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kerrada2003 at yahoo.com
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 Posted: Wed Feb 04, 2009 11:50 am Post subject: [Freeswitch-users] SIP Authentication |
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Hi,
I have a problem in SIP registration (authentication) with FreeSWITCH server. The SIP messages are:
Normal 0 false false false MicrosoftInternetExplorer4 <![endif]--> <![endif]--> /* Style Definitions */ table.MsoNormalTable {mso-style-name:"Table Normal"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0; mso-style-noshow:yes; mso-style-parent:""; mso-padding-alt:0in 5.4pt 0in 5.4pt; mso-para-margin:0in; mso-para-margin-bottom:.0001pt; mso-pagination:widow-orphan; font-size:10.0pt; font-family:"Times New Roman"; mso-ansi-language:#0400; mso-fareast-language:#0400; mso-bidi-language:#0400;} <![endif]-->
recv 292 bytes from udp/[209.82.10.250]:3458 at 16:35:24.758862:
------------------------------------------------------------------------
REGISTER sip:209.82.10.235 SIP/2.0
Via: SIP/2.0/UDP 209.82.10.250:1059
From: sip:1001@209.82.10.235
To: sip:1001@209.82.10.235
Contact: sip:1001@209.82.10.250:1059
Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8@192.168.10.23
CSeq: 597775306 REGISTER
Content-Length: 0
Expires: 3600
------------------------------------------------------------------------
send 582 bytes to udp/[209.82.10.250]:1059 at 16:35:24.763948:
------------------------------------------------------------------------
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.82.10.250:1059
From: sip:1001@209.82.10.235
To: <sip:1001@209.82.10.235>;tag=7yam2F01ZH3vH
Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8@192.168.10.23
CSeq: 597775306 REGISTER
User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
WWW-Authenticate: Digest realm="209.82.10.235", nonce="40b63193-85c2-4ed9-874e-c03f81be313d", algorithm=MD5, qop="auth"
Content-Length: 0
------------------------------------------------------------------------
recv 466 bytes from udp/[209.82.10.250]:3458 at 16:35:24.772834:
------------------------------------------------------------------------
REGISTER sip:209.82.10.235 SIP/2.0
Via: SIP/2.0/UDP 209.82.10.250:1059
From: sip:1001@209.82.10.235
To: sip:1001@209.82.10.235
Contact: sip:1001@209.82.10.250:1059
Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8@192.168.10.23
CSeq: 597775307 REGISTER
Content-Length: 0
Expires: 3600
Authorization: Digest username="1001",realm="209.82.10.235",nonce="40b63193-85c2-4ed9-874e-c03f81be313d",response="eebe0ea43319e82cc5f6dba5877de706",uri="sip:209.82.10.235"
------------------------------------------------------------------------
send 458 bytes to udp/[209.82.10.250]:1059 at 16:35:24.774354:
------------------------------------------------------------------------
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 209.82.10.250:1059
From: sip:1001@209.82.10.235
To: <sip:1001@209.82.10.235>;tag=873c4aH5vtSFD
Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8@192.168.10.23
CSeq: 597775307 REGISTER
User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Content-Length: 0
------------------------------------------------------------------------ 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 |
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brian at freeswitch.org
Guest
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| Posted: Wed Feb 04, 2009 11:55 am Post subject: [Freeswitch-users] SIP Authentication |
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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:
| Quote: | 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
|
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anthony.minessale at g...
Guest
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| Posted: Wed Feb 04, 2009 11:56 am Post subject: [Freeswitch-users] SIP Authentication |
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That's the reason, the missing params.
The client has a bug in it.
