<table cellspacing="0" cellpadding="0" border="0" ><tr><td valign="top" style="font: inherit;"><br>We're using HelpCaster softphone.<br><br>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?<br><br>Thanks!<br><br>--- On <b>Wed, 2/4/09, freeswitch-users-request@lists.freeswitch.org <i><freeswitch-users-request@lists.freeswitch.org></i></b> wrote:<br><blockquote style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;">From: freeswitch-users-request@lists.freeswitch.org <freeswitch-users-request@lists.freeswitch.org><br>Subject: Freeswitch-users Digest, Vol 32, Issue 39<br>To: freeswitch-users@lists.freeswitch.org<br>Date: Wednesday, February 4, 2009,
2:05 PM<br><br><pre>Send Freeswitch-users mailing list submissions to<br> freeswitch-users@lists.freeswitch.org<br><br>To subscribe or unsubscribe via the World Wide Web, visit<br> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>or, via email, send a message with subject or body 'help' to<br> freeswitch-users-request@lists.freeswitch.org<br><br>You can reach the person managing the list at<br> freeswitch-users-owner@lists.freeswitch.org<br><br>When replying, please edit your Subject line so it is more specific<br>than "Re: Contents of Freeswitch-users digest..."<br><br><br>Today's Topics:<br><br> 1. Re: SIP Authentication (Brian West)<br> 2. Re: origainate through sofia gateway (Michael Collins)<br> 3. Recording background music and voice is out of sync (Daniel Liang)<br> 4. Re: Q931 decoding Update (Gopalakrishnan A.N)<br> 5. mod_limit (Chav Paskov)<br> 6. Re: mod_limit (Michael Collins)<br> 7. Re: mod_limit (Chav
Paskov)<br> 8. Re: mod_limit (Michael Collins)<br><br><br>----------------------------------------------------------------------<br><br>Message: 1<br>Date: Wed, 4 Feb 2009 10:52:45 -0600<br>From: Brian West <brian@freeswitch.org><br>Subject: Re: [Freeswitch-users] SIP Authentication<br>To: freeswitch-users@lists.freeswitch.org<br>Message-ID: <7DAC91F6-2FD3-464D-AA81-321EBCADC8C0@freeswitch.org><br>Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes<br><br>What client is this? I also notice we receive port 3458 and reply to <br>port 1059...<br><br>/b<br><br>On Feb 4, 2009, at 10:17 AM, Ali Al-Rubaie wrote:<br><br>> What I have noted is that the client does not send the values for <br>> "cnonce" and "nc" in the response. I'm not sure if<br>this is the <br>> reason, however how this problem can be solved?<br>><br>> Thanks,<br>><br>> Ali<br><br><br><br><br>------------------------------<br><br>Message:
2<br>Date: Wed, 4 Feb 2009 09:41:07 -0800<br>From: Michael Collins <msc@freeswitch.org><br>Subject: Re: [Freeswitch-users] origainate through sofia gateway<br>To: freeswitch-users@lists.freeswitch.org<br>Message-ID:<br> <87f2f3b90902040941r61d669aaie949aa7cc8578a9a@mail.gmail.com><br>Content-Type: text/plain; charset=ISO-8859-1<br><br>I'll make sure the substance of this is in the wiki and I'll look for<br>references to the deprecated way and remove those.<br>-MC<br><br>On Wed, Feb 4, 2009 at 6:09 AM, Anthony Minessale<br><anthony.minessale@gmail.com> wrote:<br>> Where did you learn how to use js this way?<br>> session.originate is being misused here and is depricated and may be<br>> removed.<br>><br>> the first arg to session.originate is either undefined or a *different*<br>> session (the a leg)<br>><br>> session1 = new Session();<br>> session1.originate(undefined,<br>>
"{ignore_early_media=true}user/1008@192.168.1.122");<br>><br>><br>session1.setVariable("effective_caller_id_number","fixed0248b");<br>><br>> //once you have session1 when you originate session2 you pass session1 as<br>> the arg<br>> // the effective_caller_id is taken from session1<br>><br>> session2 = new Session();<br>> session2.originate(session1,<br>"sofia/gateway/halonet/0225490317");<br>><br>> Anyway this whole code is depricated in favor of this:<br>><br>> session1 = new<br>Session("{ignore_early_media=true}user/1008@192.168.1.122");<br>> if (session1.ready()) {<br>> <br>session1.setVariable("effective_caller_id_number","fixed0248b");<br>> session2 = new Session("sofia/gateway/halonet/0225490317",<br>session1);<br>> }<br>><br>> and could be further refactored down to this:<br>><br>> session1 = new<br>Session("{ignore_early_media=true}user/1008@192.168.1.122");<br>> if
