asterisk users - Jul 2009 - SIP IP-Trunk to be authenticated based on username and password, not IP address

Hi List;

How can one Asterisk Box A to send a SIP call for another Asterisk Box B, and
that call to be authorized based on the username and password, and not on the IP
(as the IP address of the source is not known because it keep changing)? I think
the trick in the Dial command, how to write it properly in a way that other
Asterisk Box can recognize the sip username and password which are existed in
its sip.conf?

Anyone can advise me for the main needed thing to be done to acheive this?

By the way: I succeed to let Polycom SIP phone to place a call via Asterisk
without registeration, and without setting the IP address also. But from
Asterisk Box to another Asterisk Box, I can not until now. Any help?

Regards
Bilal

And these mistery appear with Asterisk to Asterisk and does not appear between
Asterisk to other products or from any IP Phone to Asterisk? How?

Just because call came from Asterisk and was sent to Asterisk it is going to
suffer this? While if it was originated from IP Phone then no problem? And if it
is originated from Asterisk and sent for other softswitch product then it is
working without need for registeration and without need to set the IP address
(just it recognize the username and password)!!!

Why the Asterisk Box B does not recognize the username and password of the SIP
call coming from Asterisk Box A, while it can recognize this if the originator
was SIP IP Phone :) ?

Regards
Bilal


--- On Mon, 7/6/09, Thierry Wehr <t.wehr at widevoip.com> wrote:
> From: Thierry Wehr <t.wehr at widevoip.com>
> Subject: RE: SIP IP-Trunk to be authenticated based on username and
password, not IP address
> To: "'bilal ghayyad'" <bilmar_gh at yahoo.com>
> Date: Monday, July 6, 2009, 7:36 AM
> These are mistery of Asterisk like
> order codecs are negotiated
> 
> 
> Thierry Wehr
> Projets Sp?ciaux
> t.wehr at widevoip.com
> T?l: +33 (0)390 400 675
> Fax: +33 (0)390 400 676
> http://www.widevoip.com
> 
> 
> -----Message d'origine-----
> De?: bilal ghayyad [mailto:bilmar_gh at yahoo.com]
> 
> Envoy??: lundi 6 juillet 2009 12:56
> ??: t.wehr at widevoip.com
> Objet?: RE: SIP IP-Trunk to be authenticated based on
> username and password,
> not IP address
> 
> 
> Thanks a lot for your kindly answer and help.
> 
> That is fine, but need to register A on B.
> 
> The idea that we were able to place calls from Asterisk A
> to a softswitch
> using SIP trunk without registeration and it worked. But
> the softswitch was
> not asterisk. So this is possible in the softswitch, and I
> would to do same
> with the Asterisk.
> 
> >From the other side, I am surprised about:
> 
> Why the SIP IP Phone (like Polycom) can place a call via
> Asterisk without
> registeration and without setting the IP address in the
> host (actually the
> host=dynamic), so why this is not possible when Asterisk A
> send for Asterisk
> B? Why does not to be considered same as SIP IP Phone is
> sending for
> Asterisk the call and it is not registered on Asterisk (and
> its IP is not
> set also in the host parameter), but it succeed by the
> username and secret
> authentication. 
> 
> Can u help? And thanks a lot for your already helped :)
> 
> Regards
> Bilal
> 
> 
> --- On Mon, 7/6/09, Thierry Wehr <t.wehr at widevoip.com>
> wrote:
> 
> > From: Thierry Wehr <t.wehr at widevoip.com>
> > Subject: RE: SIP IP-Trunk to be authenticated based on
> username and
> password, not IP address
> > To: "'bilal ghayyad'" <bilmar_gh at yahoo.com>
> > Date: Monday, July 6, 2009, 6:28 AM
> > You MUST register one asterisk on the
> > other one
> > See examples for config
> > 
> > 
> > Asterisk A ( 10.1.1.1 )
> > 
> > Register => interco:password at 10.1.1.2
> > 
> > [interco]
> > Type=friend
> > Username=interco
> > Secret=password
> > Host=10.1.1.2
> > Contect=incoming-from-asterisk_B
> > 
> > 
> > Asterisk B ( 10.1.1.2 )
> > 
> > [interco]
> > Type=friend
> > Username=interco
> > Secret=password
> > Host=dynamic
> > Contect=incoming-from-asterisk_A
> > 
> > 
> > To dial from A to B
> > 
> > Dial(SIP/interco/${EXTEN})
> > 
> > To dial from B to A
> > 
> > Dial(SIP/interco/${EXTEN})
> > 
> > This must work has it is in production on our side
> > 
> > Thierry Wehr
> > Projets Sp?ciaux
> > t.wehr at widevoip.com
> > T?l: +33 (0)390 400 675
> > Fax: +33 (0)390 400 676
> > http://www.widevoip.com
> > 
