[Freeswitch-users] SIP Authentication
Ali Al-Rubaie
kerrada2003 at yahoo.com
Mon Feb 9 07:08:21 PST 2009
Thanks so much Anthony but I have one more question:
I was checking the source file sofia_reg.c and it seems that the code had been written iin such a way that FreeSWITCH can authenticate SIP agents based on RFC2069 and RFC2617. Is that conclusion correct?
Thanks in advance,
Message: 2
Date: Thu, 5 Feb 2009 10:46:54 -0600
From: Anthony Minessale <anthony.minessale at gmail.com>
Subject: Re: [Freeswitch-users] SIP Authentication
To: freeswitch-users at lists.freeswitch.org
Message-ID:
<191c3a030902050846o60047c30pa2890707eae386d6 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
It's optional for us but it's mandatory for the client if we exercise
the
option which we have opted to always do =D
There is no way in the code to disable sending it because we prefer the more
secure version of SIP auth.
So again it's a bug in the client for not following the protocol. It would
be considered a feature in FreeSWITCH to support limping for the sake of
this broken client and we currently do not have any plans for implementing
this feature.
On Thu, Feb 5, 2009 at 10:34 AM, Ali Al-Rubaie
<kerrada2003 at yahoo.com>wrote:
>
> We're using HelpCaster softphone.
>
> The issue here is that in Digest Authentication, if the server sends the
> parameter "qop" in the challenge then the client should respond
with the
> "cnonce" parameter. The parameter "qop" is optional in
Digest Auth. So the
> question here is that, can we configure FreeSWITCH so that it will not
send
> "qop" in the challenge?
>
> Thanks!
>
> --- On *Wed, 2/4/09, freeswitch-users-request at lists.freeswitch.org <
> freeswitch-users-request at lists.freeswitch.org>* wrote:
>
> From: freeswitch-users-request at lists.freeswitch.org <
> freeswitch-users-request at lists.freeswitch.org>
> Subject: Freeswitch-users Digest, Vol 32, Issue 39
> To: freeswitch-users at lists.freeswitch.org
> Date: Wednesday, February 4, 2009, 2:05 PM
>
> Send Freeswitch-users mailing list submissions to
> freeswitch-users at lists.freeswitch.org
>
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> When replying, please edit your Subject line so it is more specific
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>
>
> Today's Topics:
>
> 1. Re: SIP Authentication (Brian West)
> 2. Re: origainate through sofia gateway (Michael Collins)
> 3. Recording background music and voice is out of sync (Daniel Liang)
> 4. Re: Q931 decoding Update (Gopalakrishnan A.N)
> 5. mod_limit (Chav Paskov)
> 6. Re: mod_limit (Michael Collins)
> 7. Re: mod_limit (Chav
> Paskov)
> 8. Re: mod_limit (Michael Collins)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Wed, 4 Feb 2009 10:52:45 -0600
> From: Brian West <brian at freeswitch.org>
> Subject: Re: [Freeswitch-users] SIP Authentication
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <7DAC91F6-2FD3-464D-AA81-321EBCADC8C0 at freeswitch.org>
> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes
>
> What client is this? I also notice we receive port 3458 and reply to
> port 1059...
>
> /b
>
> On Feb 4, 2009, at 10:17 AM, Ali Al-Rubaie wrote:
>
> > What I have noted is that the client does not send the values for
> > "cnonce" and "nc" in the response. I'm not
sure if
> this is the
> > reason, however how this problem can be solved?
> >
> > Thanks,
> >
> > Ali
>
>
>
>
> ------------------------------
>
> Message:
> 2
> Date: Wed, 4 Feb 2009 09:41:07 -0800
> From: Michael Collins <msc at freeswitch.org>
> Subject: Re: [Freeswitch-users] origainate through sofia gateway
> To: freeswitch-users at lists.freeswitch.org
> Message-ID:
> <87f2f3b90902040941r61d669aaie949aa7cc8578a9a at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> I'll make sure the substance of this is in the wiki and I'll look
for
> references to the deprecated way and remove those.
> -MC
>
> On Wed, Feb 4, 2009 at 6:09 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > Where did you learn how to use js this way?
> > session.originate is being misused here and is depricated and may be
> > removed.
> >
> > the first arg to session.originate is either undefined or a
*different*
> > session (the a leg)
> >
> > session1 = new Session();
> > session1.originate(undefined,
> >
> "{ignore_early_media=true}user/1008 at 192.168.1.122");
> >
> >
>
session1.setVariable("effective_caller_id_number","fixed0248b");
> >
> > //once you have session1 when you originate session2 you pass
session1 as
> > the arg
> > // the effective_caller_id is taken from session1
> >
> > session2 = new Session();
> > session2.originate(session1,
> "sofia/gateway/halonet/0225490317");
> >
> > Anyway this whole code is depricated in favor of this:
> >
> > session1 = new
> Session("{ignore_early_media=true}user/1008 at 192.168.1.122");
> > if (session1.ready()) {
> >
>
session1.setVariable("effective_caller_id_number","fixed0248b");
> > session2 = new
Session("sofia/gateway/halonet/0225490317",
> session1);
> > }
> >
> > and could be further refactored down to this:
> >
> > session1 = new
> Session("{ignore_early_media=true}user/1008 at 192.168.1.122");
> > if
> (session1.ready()) {
> >
>
session1.setVariable("effective_caller_id_number","fixed0248b");
> > session1.execute("bridge",
> "sofia/gateway/halonet/0225490317");
> > }
> >
> > or down to this one line of code that will setup the call detached
from
> the
> > script and exit.