(session1.ready()) {<br>> <br>session1.setVariable("effective_caller_id_number","fixed0248b");<br>> session1.execute("bridge",<br>"sofia/gateway/halonet/0225490317");<br>> }<br>><br>> or down to this one line of code that will setup the call detached from<br>the<br>> script and exit.<br>><br>> var result = apiExecute("originate",<br>><br>"{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008@192.168.1.122<br>> bridge:sofia/gateway/halonet/0225490317 inline");<br>><br>> if you dont care about the result and want to exit even before the call is<br>> completed.<br>><br>> var result = apiExecute("bgapi", "originate<br>><br>{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008@192.168.1.122<br>> bridge:sofia/gateway/halonet/0225490317 inline");<br>><br>><br>><br>> On Wed, Feb 4, 2009 at
2:51 AM, Jacek Sokulski<br><jsokulski@dotsystems.pl><br>> wrote:<br>>><br>>> We have tried setting both effective_caller_id_number and<br>>> origination_caller_id_number:<br>>><br>>><br>>><br>session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15);<br>>> but the problem still exists. The solution we have found for the case<br>>> when we originate two calls, local and external, is as follow:<br>>><br>>> session1 = new Session();<br>>><br>session1.originate(session1,"user/1003@192.168.1.122",15);//local<br>>> if(session1.ready()) {<br>>> session1.execute("execute_extension","00930691688627<br>XML<br>>> default");//external<br>>> }<br>>><br>>> so the external call goes through the dialplan.<br>>> It does not work if both calls are external. One possible solution<br>could<br>>>
be<br>>> to pass the originating call through dialplan (loopback?) but we have<br>not<br>>> managed<br>>> to figure out how to do it.<br>>><br>>> Thanks<br>>> Jacek<br>>><br>>> Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner pisze:<br>>> > Oops! Well, fortunately I don't use that voip provider<br>anymore (nor the<br>>> > script).<br>>> ><br>>> > Thanks Brian.<br>>> ><br>>> > Nicolas<br>>> ><br>>> > On Tue, Feb 3, 2009 at 2:25 PM, Brian West<br><brian@freeswitch.org> wrote:<br>>> > > YOU should NEVER use this method or call setCallerData at<br>all you<br>>> > > should use the correct methods to override the callerid.<br>>> > ><br>>> > > If its a B-Leg born from an A-Leg you use these on the on<br>the A-Leg:<br>>> > ><br>>> > ><br>>> >
><br>http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name<br>>> > ><br>>> > ><br>http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number<br>>> > ><br>>> > > If you're originating you use this:<br>>> > ><br>>> > ><br>>> > ><br>http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name<br>>> > ><br>>> > ><br>http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number<br>>> > ><br>>> > > /b<br>>> ><br>>> > _______________________________________________<br>>> > Freeswitch-users mailing list<br>>> > Freeswitch-users@lists.freeswitch.org<br>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>>>
><br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>>> > http://www.freeswitch.org<br>>><br>>><br>>> _______________________________________________<br>>> Freeswitch-users mailing list<br>>> Freeswitch-users@lists.freeswitch.org<br>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>>><br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>>> http://www.freeswitch.org<br>><br>><br>><br>><br>> --<br>> Anthony Minessale II<br>><br>> FreeSWITCH http://www.freeswitch.org/<br>> ClueCon http://www.cluecon.com/<br>><br>> AIM: anthm<br>> MSN:anthony_minessale@hotmail.com<br>> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<br>> IRC: irc.freenode.net #freeswitch<br>><br>> FreeSWITCH Developer Conference<br>> sip:888@conference.freeswitch.org<br>>
iax:guest@conference.freeswitch.org/888<br>> googletalk:conf+888@conference.freeswitch.org<br>> pstn:213-799-1400<br>><br>> _______________________________________________<br>> Freeswitch-users mailing list<br>> Freeswitch-users@lists.freeswitch.org<br>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> http://www.freeswitch.org<br>><br>><br><br><br><br>------------------------------<br><br>Message: 3<br>Date: Wed, 4 Feb 2009 09:43:10 -0800<br>From: "Daniel Liang" <Daniell@airg.com><br>Subject: [Freeswitch-users] Recording background music and voice is<br> out of sync<br>To: <freeswitch-users@lists.freeswitch.org><br>Message-ID:<br> <0B02E756F603CC409EB553879B090CC80A23EBB5@HPEXCHVS01.exchange.airg><br>Content-Type: text/plain; charset="us-ascii"<br><br> What I did was the following:<br> <br>First, I sent the
