlty different approach. I use a GSM gateway to
provide what are effectively "cellular trunks" that provide 911 (and
411 for that matter) through a cellular carrier.

It seems an inexpensive way to ensure 911 service in what is otherwise
a 100% VOIP situation.

Michael
--
Michael Graves
mgravesmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
[EMAIL PROTECTED]



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[asterisk-users] Bip viol. in E1 cards

[Everton Goularth]

Hi all..

I`m with a problem in some E1 cards. The value of Bipolar Viol is too 
big, and some times the link seems well, but isn`t work correctly, like 
as if the link is down.


Normally, I use to stop asterisk, remove the modules from my card, use 
modprove to set up they again, and start asterisk, and it come to work, 
but in some times I have to reboot the server, and then the value of 
Bipolar Viol. seems short than before, so everything works well again.


My values:

asterisk*CLI> zap show status
Description  Alarms  IRQbpviol CRC4   
Fra Codi Options  LBO
Tormenta 3 (PCI) Quad E1 Card 0 Span 1   OK  0  5904   0  
CCS HDB3 YEL  0 db (CSU)/0-133 feet (DSX-1)
Tormenta 3 (PCI) Quad E1 Card 0 Span 2   OK  0  9449   0  
CCS HDB3 YEL  0 db (CSU)/0-133 feet (DSX-1)
Tormenta 3 (PCI) Quad E1 Card 0 Span 3   BLU/RED 0  0  0  
CAS Unk  YEL  0 db (CSU)/0-133 feet (DSX-1)
Tormenta 3 (PCI) Quad E1 Card 0 Span 4   BLU/RED 0  0  0  
CAS Unk  YEL  0 db (CSU)/0-133 feet (DSX-1)


How could I solve this problem? What can make this value of Bipolar Vil. 
so big?


thank`s for the opportunity.
Everton
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Re: [asterisk-users] DUNDi and SIP

[Jeremy Mann]

No.

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Tuesday, April 22, 2008 6:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP

Jeremy,

Did you get this working?



On Thu, Apr 17, 2008 at 7:38 AM, Jeremy Mann <[EMAIL PROTECTED]> wrote:
> I have it working via IAX, when I try changing everything to SIP I can't 
> specify a username and an extension, so it becomes useless.
>
>
>
>  -Original Message-
>  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
>  Sent: Thursday, April 17, 2008 6:51 AM
>  To: Asterisk Users Mailing List - Non-Commercial Discussion
>  Subject: Re: [asterisk-users] DUNDi and SIP
>
>  Jeremy,
>
>  Here is a working sample to compare to. This is an IAX2 setup, but the
>  only difference is in the mapping change IAX2 to SIP. Notice the 4th
>  setting in the mapping? It defines to use the IAX2 peer "priv" with
>  the secret generated of the key defined in the peers section of
>  dundi.conf. When you look at the peer in iax.conf on the remote box,
>  there is no host entry and it uses dbsecret=dundi/secret, the
>
>  dundi.conf
>  priv => dundi-internal,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
>
>  [00:19:66:1C:78:D5] ; Dev Box
>  model = symmetric
>  host = 192.168.99.252
>  inkey = eus
>  outkey = eus
>  include = priv
>  permit = priv
>  qualify = yes
>
>
>  From iax.conf
>  [priv]
>  type=friend
>  dbsecret=dundi/secret
>  context=longdistance
>
>  Hope this helps, in your case Dundi will save you a world of work on
>  configuring that many systems, in fact if you structure Dundi like
>  spokes around a small number of master servers, the config gets real
>  easy.Let me know how it goes.
>
>  On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann <[EMAIL PROTECTED]> wrote:
>  >
>  >
>  >
>  >
>  > I'm a little confused with DUNDi and SIP as the backend channel type:
>  >
>  >
>  >
>  > Dundi.conf:
>  >
>  > [mappings]
>  >
>  > priv => dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
>  >
>  >
>  >
>  > Using the above, the dial string passed to the person on the other box is
>  > SIP/[EMAIL PROTECTED]
>  >
>  >
>  >
>  > How can you use authentication, along with SIP, along with specifying
>  > extension?
>  >
>  >
>  >
>  > My sip.conf has a friend defined:
>  >
>  >
>  >
>  > [priv]
>  >
>  > host=dynamic
>  >
>  > secret=priv
>  >
>  > disallow=all
>  >
>  > allow=ulaw
>  >
>  > canreinvite=no
>  >
>  > nat=no
>  >
>  > context=from-internal\
>  >
>  > type=friend
>  >
>  >
>  >
>  > I need to specify the sip channel to use the priv peer, priv secret, and
>  > pass the extension.  I've tried defining my mapping as:
>  >
>  >
>  >
>  > Priv => dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
>  >
>  >
>  >
>  > But obviously the console on the far end complains that peer
>  > a.b.c.d/${NUMBER} cannot be found.
>  >
>  >
>  >
>  > Thanks for any insight into this.  I'd prefer not having to define a sip
>  > peer per box(I have 25 connected in my dundi cloud), nor would I like to
>  > enable anonymous SIP calls, as I have the ports open to the world for
>  > inbound sip from bandwidth.com
>  >
>  >
>  >
>  >
>  >  
>  >  This e-mail, facsimile, or letter and any files or attachments transmitted
>  > with it contains information that is confidential and privileged. This
>  > information is intended only for the use of the individual(s) and
>  > entity(ies) to whom it is addressed. If you are the intended recipient,
>  > further disclosures are prohibited without proper authorization. If you are
>  > not the intended recipient, any disclosure, copying, printing, or use of
>  > this information is strictly prohibited and possibly a violation of federal
>  > or state law and regulations. If you have received this information in
>  > error, please notify Texas Health Management Group immediately at
>  > 1-817-310-4999. Texas Health Management Group, its subsidiaries, and
>  > affiliates hereby claim all applicable privileges related to this
>  > information.
>  >
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>  >
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>  >
>
>
>
>  --
>  *
>  Bruce Reeves, dCAp
>  EUS Networks
>  Office: 212-624-5943
>  Web: www.euscorp.com
>  
>
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>  This e-mail, facsimile, or letter and any files or attachments transmitted 
> with it contains information that is confidenti