> > 
> > -----Message d'origine-----
> > De?: bilal ghayyad [mailto:bilmar_gh at yahoo.com]
> > 
> > Envoy??: lundi 6 juillet 2009 10:03
> > ??: t.wehr at widevoip.com
> > Objet?: RE: SIP IP-Trunk to be authenticated based
> on
> > username and password,
> > not IP address
> > 
> > 
> > The [] same as username, but Asterisk B reject calls
> came
> > from Asterisk A.
> > Anything need to be placed in the Dial command?
> > 
> > Why Asterisk B is not able to authenticate the call
> came
> > from Asterisk A
> > based on the sip username and secret?
> > 
> > Regards
> > Bilal
> > 
> > --- On Mon, 7/6/09, Thierry Wehr <t.wehr at widevoip.com>
> > wrote:
> > 
> > > From: Thierry Wehr <t.wehr at widevoip.com>
> > > Subject: RE: SIP IP-Trunk to be authenticated
> based on
> > username and
> > password, not IP address
> > > To: "'bilal ghayyad'" <bilmar_gh at
yahoo.com>
> > > Date: Monday, July 6, 2009, 3:45 AM
> > > Authentication is based on [],
> > > username, fromuser, secret
> > > 
> > > If [] different from username you must set
> fromuser
> > > 
> > > 
> > > Thierry Wehr
> > > Projets Sp?ciaux
> > > t.wehr at widevoip.com
> > > T?l: +33 (0)390 400 675
> > > Fax: +33 (0)390 400 676
> > > http://www.widevoip.com
> > > 
> > > -----Message d'origine-----
> > > De?: bilal ghayyad [mailto:bilmar_gh at yahoo.com]
> > > 
> > > Envoy??: lundi 6 juillet 2009 09:10
> > > ??: t.wehr at widevoip.com
> > > Objet?: RE: SIP IP-Trunk to be authenticated
> based
> > on
> > > username and password,
> > > not IP address
> > > 
> > > 
> > > That is correct if u mean at destination
> Asterisk, but
> > what
> > > about source
> > > Asterisk? Sure there is something else need to
> be
> > > configured in the SIP
> > > Trunk and maybe in the Dial command?
> > > 
> > > I was think in the "fromuser" parameter to be
> used,
> > what do
> > > u think? (In the
> > > source Asterisk SIP configuration)?
> > > 
> > > Any help?
> > > Regards
> > > Bilal
> > > 
> > > --- On Sun, 7/5/09, Thierry Wehr - WideVOIP
> <t.wehr at widevoip.com>
> > > wrote:
> > > 
> > > > From: Thierry Wehr - WideVOIP <t.wehr at widevoip.com>
> > > > Subject: RE: SIP IP-Trunk to be
> authenticated
> > based on
> > > username and
> > > password, not IP address
> > > > To: "'bilal ghayyad'" <bilmar_gh at
yahoo.com>
> > > > Date: Sunday, July 5, 2009, 6:42 PM
> > > > Hello
> > > > 
> > > > In sip.conf
> > > > 
> > > > [331701010]
> > > > host=dynamic
> > > > username=331701010
> > > > secret=my_password
> > > > context=outhoing
> > > > 
> > > > this must do the job
> > > > 
> > > > a++
> > > > 
> > > > 
> > > > Thierry Wehr
> > > > Directeur Technique
> > > > t.wehr at widevoip.com
> > > > T?l: +33 (0)390 400 675
> > > > Fax: +33 (0)390 400 676
> > > > http://www.widevoip.com
> > > > 
> > > > 
> > > > 
> > > > 
> > > > -----Message d'origine-----
> > > > De?: bilal ghayyad [mailto:bilmar_gh at yahoo.com]
> > > > 
> > > > Envoy??: dimanche 5 juillet 2009 23:07
> > > > ??: asterisk-users at lists.digium.com
> > > > Cc?: t.wehr at widevoip.com;
> > > > joakimsen at gmail.com;
> > > > amitsalunkhe21 at gmail.com
> > > > Objet?: SIP IP-Trunk to be authenticated
> based
> > on
> > > username
> > > > and password, not
> > > > IP address
> > > > 
> > > > 
> > > > Hi List;
> > > > 
> > > > How can one Asterisk Box A to send a SIP
> call
> > for
> > > another
> > > > Asterisk Box B,
> > > > and that call to be authorized based on the
> > username
> > > and
> > > > password, and not
> > > > on the IP (as the IP address of the source
> is
> > not
> > > known
> > > > because it keep
> > > > changing)? I think the trick in the Dial
> command,
> > how
> > > to
> > > > write it properly
> > > > in a way that other Asterisk Box can
> recognize
> > the
> > > sip
> > > > username and password
> > > > which are existed in its sip.conf?
> > > > 
> > > > Anyone can advise me for the main needed
> thing to
> > be
> > > done
> > > > to acheive this?
> > > > 
> > > > By the way: I succeed to let Polycom SIP
> phone
> > to
> > > place a
> > > > call via Asterisk
> > > > without registeration, and without setting
> the
> > IP
> > > address
> > > > also. But from
> > > > Asterisk Box to another Asterisk Box, I can
> not
> > until
> > > now.
> > > > Any help?
> > > > 
> > > > Regards
> > > > Bilal
> > > > 
> > > > 
> > > > 
> > > > ? ? ? 
> > > > 
> > > > 
> > > 
> > > 
> > > ? ? ? 
> > > 
> > > 
> > 
> > 
> > ? ? ? 
> > 
> > 
> 
> 
> ? ? ? 
> 
>