[Freeswitch-users] SIP Authentication
Ali Al-Rubaie
kerrada2003 at yahoo.com
Thu Feb 5 08:34:08 PST 2009
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
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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
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 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
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> 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.
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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
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> 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
>
------------------------------
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