[asterisk-users] Cisco 79X1 speaker issue

[Christophorus Laube]

Hi list,

I have a couple of Cisco 79X1 running well behind an asterisk. My
question is more dedicated to the ones of you knowing some tricks with
these phones. Does anyone of you know if there is a possibility to use
the speaker and the microphone of the handset? For now, when I activate
the speaker the microphone of the handset is forced off and cannot be
used anymore.
Thanks in advance.
Regards,

Christophorus Laube


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[asterisk-users] Asterisk sends 486 Busy Here instead of 600 Busy Everywhere

[Aadilkhan Maniyar]

Hi,
 
We have a scenario wherein the endpoint needs to send a 600 Busy
Everywhere after receiving an INVITE. I am using SIPp as this end point.
SIPp is configured as UE2.
Now when UE1 calls UE2 (SIPp) receives the INVITE and responds with a
600 Busy Everywhere.  
But when Asterisk receives this 600 response it sends out a 486 Busy
Here to UE1.
 
Ideally Asterisk should be relaying the 600 response. What I fail to get
is, why does Asterisk need to send 486 instead of 600.
 
Is there any configuration that needs to be done in order to achieve
this or is this a default behavior of Asterisk.
 
I am using Asterisk 1.4.17.
 
Thanks in Advance.
 
Regards,
Aadil
 
 
 
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Re: [asterisk-users] WARNING: Remote host can't match request NOTIFY to call on Audiocodes MP-124 FXS

[Grey Man]

For blind transfers Asterisk will send the call back to the dial plan
and into the TRANSFER (I think, could be a different name) context if
it exists. Within that context you can access the channel that was
answered on the original call using ${DIALEDPEERNUMBER}.

Note that this mechanism cannot be use for attended transfers as they
are not sent back to the dial plan for processing.

Regards,

Greyman.

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Re: [asterisk-users] OT: Linksys devices send incorrect REGISTER

[Grey Man]

I would suspect it's an Asterisk issue and not a Linksys issue. We use
a non-Asterisk registrar with 1000's of Linksys devices and don't have
that problem.

If you are starting to get a lot of registration traffic it would be a
good time to look at a way at moving it off Asterisk. Asterisk is
great for the media and feature side of the PBX but there are better
solutions for signalling and registrations such as OpenSER.

Regards,

Greyman.

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[asterisk-users] WARNING: Remote host can't match request NOTIFY to call on Audiocodes MP-124 FXS

[Atis Lezdins]

Hi,

I experience my log flooded with warning messages like this:

[Apr 14 01:30:24] WARNING[19514] chan_sip.c: Remote host can't match
request NOTIFY to call
'[EMAIL PROTECTED]'. Giving up

I traced this down to point when we added to sip.conf status notifications:

allowsubscribe=yes
rtcachefriends=yes

So, those notifications allow for queue to display (In Use) etc, and
creates no warnings for other devices except Audiocodes gateway.

I wonder is there any way how to disable this message in Asterisk, or
make Audiocodes act correctly?

Below is the sip debug for this (xx.xx.xx.xx is Audiocodes,
yy.yy.yy.yy is Asterisk).