playback command:<br> <br>call-command: execute<br>execute-app-name: playback<br>execute-app-arg: <filename><br> <br>Then I send uuid_record (Sorry, it was not Record command):<br> <br>api uuid_record <uuid> start <filename> 120<br> <br>I also tried replacing the playback command with:<br>api uuid_displace <uuid> start <filename> 0 mux<br> <br>But the end results are the same. The recorded user's voice is about 0.5<br>second behind the expected result.<br> <br>Thanks,<br>Daniel<br> <br> <br>-----Original Message-----<br>From: freeswitch-users-bounces@lists.freeswitch.org<br>[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of<br>Brian West<br>Sent: February 3, 2009 6:36 PM<br>To: freeswitch-users@lists.freeswitch.org<br>Subject: Re: [Freeswitch-users] Recording background music and voice is<br>outof sync<br> <br>Can you show us an example of how you're doing this? Playback and<br>Record aren't async so
you'll need to show us how you're doing<br>this.<br> <br>Also don't hijack threads you hit replay on the one "Re: [Freeswitch-<br>users] FreeSwitch setup as a "Dumb" SBC" as the subject, deleted<br>the<br>subject and started a new body. That hijacks the thread and that can<br>cause your problem to go ignored in some cases if people aren't<br>interested in the thread topic depending on how their reader threads the<br>emails.<br> <br>Please click new message and type freeswitch- users@lists.freeswitch.org<br>in and then input your subject and body to start a new thread.<br> <br>Thanks,<br>Brian West<br>FreeSWITCH.org<br> <br><br>On Feb 3, 2009, at 8:21 PM, Daniel Liang wrote:<br> <br>> Hi,<br>><br>> I was trying to record a background music with a user's voice at the <br>> same time. I did a playback and started recording. But the recorded <br>> user's voice and the background music is about 0.5 second out of sync.<br><br>> I also tried
to use uuid_displace instead of playback, but I got the <br>> same result.<br> <br><br>_______________________________________________<br>Freeswitch-users mailing list<br>Freeswitch-users@lists.freeswitch.org<br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>http://www.freeswitch.org <http://www.freeswitch.org/> <br><br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br>URL:<br>http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/2d1124e2/attachment-0001.html<br><br><br>------------------------------<br><br>Message: 4<br>Date: Wed, 4 Feb 2009 23:26:14 +0530<br>From: "Gopalakrishnan A.N" <saigop@gmail.com><br>Subject: Re: [Freeswitch-users] Q931 decoding Update<br>To: freeswitch-users@lists.freeswitch.org<br>Message-ID:<br>
<2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com><br>Content-Type: text/plain; charset="iso-8859-1"<br><br>Hi,<br> Its a awesome. Can the packet capturing be done with event socket?<br><br>-- <br>Thank you with regards,<br>Gopal,<br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br>URL:<br>http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/74f6434a/attachment-0001.html<br><br><br>------------------------------<br><br>Message: 5<br>Date: Wed, 04 Feb 2009 09:59:48 -0800<br>From: Chav Paskov <chavpaskov@shaw.ca><br>Subject: [Freeswitch-users] mod_limit<br>To: freeswitch-users@lists.freeswitch.org<br>Message-ID: <4989D794.1010805@shaw.ca><br>Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br><br>Hi ,<br>is it possible to use mod_limit in case if the end point is not <br>registered / gateway for
example/.<br>Regards<br>Chav<br><br><br><br>------------------------------<br><br>Message: 6<br>Date: Wed, 4 Feb 2009 10:06:52 -0800<br>From: Michael Collins <msc@freeswitch.org><br>Subject: Re: [Freeswitch-users] mod_limit<br>To: freeswitch-users@lists.freeswitch.org<br>Message-ID:<br> <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0@mail.gmail.com><br>Content-Type: text/plain; charset=ISO-8859-1<br><br>On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov <chavpaskov@shaw.ca> wrote:<br>> Hi ,<br>> is it possible to use mod_limit in case if the end point is not<br>> registered / gateway for example/.<br><br>Could you add some detail to this question? What are you trying to do?<br>(mod_limit may or may not work, but there might be another solution<br>which is why I am asking.)<br><br>-MC<br><br>> Regards<br>> Chav<br>><br>> _______________________________________________<br>> Freeswitch-users mailing list<br>>