> >
> > var result = apiExecute("originate",
> >
>
"{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122
> > bridge:sofia/gateway/halonet/0225490317 inline");
> >
> > if you dont care about the result and want to exit even before the
call is
> > completed.
> >
> > var result = apiExecute("bgapi", "originate
> >
>
{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122
> > bridge:sofia/gateway/halonet/0225490317 inline");
> >
> >
> >
> > On Wed, Feb 4, 2009 at
> 2:51 AM, Jacek Sokulski
> <jsokulski at dotsystems.pl>
> > wrote:
> >>
> >> We have tried setting both effective_caller_id_number and
> >> origination_caller_id_number:
> >>
> >>
> >>
>
session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15);
> >> but the problem still exists. The solution we have found for the
case
> >> when we originate two calls, local and external, is as follow:
> >>
> >> session1 = new Session();
> >>
>
session1.originate(session1,"user/1003 at 192.168.1.122",15);//local
> >> if(session1.ready()) {
> >>
session1.execute("execute_extension","00930691688627
> XML
> >> default");//external
> >> }
> >>
> >> so the external call goes through the dialplan.
> >> It does not work if both calls are external. One possible
solution
> could
> >>
> be
> >> to pass the originating call through dialplan (loopback?) but we
have
> not
> >> managed
> >> to figure out how to do it.
> >>
> >> Thanks
> >> Jacek
> >>
> >> Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner
pisze:
> >> > Oops! Well, fortunately I don't use that voip provider
> anymore (nor the
> >> > script).
> >> >
> >> > Thanks Brian.
> >> >
> >> > Nicolas
> >> >
> >> > On Tue, Feb 3, 2009 at 2:25 PM, Brian West
> <brian at freeswitch.org> wrote:
> >> > > YOU should NEVER use this method or call setCallerData
at
> all you
> >> > > should use the correct methods to override the
callerid.
> >> > >
> >> > > If its a B-Leg born from an A-Leg you use these on the
on
> the A-Leg:
> >> > >
> >> > >
> >> >
> >
> http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name
> >> > >
> >> > >
>
http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number
> >> > >
> >> > > If you're originating you use this:
> >> > >
> >> > >
> >> > >
>
http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name
> >> > >
> >> > >
>
http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number
> >> > >
> >> > > /b
> >> >
> >> > _______________________________________________
> >> > Freeswitch-users mailing list
> >> > Freeswitch-users at lists.freeswitch.org
> >> >
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>
> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
<MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
<sip%3A888 at conference.freeswitch.org>
> >
> iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org
<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> ------------------------------
>
> Message: 3
> Date: Wed, 4 Feb 2009 09:43:10 -0800
> From: "Daniel Liang" <Daniell at airg.com>
> Subject: [Freeswitch-users] Recording background music and voice is
> out of sync
> To: <freeswitch-users at lists.freeswitch.org>
> Message-ID:
> <0B02E756F603CC409EB553879B090CC80A23EBB5 at HPEXCHVS01.exchange.airg>
> Content-Type: text/plain; charset="us-ascii"
>
> What I did was the following:
>
> First, I sent the
> playback command:
>
> call-command: execute
> execute-app-name: playback
> execute-app-arg: <filename>
>
> Then I send uuid_record (Sorry, it was not Record command):
>
> api uuid_record <uuid> start <filename> 120
>
> I also tried replacing the playback command with:
> api uuid_displace <uuid> start <filename> 0 mux
>
> But the end results are the same. The recorded user's voice is about
0.5
> second behind the expected result.
>
> Thanks,
> Daniel
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Brian West
> Sent: February 3, 2009 6:36 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Recording background music and voice is
> outof sync
>
> Can you show us an example of how you're doing this? Playback and
> Record aren't async so
> you'll need to show us how you're doing
> this.
>
> Also don't hijack threads you hit replay on the one "Re:
[Freeswitch-
> users] FreeSwitch setup as a "Dumb" SBC" as the subject,
deleted
> the
> subject and started a new body. That hijacks the thread and that can
> cause your problem to go ignored in some cases if people aren't
> interested in the thread topic depending on how their reader threads the
> emails.
>
> Please click new message and type freeswitch- users at lists.freeswitch.org
> in and then input your subject and body to start a new thread.
>
> Thanks,
> Brian West
> FreeSWITCH.org
>
>
> On Feb 3, 2009, at 8:21 PM, Daniel Liang wrote:
>
> > Hi,
> >
> > I was trying to record a background music with a user's voice at
the
> > same time. I did a playback and started recording. But the recorded
> > user's voice and the background music is about 0.5 second out of
sync.