Regards,
Atis

-


[Apr 14 01:30:24] VERBOSE[19514] logger.c: Scheduling destruction of
SIP dialog '[EMAIL PROTECTED]' in 32000 ms
(Method: NOTIFY)
[Apr 14 01:30:24] VERBOSE[19514] logger.c: Reliably Transmitting (NAT)
to xx.xx.xx.xx:5060:
NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK788fbefd;rport
From: "Unknown" ;tag=as436bf308
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 92

Messages-Waiting: no
Message-Account: sip:[EMAIL PROTECTED]
Voice-Message: 0/0 (0/0)

---
[Apr 14 01:30:24] VERBOSE[19514] logger.c:
<--- SIP read from xx.xx.xx.xx:5060 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK788fbefd;rport
From: "Unknown" ;tag=as436bf308
To: ;tag=1c73477527
Call-ID: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
Contact: 
Supported: em,timer,replaces,path
Allow: 
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Content-Length: 0


<->
[Apr 14 01:30:24] VERBOSE[19514] logger.c: --- (10 headers 0 lines) ---
[Apr 14 01:30:24] WARNING[19514] chan_sip.c: Remote host can't match
request NOTIFY to call '[EMAIL PROTECTED]'.
Giving up.


-- 
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] re-invite (bypass asterisk) post call establishment

[Benjamin Jacob]

Hi again,
I tried this again, but the reInvite happens immediately after the 200 OK/ACK. 
And then the D() specified DTMF is sent.

Attached is the SIP trace for the calls.
I call (from Asterisk) - 0119198807x 
After connect, I dial - 31927x.
This number 31927x is the conference bridge and I need to send DTMF (the 
bridge PIN) to it after connection. But alas, the reinvite happens before the 
D() is executed.
The SIP gateway is MySIPGateway at 204.aaa.bbb.ccc. 

cheers
- Ben.




Steve Davies <[EMAIL PROTECTED]> wrote: 2008/4/22 Benjamin Jacob :
[snip]
>
> So, my question : once the SDPs are exchanged, what will happen to the DTMFs
> sent by Asterisk using sendDTMF or the D option in dial.
>
[snip]

As far as I can tell, the D() option will be executed before the
re-invite takes place, so Asterisk will still be in-line. I believe
that the dial is not considered "complete/connected" until the D() is
finished.

Cheers,
Steve

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-
Be a better friend, newshound, and know-it-all with Yahoo! Mobile.  Try it now.

reInvite
Description: 1957794313-reInvite
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Re: [asterisk-users] Disable transfer on all calls

[Dinesh Nair]

On Tue, 22 Apr 2008 11:54:41 +0100, Grey Man wrote:
> The best option is to put a SIP Proxy in front of your Asterisk sever
> and block REFER requests.

or just comment out the block in chan_sip.c which handles the refers. 

-- 
Regards,   /\_/\   "All dogs go to heaven."
[EMAIL PROTECTED](0 0)   http://www.openmalaysiablog.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo "The opinions here in no way reflect the opinions of my $a $b."  |
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+=+

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[asterisk-users] Check the answered channel in simultaneous sip call

[Maurizio]

Hello
I usually make 
simultaneous calls using DIAL 
with "SIP/121&SIP/122|10" and all work 
fine.


What I would like to do is 
to know witch is the called? 121 or 122

So suppose that the number 
121 and 122 ringing at the same time. I answer the first 121. When I try to 
make 
a transfer call I want to jump this call to the second number 122.



But for make this a want 
to use just #1 key without any other digit.
[featuremap]
blindxfer => 
#1

In the same way if I have 
the call to the number 122, pressing the key #1 I want to jump the call to the 
number 121.

Are there any solution or 
workaround?


Many thanks to 
everybody

_
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Re: [asterisk-users] Can I roll my own E911?

[Alex Balashov]

Alex Balashov wrote:

> Adam Moffett wrote:
> 
>> Assuming I only operate in one municipality (I do), and assuming I 
>> made some sort of connection to the emergency services center in this 
>> area, via SIP or a T1 or whatever
> 
> PSAPs won't do VoIP.  In fact, they won't do anything like PRI;  for the 
> most part, T1s to PSAPs are some type of E&M / FGB / FGD.
> 
> I am also not sure that you can actually get an access circuit to a 
> PSAP.  It may be that the only way to reach one is through 911 TCICs in 
> an SS7 interconnection IMT.  Of course, all these E911 companies have to 
> be able to dump calls into PSAPs somehow, but they probably ride the 
> services of some nationwide CLEC in order to do it.

Also, even if you could pull this off, I don't see how rolling your own 
E911 services (and associated reliability issues) could possibly be cost 
-effective compared to using one of the E911 companies.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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[asterisk-users] Quality problems with ISDN PRI

[Carles Pina i Estany]

Hello,

We have an Asterisk server with a TE410P Quad-Span togglable E1/T1/J1
card, 3 SPANs configured and OK and one SPAN unconfigured.