Freeswitch-users@lists.freeswitch.org<br>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> http://www.freeswitch.org<br>><br><br><br><br>------------------------------<br><br>Message: 7<br>Date: Wed, 04 Feb 2009 10:54:56 -0800<br>From: Chav Paskov <chavpaskov@shaw.ca><br>Subject: Re: [Freeswitch-users] mod_limit<br>To: freeswitch-users@lists.freeswitch.org<br>Message-ID: <4989E480.1080105@shaw.ca><br>Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br><br>Michael Collins wrote:<br>> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov <chavpaskov@shaw.ca><br>wrote:<br>> <br>>> Hi ,<br>>> is it possible to use mod_limit in case if the end point is not<br>>> registered / gateway for example/.<br>>> <br>><br>> Could you add some detail to this question? What are you trying to do?<br>>
(mod_limit may or may not work, but there might be another solution<br>> which is why I am asking.)<br>><br>> -MC<br>><br>> <br>>> Regards<br>>> Chav<br>>><br>>> _______________________________________________<br>>> Freeswitch-users mailing list<br>>> Freeswitch-users@lists.freeswitch.org<br>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>>><br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>>> http://www.freeswitch.org<br>>><br>>> <br>><br>> _______________________________________________<br>> Freeswitch-users mailing list<br>> Freeswitch-users@lists.freeswitch.org<br>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> http://www.freeswitch.org<br>><br>> <br>i have few gateways under my ACL that
are allowed to send calls to FS, <br>but i want to be able to enforce "capacity" policy on the traffic <br>coming from any one of them depending on total termination capacity on <br>my termination end.<br>Let say GW 1 has to be limited to make 10 simultaneous calls while GW2 <br>could make up to 30 and so on.<br>Regards<br>Chav<br><br><br><br>------------------------------<br><br>Message: 8<br>Date: Wed, 4 Feb 2009 11:05:09 -0800<br>From: Michael Collins <msc@freeswitch.org><br>Subject: Re: [Freeswitch-users] mod_limit<br>To: freeswitch-users@lists.freeswitch.org<br>Message-ID:<br> <87f2f3b90902041105l50f51f08t230bab8d69eefb4e@mail.gmail.com><br>Content-Type: text/plain; charset=ISO-8859-1<br><br>On Wed, Feb 4, 2009 at 10:54 AM, Chav Paskov <chavpaskov@shaw.ca> wrote:<br>> Michael Collins wrote:<br>>> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov <chavpaskov@shaw.ca><br>wrote:<br>>><br>>>> Hi
,<br>>>> is it possible to use mod_limit in case if the end point is not<br>>>> registered / gateway for example/.<br>>>><br>>><br>>> Could you add some detail to this question? What are you trying to do?<br>>> (mod_limit may or may not work, but there might be another solution<br>>> which is why I am asking.)<br>>><br>>> -MC<br>>><br>>><br>>>> Regards<br>>>> Chav<br>>>><br>>>> _______________________________________________<br>>>> Freeswitch-users mailing list<br>>>> Freeswitch-users@lists.freeswitch.org<br>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>>>><br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>>>> http://www.freeswitch.org<br>>>><br>>>><br>>><br>>> _______________________________________________<br>>>
Freeswitch-users mailing list<br>>> Freeswitch-users@lists.freeswitch.org<br>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>>><br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>>> http://www.freeswitch.org<br>>><br>>><br>> i have few gateways under my ACL that are allowed to send calls to FS,<br>> but i want to be able to enforce "capacity" policy on the<br>traffic<br>> coming from any one of them depending on total termination capacity on<br>> my termination end.<br>> Let say GW 1 has to be limited to make 10 simultaneous calls while GW2<br>> could make up to 30 and so on.<br><br>I'm sure that this is possible. I don't personally have a way to test<br>all of this but I know that a number of our users are doing things<br>like this currently. Can you hop on to the IRC channel? #freeswitch on<br>irc.freenode.net. A lot of people there can help with
this one.<br><br>-MC (IRC: mercutioviz)<br><br>> Regards<br>> Chav<br>><br>> _______________________________________________<br>> Freeswitch-users mailing list<br>> Freeswitch-users@lists.freeswitch.org<br>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> http://www.freeswitch.org<br>><br><br><br><br>------------------------------<br><br>_______________________________________________<br>Freeswitch-users mailing list<br>Freeswitch-users@lists.freeswitch.org<br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>http://www.freeswitch.org<br><br><br>End of Freeswitch-users Digest, Vol 32, Issue 39<br>************************************************<br></pre></blockquote></td></tr></table><br>