On Wed, Feb 4, 2009 at 10:17 AM, Ali Al-Rubaie <kerrada2003@yahoo.com (kerrada2003@yahoo.com)> wrote:
| Quote: | Hi,
I have a problem in SIP registration (authentication) with FreeSWITCH server. The SIP messages are:
recv 292 bytes from udp/[209.82.10.250]:3458 at 16:35:24.758862:
------------------------------------------------------------------------
REGISTER sip:209.82.10.235 SIP/2.0
Via: SIP/2.0/UDP 209.82.10.250:1059
From: sip:1001@209.82.10.235 ([email]sip%3A1001@209.82.10.235[/email])
To: sip:1001@209.82.10.235 ([email]sip%3A1001@209.82.10.235[/email])
Contact: sip:1001@209.82.10.250:1059
Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8@192.168.10.23 (fc383368-e3e0-4c32-b87b-c3127d0cc2d8@192.168.10.23)
CSeq: 597775306 REGISTER
Content-Length: 0
Expires: 3600
------------------------------------------------------------------------
send 582 bytes to udp/[209.82.10.250]:1059 at 16:35:24.763948:
------------------------------------------------------------------------
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.82.10.250:1059
From: sip:1001@209.82.10.235 ([email]sip%3A1001@209.82.10.235[/email])
To: <sip:1001@209.82.10.235 ([email]sip%3A1001@209.82.10.235[/email])>;tag=7yam2F01ZH3vH
Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8@192.168.10.23 (fc383368-e3e0-4c32-b87b-c3127d0cc2d8@192.168.10.23)
CSeq: 597775306 REGISTER
User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
WWW-Authenticate: Digest realm="209.82.10.235", nonce="40b63193-85c2-4ed9-874e-c03f81be313d", algorithm=MD5, qop="auth"
Content-Length: 0
------------------------------------------------------------------------
recv 466 bytes from udp/[209.82.10.250]:3458 at 16:35:24.772834:
------------------------------------------------------------------------
REGISTER sip:209.82.10.235 SIP/2.0
Via: SIP/2.0/UDP 209.82.10.250:1059
From: sip:1001@209.82.10.235 ([email]sip%3A1001@209.82.10.235[/email])
To: sip:1001@209.82.10.235 ([email]sip%3A1001@209.82.10.235[/email])
Contact: sip:1001@209.82.10.250:1059
Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8@192.168.10.23 (fc383368-e3e0-4c32-b87b-c3127d0cc2d8@192.168.10.23)
CSeq: 597775307 REGISTER
Content-Length: 0
Expires: 3600
Authorization: Digest username="1001",realm="209.82.10.235",nonce="40b63193-85c2-4ed9-874e-c03f81be313d",response="eebe0ea43319e82cc5f6dba5877de706",uri="sip:209.82.10.235"
------------------------------------------------------------------------
send 458 bytes to udp/[209.82.10.250]:1059 at 16:35:24.774354:
------------------------------------------------------------------------
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 209.82.10.250:1059
From: sip:1001@209.82.10.235 ([email]sip%3A1001@209.82.10.235[/email])
To: <sip:1001@209.82.10.235 ([email]sip%3A1001@209.82.10.235[/email])>;tag=873c4aH5vtSFD
Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8@192.168.10.23 (fc383368-e3e0-4c32-b87b-c3127d0cc2d8@192.168.10.23)
CSeq: 597775307 REGISTER
User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Content-Length: 0
------------------------------------------------------------------------ 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
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@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@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400 |
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kerrada2003 at yahoo.com
Guest
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| Posted: Thu Feb 05, 2009 11:38 am Post subject: [Freeswitch-users] SIP Authentication |
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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@lists.freeswitch.org <freeswitch-users-request@lists.freeswitch.org> wrote:
| Quote: | From: freeswitch-users-request@lists.freeswitch.org <freeswitch-users-request@lists.freeswitch.org>
Subject: Freeswitch-users Digest, Vol 32, Issue 39
To: freeswitch-users@lists.freeswitch.org
Date: Wednesday, February 4, 2009, 2:05 PM