In our tests it works fine, but when it has a big laod of calls (say,
from 40 to 60) we have quality problems: some calls has the sound
cut-off (during the call, voice was not stable)

The IRQ card is alone, CPU load was not high, network was fine for sure.
This server is receiving the calls from SIP channels and routing to the
primaries. It's a HP server, multicore, multiCPU.

I'm wondering if someone has had these kind of problems (quality
problems, sound cut off) with 40 and 60 calls but not with 2 or 3, using
Digium cards.

Bit later I will call to Digium but I thought that here there is lot of
people with lot of experience with these cards.

Thank you,

-- 
Carles Pina i EstanyGPG id: 0x8CBDAE64
http://pinux.info   Manresa - Barcelona

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Re: [asterisk-users] DUNDi and SIP

[Bruce Reeves]

Jeremy,

Did you get this working?



On Thu, Apr 17, 2008 at 7:38 AM, Jeremy Mann <[EMAIL PROTECTED]> wrote:
> I have it working via IAX, when I try changing everything to SIP I can't 
> specify a username and an extension, so it becomes useless.
>
>
>
>  -Original Message-
>  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
>  Sent: Thursday, April 17, 2008 6:51 AM
>  To: Asterisk Users Mailing List - Non-Commercial Discussion
>  Subject: Re: [asterisk-users] DUNDi and SIP
>
>  Jeremy,
>
>  Here is a working sample to compare to. This is an IAX2 setup, but the
>  only difference is in the mapping change IAX2 to SIP. Notice the 4th
>  setting in the mapping? It defines to use the IAX2 peer "priv" with
>  the secret generated of the key defined in the peers section of
>  dundi.conf. When you look at the peer in iax.conf on the remote box,
>  there is no host entry and it uses dbsecret=dundi/secret, the
>
>  dundi.conf
>  priv => dundi-internal,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
>
>  [00:19:66:1C:78:D5] ; Dev Box
>  model = symmetric
>  host = 192.168.99.252
>  inkey = eus
>  outkey = eus
>  include = priv
>  permit = priv
>  qualify = yes
>
>
>  From iax.conf
>  [priv]
>  type=friend
>  dbsecret=dundi/secret
>  context=longdistance
>
>  Hope this helps, in your case Dundi will save you a world of work on
>  configuring that many systems, in fact if you structure Dundi like
>  spokes around a small number of master servers, the config gets real
>  easy.Let me know how it goes.
>
>  On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann <[EMAIL PROTECTED]> wrote:
>  >
>  >
>  >
>  >
>  > I'm a little confused with DUNDi and SIP as the backend channel type:
>  >
>  >
>  >
>  > Dundi.conf:
>  >
>  > [mappings]
>  >
>  > priv => dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
>  >
>  >
>  >
>  > Using the above, the dial string passed to the person on the other box is
>  > SIP/[EMAIL PROTECTED]
>  >
>  >
>  >
>  > How can you use authentication, along with SIP, along with specifying
>  > extension?
>  >
>  >
>  >
>  > My sip.conf has a friend defined:
>  >
>  >
>  >
>  > [priv]
>  >
>  > host=dynamic
>  >
>  > secret=priv
>  >
>  > disallow=all
>  >
>  > allow=ulaw
>  >
>  > canreinvite=no
>  >
>  > nat=no
>  >
>  > context=from-internal\
>  >
>  > type=friend
>  >
>  >
>  >
>  > I need to specify the sip channel to use the priv peer, priv secret, and
>  > pass the extension.  I've tried defining my mapping as:
>  >
>  >
>  >
>  > Priv => dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
>  >
>  >
>  >
>  > But obviously the console on the far end complains that peer
>  > a.b.c.d/${NUMBER} cannot be found.
>  >
>  >
>  >
>  > Thanks for any insight into this.  I'd prefer not having to define a sip
>  > peer per box(I have 25 connected in my dundi cloud), nor would I like to
>  > enable anonymous SIP calls, as I have the ports open to the world for
>  > inbound sip from bandwidth.com
>  >
>  >
>  >
>  >
>  >  
>  >  This e-mail, facsimile, or letter and any files or attachments transmitted
>  > with it contains information that is confidential and privileged. This
>  > information is intended only for the use of the individual(s) and
>  > entity(ies) to whom it is addressed. If you are the intended recipient,
>  > further disclosures are prohibited without proper authorization. If you are
>  > not the intended recipient, any disclosure, copying, printing, or use of
>  > this information is strictly prohibited and possibly a violation of federal
>  > or state law and regulations. If you have received this information in
>  > error, please notify Texas Health Management Group immediately at
>  > 1-817-310-4999. Texas Health Management Group, its subsidiaries, and
>  > affiliates hereby claim all applicable privileges related to this
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>  >
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>
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> disclosures are prohibited without proper authorization. If you are not 

Re: [asterisk-users] Disable transfer on all calls

[Grey Man]

Hi Danila,

You can't turn them transfers off with Asterisk.