>
> > I also tried
> to use uuid_displace instead of playback, but I got the
> > same result.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org <http://www.freeswitch.org/>
>
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>
> ------------------------------
>
> Message: 4
> Date: Wed, 4 Feb 2009 23:26:14 +0530
> From: "Gopalakrishnan A.N" <saigop at gmail.com>
> Subject: Re: [Freeswitch-users] Q931 decoding Update
> To: freeswitch-users at lists.freeswitch.org
> Message-ID:
>
> <2ea4d47e0902040956v75c5472foa4649c50b7340484 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi,
> Its a awesome. Can the packet capturing be done with event socket?
>
> --
> Thank you with regards,
> Gopal,
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> ------------------------------
>
> Message: 5
> Date: Wed, 04 Feb 2009 09:59:48 -0800
> From: Chav Paskov <chavpaskov at shaw.ca>
> Subject: [Freeswitch-users] mod_limit
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <4989D794.1010805 at shaw.ca>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Hi ,
> is it possible to use mod_limit in case if the end point is not
> registered / gateway for
> example/.
> Regards
> Chav
>
>
>
> ------------------------------
>
> Message: 6
> Date: Wed, 4 Feb 2009 10:06:52 -0800
> From: Michael Collins <msc at freeswitch.org>
> Subject: Re: [Freeswitch-users] mod_limit
> To: freeswitch-users at lists.freeswitch.org
> Message-ID:
> <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov <chavpaskov at shaw.ca>
wrote:
> > Hi ,
> > is it possible to use mod_limit in case if the end point is not
> > registered / gateway for example/.
>
> Could you add some detail to this question? What are you trying to do?
> (mod_limit may or may not work, but there might be another solution
> which is why I am asking.)
>
> -MC
>
> > Regards
> > Chav
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> >
> Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> ------------------------------
>
> Message: 7
> Date: Wed, 04 Feb 2009 10:54:56 -0800
> From: Chav Paskov <chavpaskov at shaw.ca>
> Subject: Re: [Freeswitch-users] mod_limit
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <4989E480.1080105 at shaw.ca>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Michael Collins wrote:
> > On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov
<chavpaskov at shaw.ca>
> wrote:
> >
> >> Hi ,
> >> is it possible to use mod_limit in case if the end point is not
> >> registered / gateway for example/.
> >>
> >
> > Could you add some detail to this question? What are you trying to
do?
> >
> (mod_limit may or may not work, but there might be another solution
> > which is why I am asking.)
> >
> > -MC
> >
> >
> >> Regards
> >> Chav
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> i have few gateways under my ACL that
> are allowed to send calls to FS,
> but i want to be able to enforce "capacity" policy on the
traffic
> coming from any one of them depending on total termination capacity on
> my termination end.
> Let say GW 1 has to be limited to make 10 simultaneous calls while GW2
> could make up to 30 and so on.
> Regards
> Chav
>
>
>
> ------------------------------
>
> Message: 8
> Date: Wed, 4 Feb 2009 11:05:09 -0800
> From: Michael Collins <msc at freeswitch.org>
> Subject: Re: [Freeswitch-users] mod_limit
> To: freeswitch-users at lists.freeswitch.org
> Message-ID:
> <87f2f3b90902041105l50f51f08t230bab8d69eefb4e at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> On Wed, Feb 4, 2009 at 10:54 AM, Chav Paskov <chavpaskov at shaw.ca>
wrote:
> > Michael Collins wrote:
> >> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov
<chavpaskov at shaw.ca>
> wrote:
> >>
> >>> Hi
> ,
> >>> is it possible to use mod_limit in case if the end point is
not
> >>> registered / gateway for example/.
> >>>
> >>
> >> Could you add some detail to this question? What are you trying
to do?
> >> (mod_limit may or may not work, but there might be another
solution
> >> which is why I am asking.)
> >>
> >> -MC
> >>
> >>
> >>> Regards
> >>> Chav
> >>>
> >>> _______________________________________________
> >>> Freeswitch-users mailing list
> >>> Freeswitch-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >> _______________________________________________
> >>
> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> > i have few gateways under my ACL that are allowed to send calls to
FS,
> > but i want to be able to enforce "capacity" policy on the
> traffic
> > coming from any one of them depending on total termination capacity
on
> > my termination end.
> > Let say GW 1 has to be limited to make 10 simultaneous calls while
GW2
> > could make up to 30 and so on.
>
> I'm sure that this is possible. I don't personally have a way to
test
> all of this but I know that a number of our users are doing things
> like this currently. Can you hop on to the IRC channel? #freeswitch on
> irc.freenode.net. A lot of people there can help with
> this one.
>
> -MC (IRC: mercutioviz)
>
> > Regards
> > Chav
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> ------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> End of Freeswitch-users Digest, Vol 32, Issue 39
> ************************************************
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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End of Freeswitch-users Digest, Vol 32, Issue 50
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