The best option is to put a SIP Proxy in front of your Asterisk sever
and block REFER requests.

Regards,

Greyman.

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Re: [asterisk-users] Parsing incoming extension till first @

[Ali Jawad]

Thx again patrick it worked, I used

[google-in]
exten => _.,1,Set(dst=${CUT(EXTEN,@,1)})
exten => _.,1,Dial(SIP/[EMAIL PROTECTED])

while it should have been

[google-in]
exten => _.,1,Set(dst=${CUT(EXTEN,@,1)})
exten => _.,2,Dial(SIP/[EMAIL PROTECTED])

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Re: [asterisk-users] lots of warnings from translate.c

[David Boyd]

On Tue, 2008-04-22 at 12:28 +0200, Francesco Castellano wrote:
> We have a couple of servers with asterisk 1.4.19 and zaptel 1.4.10,
> acting as gateways from SIP to ISDN PRI interfaces. Each has one
> Digium TE420 (with hardware echo cancellation) and one TC400B for
> transcoding, in order to handle 60/90 contemporary calls in peak
> hours.
> 
> In my logs there are hundreds of thousand warnigs per day like these:
> 
> transcode.c: no samples for lintoulaw
> transcode.c: zapg729toalaw did not update samples ###
> 
> Could you suggest me what are the possible causes for that? Are they
> signs of bad audio quality? Any ideas for resolving these issues?
> 
> In addition I can say that we are using a quite large jitter buffer in
> zapata.conf:
> 
> jitterbuffers=16 (=> 0.32s)
> 
> Moreover, it uses the fixed implementation, because when I tried the
> adaptive one I experienced one-way audio.
> Finally I have to note that, using a Siemens IP phone (G.729 no
> AnnexB) in conditions of no load on servers, I could replicate
> non-deterministically (sigh!) each of these problems, with a very
> noisy audio, and a annoying period of silence during the first seconds
> of call.
> 
> Regards,
> Francesco
> 
> PS. Previous versions of asterisk and zaptel presented an identical situation.
> 
Have you tried additional types of phones and if so can you produce the
same non-deterministic problems?


Dave


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Re: [asterisk-users] Parsing incoming extension till first @

[Ali Jawad]

Thanks Patrick this resulted in

   -- Executing [EMAIL PROTECTED]@google-in:1]
Set("Gtalk/jabber1-c06e", "dst=009613041705") in new stack
-- Auto fallthrough, channel 'Gtalk/jabber1-c06e' status is 'UNKNOWN'

It seems to have cut the correct part but I am not sure about the rest
of it, it is causing auto fallthrough with status UNKNOWN

On Tue, Apr 22, 2008 at 1:33 PM, Philipp Kempgen
<[EMAIL PROTECTED]> wrote:
> Ali Jawad schrieb:
>
>
>
>  > When I dial a number it reaches the asterisk switch as [EMAIL 
> PROTECTED]@123.com
>  > what I need to do is to parse the abc and send it to my pstn gateway
>  > as in
>  >
>  > exten => _.,2,Dial(SIP/[EMAIL PROTECTED])
>  >
>  > this does work but I do have a varying number of numbers before the @
>  >
>  > exten => _.,1,Dial(SIP/${EXTEN:0:[EMAIL PROTECTED])
>  >
>  > Well can I use some kind of regular expression to take all numbers
>  > before the first @ and send them to the pstn
>  >
>  > something like
>  >
>  > exten => _.,1,Dial(SIP/${regexp(condition,Exten)[EMAIL PROTECTED])
>
>  core show function CUT
>
>  Regards,
>   Philipp Kempgen
>
>  --
>  amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
> Let's use IT to solve problems and not to create new ones.
>   Asterisk? -> http://www.das-asterisk-buch.de
>
>  Geschäftsführer: Stefan Wintermeyer
>  Handelsregister: Neuwied B 14998
>
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-- 
-- 
With Regards
Ali Jawad System Administrator
http://www.alijawad.org
Phone : +961-01-559031
Mobile : +961-03-041705





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Re: [asterisk-users] Parsing incoming extension till first @

[Philipp Kempgen]

Ali Jawad schrieb:

> When I dial a number it reaches the asterisk switch as [EMAIL 
> PROTECTED]@123.com
> what I need to do is to parse the abc and send it to my pstn gateway
> as in
> 
> exten => _.,2,Dial(SIP/[EMAIL PROTECTED])
> 
> this does work but I do have a varying number of numbers before the @
> 
> exten => _.,1,Dial(SIP/${EXTEN:0:[EMAIL PROTECTED])
> 
> Well can I use some kind of regular expression to take all numbers
> before the first @ and send them to the pstn
> 
> something like
> 
> exten => _.,1,Dial(SIP/${regexp(condition,Exten)[EMAIL PROTECTED])

core show function CUT

Regards,
  Philipp Kempgen

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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Re: [asterisk-users] Can I roll my own E911?

[Alex Balashov]

Adam Moffett wrote:

> Assuming I only operate in one municipality (I do), and assuming I made 
> some sort of connection to the emergency services center in this area, 
> via SIP or a T1 or whatever

PSAPs won't do VoIP.  In fact, they won't do anything like PRI;  for the 
most part, T1s to PSAPs are some type of E&M / FGB / FGD.

I am also not sure that you can actually get an access circuit to a 
PSAP.  It may be that the only way to reach one is through 911 TCICs in 
an SS7 interconnection IMT.  Of course, all these E911 companies have to 
be able to dump calls into PSAPs somehow, but they probably ride the 
services of some nationwide CLEC in order to do it.

> does asterisk have a way for me to send the E911 address data?

Asterisk does not have an ALI interface as far as I know.  But it is 
possible to patch the call through various appliances that do.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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[asterisk-users] lots of warnings from translate.c

[Francesco Castellano]

We have a couple of servers with asterisk 1.4.19 and zaptel 1.4.10,
acting as gateways from SIP to ISDN PRI interfaces. Each has one
Digium TE420 (with hardware echo cancellation) and one TC400B for
transcoding, in order to handle 60/90 contemporary calls in peak
hours.

In my logs there are hundreds of thousand warnigs per day like these:

transcode.c: no samples for lintoulaw
transcode.c: zapg729toalaw did not update samples ###

Could you suggest me what are the possible causes for that? Are they
signs of bad audio quality? Any ideas for resolving these issues?

In addition I can say that we are using a quite large jitter buffer in
zapata.conf:

jitterbuffers=16 (=> 0.32s)

Moreover, it uses the fixed implementation, because when I tried the
adaptive one I experienced one-way audio.
Finally I have to note that, using a Siemens IP phone (G.729 no
AnnexB) in conditions of no load on servers, I could replicate
non-deterministically (sigh!) each of these problems, with a very
noisy audio, and a annoying period of silence during the first seconds
of call.

Regards,
Francesco

PS. Previous versions of asterisk and zaptel presented an identical situation.

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[asterisk-users] features.conf Problem with DTMF_sequence

[Sergey Shumeyko]

Hello,
I have following problem with my Asterisk installation (version 1.6.0. beta
7.1). I want to assign start record conversation to #7 and stop record
conversation to #8, but it isn't working (on previous Asterisk 1.2.17 it was
working fine).  When I assign those functions to 7/8 (without #)
correspondingly it also works fine, but it works only from caller side. I
would appreciate very much if somebody can take a look at my configuration
below and give me comments what I am doing wrong.


My configuration:

features.conf:
[featuremap]
blindxfer => 111222333  ; Blind transfer  (default is #)
disconnect => 444555666  ; Disconnect  (default is *)
;automon => *1 ; One Touch Record a.k.a. Touch Monitor
;atxfer => *2  ; Attended transfer
;parkcall => #72; Park call (one step parking)
;automixmon => *3; One Touch Record a.k.a. Touch MixMonitor


[applicationmap]
testfeature => #9,peer/both,Playback,beep

record_start => #7,self/both,Macro,RECORD_START   ; doesn't work, peer or
self doesn't make difference
record_stop => #8,self/both,Macro,RECORD_STOP ; doesn't work, peer or
self doesn't make difference
;record_start => 7,self/both,Macro,RECORD_START; works fine only from
caller side
;record_stop => 8,self/both,Macro,RECORD_STOP  ; works fine only from
caller side

extension.conf:
[general]
autofallthrough=yes

[macro-RECORD_START]
exten => s,1,Playback(beep)
exten => s,2,AGI(${AGI_SERVER}${RECORD_AGI}?MODE=start)

[macro-RECORD_STOP]
exten => s,1,AGI(${AGI_SERVER}${RECORD_AGI}?MODE=stop)


-- 
Regards,  Shuma
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[asterisk-users] Parsing incoming extension till first @

[Ali Jawad]

Hi All

When I dial a number it reaches the asterisk switch as [EMAIL PROTECTED]@123.com
what I need to do is to parse the abc and send it to my pstn gateway
as in

exten => _.,2,Dial(SIP/[EMAIL PROTECTED])

this does work but I do have a varying number of numbers before the @

exten => _.,1,Dial(SIP/${EXTEN:0:[EMAIL PROTECTED])

Well can I use some kind of regular expression to take all numbers
before the first @ and send them to the pstn

something like

exten => _.,1,Dial(SIP/${regexp(condition,Exten)[EMAIL PROTECTED])

thx

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Re: [asterisk-users] Conditional include=> ?

[Philipp Kempgen]

Steve Davies schrieb:

> Does anyone have a clever method of doing a conditional "include =>"
> line in the dialplan?
> 
> I want to include a bunch of standard contexts, but in the middle of
> the bunch have one or more conditionally included, a bit like:
> 
> [default]
> include => start-here
> include => then-here
> if $[{COMPANY} = A]
> include => company-A
> endif
> if $[{COMPANY} = B]
> include => company-B
> endif
> if $[{COMPANY} = C]
> include => company-C
> endif
> include => end-here
> 
> 
> The list of conditional variables, and of static includes is
> potentially large, so I do not want to create several separate sets of
> contexts with lots of repetition as it is likely to get out-of-control
> very quickly.

I suppose #ifdefs and a preprocessor like cpp is not what you're
looking for? You didn't make clear at what stage you want these
includes to be replaced / evaluated. I guess you want Asterisk to
do it.

Regards,
  Philipp Kempgen

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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[asterisk-users] OT: Linksys devices send incorrect REGISTER

[CSB]

We have a situation where various Linksys devices lose their registration
with Asterisk periodically. This seems to occur only during the day when the
system is busy. Having reviewed the logs, it appears that the device
response is out of sync with what Asterisk expects. This occurs with various
Linksys devices running various firmware (User-Agent: Linksys/SPA942-5.2.5,
Linksys/SPA2102-3.3.6). An example:

 

10:31:05 

Register received nonce 7392c294 response c1892f1c1bd0e56aa85f03a32c5f14d1

trying sent back

401 sent back nonce 725162e4

Register received nonce 7392c294 response c1892f1c1bd0e56aa85f03a32c5f14d1

10:31:06

Trying sent back

401 sent back nonce 5774e85e

Register received nonce 725162e4 response bd8943615bc4239b8f90533a78ef4ccb

Trying sent back

401 sent back nonce 56e89c8a

Register received nonce 725162e4 response bd8943615bc4239b8f90533a78ef4ccb

 

On the face of it this seems to be a Linksys issue but I wondered if anyone
else had experienced something similar? Since if occurs only during busy
times I wonder if Asterisk is taking longer than the Linksys expects to
reply with a 401 which is re-transmitting the Register but by then the nonce
is stale.

 

Anyway, any suggestions appreciated.

 

Asterisk 1.2.27

 

Regards

 

Cameron

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Re: [asterisk-users] Conditional include=> ?

[Tzafrir Cohen]

On Tue, Apr 22, 2008 at 09:55:37AM +0100, Steve Davies wrote:
> Hi,
> 
> Does anyone have a clever method of doing a conditional "include =>"
> line in the dialplan?
> 
> I want to include a bunch of standard contexts, but in the middle of
> the bunch have one or more conditionally included, a bit like:
> 
> [default]
> include => start-here
> include => then-here
> if $[{COMPANY} = A]
> include => company-A
> endif
> if $[{COMPANY} = B]
> include => company-B
> endif
> if $[{COMPANY} = C]
> include => company-C
> endif
> include => end-here
> 
> 
> The list of conditional variables, and of static includes is
> potentially large, so I do not want to create several separate sets of
> contexts with lots of repetition as it is likely to get out-of-control
> very quickly.

A different approach:

[company-base](!)
; common settings

[company-A](company-base)
; specific for company A

[company-B](company-base)
; specific for company B

[company-C](company-base)
; specific for company C


Keep in mind you can also use:

[sub-template](!,base-template)

And:

[context](template1,template2)

But one limitation is that you can only add: no way to remove line added
by a template your context uses.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] re-invite (bypass asterisk) post call establishment

[Steve Davies]

2008/4/22 Benjamin Jacob <[EMAIL PROTECTED]>:
[snip]
>
> So, my question : once the SDPs are exchanged, what will happen to the DTMFs
> sent by Asterisk using sendDTMF or the D option in dial.
>
[snip]

As far as I can tell, the D() option will be executed before the
re-invite takes place, so Asterisk will still be in-line. I believe
that the dial is not considered "complete/connected" until the D() is
finished.

Cheers,
Steve

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[asterisk-users] Conditional include=> ?

[Steve Davies]

Hi,

Does anyone have a clever method of doing a conditional "include =>"
line in the dialplan?

I want to include a bunch of standard contexts, but in the middle of
the bunch have one or more conditionally included, a bit like:

[default]
include => start-here
include => then-here
if $[{COMPANY} = A]
include => company-A
endif
if $[{COMPANY} = B]
include => company-B
endif
if $[{COMPANY} = C]
include => company-C
endif
include => end-here


The list of conditional variables, and of static includes is
potentially large, so I do not want to create several separate sets of
contexts with lots of repetition as it is likely to get out-of-control
very quickly.

Help?
Steve

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[asterisk-users] Strange SIP packet

[fadey]

Hi, everyone. 
I'm testing a SIP cable modem with asterisk. The REGISTER packet it
sends has varios "Authorization" sections, which I have never seen
before. Asterisk sends back "401 Unauthorized". I've checked username
and password both in cable modem config and in asterisk sip.conf. They
are correct. So, is the reason for "401 Unauthorized" those multiple
"Authorization" sections? If anyone is more or less familiar with SIP,
please could you take a look at the packet trace below. Is it ok? 
Thanks in advance. 

15:06:48.346157 IP 10.2.0.63.5060 > 192.168.0.4.5060: UDP, length 1345 
E..]\...?.O. 
..?.I".REGISTER sip:192.168.0.4 SIP/2.0 
From:
968953939;tag=94b7ef58-a02003f-13c4-45026-20-3eb775bd-20 
To: 968953939 
Call-ID: 94b7b630-a02003f-13c4-45026-20-6733142f-20 
CSeq: 9898 REGISTER 
Via: SIP/2.0/UDP 10.2.0.63:5060;branch=z9hG4bK-63b06-185692fe-78a7e24 
Max-Forwards: 70 
Supported: timer,replaces,join,100rel 
User-Agent: ARRIS-TM501B release v.05.02.0X SN/0015960DBC24 
Contact: 968953939 
Authorization: Digest
username="968953939",realm="asterisk",nonce="5e8e5c3a",uri="sip:192.168.0.4",response="918f7f23dd8ae8e8fc9465940e8914db",algorithm=MD5
 
Authorization: Digest
username="968953939",realm="asterisk",nonce="5f3ff920",uri="sip:192.168.0.4",response="d3acf42b9afa65708936b525bb912f36",algorithm=MD5
 
Authorization: Digest
username="968953939",realm="asterisk",nonce="60b3eb72",uri="sip:192.168.0.4",response="715c6da5586490ee90f16b39f57b98ac",algorithm=MD5
 
Authorization: Digest
username="968953939",realm="asterisk",nonce="637cb9c0",uri="sip:192.168.0.4",response="49a55de9aab1776c1f7f081465a8c593",algorithm=MD5
 
Authorization: Digest
username="968953939",realm="asterisk",nonce="2d73aae7",uri="sip:192.168.0.4",response="1dd245c05ade19d7d4fc85c21e5e86fe",algorithm=MD5
 
Allow: INVITE,ACK,BYE,CANCEL,NOTIFY,PRACK,UPDATE,OPTIONS 
Content-Length: 0


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Re: [asterisk-users] Basic Possiblity Question.

[rupak shrestha]

Hi Steve thanks for your suggestion.
 
Just want to check with you if the Internet has a Nice document regarding 
this.i will have no problems with open vpn as i have installed it b4.It's only 
the SIP and IAX2 which you mentioned abt.and i have no idea what these 2 things 
are.do you recommend me a document which can make me understand what is 
IAX2,what is SIP.
Appreciate your help
Thank you.> Date: Mon, 21 Apr 2008 13:21:50 -0400> From: [EMAIL PROTECTED]> To: 
asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Basic Possiblity 
Question.> > I would look at setting up OpenVPN on each of the Asterisk boxen 
and> running SIP between them. I have read that IAX2 is much better now,> but I 
have had many major voice quality issues with it.> > With OpenVPN, all Asterisk 
boxen appear to each other as being on the> same subnet. This gives you ease in 
using SIP and added security on> many different facets.> > Thanks,> Steve 
Totaro> > On 4/21/08, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:> >> > Sure - 
read up on IAX for a few good points.> >> > PaulH> >> > > rupak shrestha 
<[EMAIL PROTECTED]> wrote:> > >> > >> > > Hi all, i have a basic question on 
asterisk.The below is my scenerao.> > > I have my sales offices around the 
globe.Theyare all connected with> > > Speed Internet connection.I don't mind 
installing 1 asterisk box in each> > > site.i don't mind using IP phone.i just 
wanted to call them for free at> > > the cost of existing internet connectionwe 
have at each site.All the> > > asterisk box will be connected with TCP/IP with 
one of it's NIC card> > > having a WAN connectivity.is it possible with 
asterisk.Please let me> > > know.Thanx> > > 
_> > > Going 
green? See the top 12 foods to eat organic.> > > 
http://green.msn.com/galleries/photos/photos.aspx?gid=164&ocid=T003MSN51N> > > 
1653A> >> > ___> > -- Bandwidth and 
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[asterisk-users] caller groups

[gilbert saunders]

hi
   
  i have two different phone lines connected to my asterisk server but i want 
one line  to be used only by a certain phone for outbound dialing and receiving 
calls...
   
  i know how to do it but then i have to make that phone known in another group 
and i dont think that that phone would be able to do a pick up anywhere with 
the *8 key or am i wrong
   
  and is there any other way that i can do it that only that particular phone 
can use that particular phone line? 
   
  thanks

   
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