| Quote: | Send Freeswitch-users mailing list submissions to freeswitch-users@lists.freeswitch.orgTo subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersor, via email, send a message with subject or body 'help' to freeswitch-users-request@lists.freeswitch.orgYou can reach the person managing the list at freeswitch-users-owner@lists.freeswitch.orgWhen replying, please edit your Subject line so it is more specificthan "Re: Contents of Freeswitch-users digest..."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: 1Date: Wed, 4 Feb 2009 10:52:45 -0600From: Brian West <brian@freeswitch.org>Subject: Re: [Freeswitch-users] SIP AuthenticationTo: freeswitch-users@lists.freeswitch.orgMessage-ID: <7DAC91F6-2FD3-464D-AA81-321EBCADC8C0@freeswitch.org>Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yesWhat client is this? I also notice we receive port 3458 and reply to port 1059.../bOn 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 ifthis is the > reason, however how this problem can be solved?>> Thanks,>> Ali------------------------------Message:
2Date: Wed, 4 Feb 2009 09:41:07 -0800From: Michael Collins <msc@freeswitch.org>Subject: Re: [Freeswitch-users] origainate through sofia gatewayTo: freeswitch-users@lists.freeswitch.orgMessage-ID: <87f2f3b90902040941r61d669aaie949aa7cc8578a9a@mail.gmail.com>Content-Type: text/plain; charset=ISO-8859-1I'll make sure the substance of this is in the wiki and I'll look forreferences to the deprecated way and remove those.-MCOn Wed, Feb 4, 2009 at 6:09 AM, Anthony Minessale<anthony.minessale@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@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 = newSession("{ignore_early_media=true}user/1008@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 = newSession("{ignore_early_media=true}user/1008@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 fromthe> script and exit.>> var result = apiExecute("originate",>"{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008@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@192.168.1.122> bridge:sofia/gateway/halonet/0225490317 inline");>>>> On Wed, Feb 4, 2009 at
2:51 AM, Jacek Sokulski<jsokulski@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@192.168.1.122",15);//local>> if(session1.ready()) {>> session1.execute("execute_extension","00930691688627XML>> default");//external>> }>>>> so the external call goes through the dialplan.>> It does not work if both calls are external. One possible solutioncould>>
be>> to pass the originating call through dialplan (loopback?) but we havenot>> 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 provideranymore (nor the>> > script).>> >>> > Thanks Brian.>> >>> > Nicolas>> >>> > On Tue, Feb 3, 2009 at 2:25 PM, Brian West<brian@freeswitch.org> wrote:>> > > YOU should NEVER use this method or call setCallerData atall 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 onthe A-Leg:>> > >>> > >>> >
iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org> pstn:213-799-1400>> _______________________________________________> Freeswitch-users mailing list> Freeswitch-users@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: 3Date: Wed, 4 Feb 2009 09:43:10 -0800From: "Daniel Liang" <Daniell@airg.com>Subject: [Freeswitch-users] Recording background music and voice is out of syncTo: <freeswitch-users@lists.freeswitch.org>Message-ID: <0B02E756F603CC409EB553879B090CC80A23EBB5@HPEXCHVS01.exchange.airg>Content-Type: text/plain; charset="us-ascii" What I did was the following: First, I sent the
playback command: call-command: executeexecute-app-name: playbackexecute-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 0.5second behind the expected result. Thanks,Daniel -----Original Message-----From: freeswitch-users-bounces@lists.freeswitch.org[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf OfBrian WestSent: February 3, 2009 6:36 PMTo: freeswitch-users@lists.freeswitch.orgSubject: Re: [Freeswitch-users] Recording background music and voice isoutof sync Can you show us an example of how you're doing this? Playback andRecord aren't async so
you'll need to show us how you're doingthis. Also don't hijack threads you hit replay on the one "Re: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC" as the subject, deletedthesubject and started a new body. That hijacks the thread and that cancause your problem to go ignored in some cases if people aren'tinterested in the thread topic depending on how their reader threads theemails. Please click new message and type freeswitch- users@lists.freeswitch.orgin and then input your subject and body to start a new thread. Thanks,Brian WestFreeSWITCH.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 listFreeswitch-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org <http://www.freeswitch.org/> -------------- next part --------------An HTML attachment was scrubbed...URL:http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/2d1124e2/attachment-0001.html------------------------------Message: 4Date: Wed, 4 Feb 2009 23:26:14 +0530From: "Gopalakrishnan A.N" <saigop@gmail.com>Subject: Re: [Freeswitch-users] Q931 decoding UpdateTo: freeswitch-users@lists.freeswitch.orgMessage-ID:
<2ea4d47e0902040956v75c5472foa4649c50b7340484@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,-------------- next part --------------An HTML attachment was scrubbed...URL:http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/74f6434a/attachment-0001.html------------------------------Message: 5Date: Wed, 04 Feb 2009 09:59:48 -0800From: Chav Paskov <chavpaskov@shaw.ca>Subject: [Freeswitch-users] mod_limitTo: freeswitch-users@lists.freeswitch.orgMessage-ID: <4989D794.1010805@shaw.ca>Content-Type: text/plain; charset=ISO-8859-1; format=flowedHi ,is it possible to use mod_limit in case if the end point is not registered / gateway for
example/.RegardsChav------------------------------Message: 6Date: Wed, 4 Feb 2009 10:06:52 -0800From: Michael Collins <msc@freeswitch.org>Subject: Re: [Freeswitch-users] mod_limitTo: freeswitch-users@lists.freeswitch.orgMessage-ID: <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0@mail.gmail.com>Content-Type: text/plain; charset=ISO-8859-1On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov <chavpaskov@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 solutionwhich is why I am asking.)-MC> Regards> Chav>> _______________________________________________> Freeswitch-users mailing list>
Freeswitch-users@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: 7Date: Wed, 04 Feb 2009 10:54:56 -0800From: Chav Paskov <chavpaskov@shaw.ca>Subject: Re: [Freeswitch-users] mod_limitTo: freeswitch-users@lists.freeswitch.orgMessage-ID: <4989E480.1080105@shaw.ca>Content-Type: text/plain; charset=ISO-8859-1; format=flowedMichael Collins wrote:> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov <chavpaskov@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@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@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.RegardsChav------------------------------Message: 8Date: Wed, 4 Feb 2009 11:05:09 -0800From: Michael Collins <msc@freeswitch.org>Subject: Re: [Freeswitch-users] mod_limitTo: freeswitch-users@lists.freeswitch.orgMessage-ID: <87f2f3b90902041105l50f51f08t230bab8d69eefb4e@mail.gmail.com>Content-Type: text/plain; charset=ISO-8859-1On Wed, Feb 4, 2009 at 10:54 AM, Chav Paskov <chavpaskov@shaw.ca> wrote:> Michael Collins wrote:>> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov <chavpaskov@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@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@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 thetraffic> 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 testall of this but I know that a number of our users are doing thingslike this currently. Can you hop on to the IRC channel? #freeswitch onirc.freenode.net. A lot of people there can help with
this one.-MC (IRC: mercutioviz)> Regards> Chav>> _______________________________________________> Freeswitch-users mailing list> Freeswitch-users@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 listFreeswitch-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.orgEnd of Freeswitch-users Digest, Vol 32, Issue 39************************************************ |
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anthony.minessale at g...
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| Posted: Thu Feb 05, 2009 11:48 am Post subject: [Freeswitch-users] SIP Authentication |
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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@yahoo.com (kerrada2003@yahoo.com)> wrote:
| Quote: |
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@lists.freeswitch.org (freeswitch-users-request@lists.freeswitch.org) <freeswitch-users-request@lists.freeswitch.org (freeswitch-users-request@lists.freeswitch.org)> wrote:
| Quote: | From: freeswitch-users-request@lists.freeswitch.org (freeswitch-users-request@lists.freeswitch.org) <freeswitch-users-request@lists.freeswitch.org (freeswitch-users-request@lists.freeswitch.org)>
Subject: Freeswitch-users Digest, Vol 32, Issue 39
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Date: Wednesday, February 4, 2009, 2:05 PM
| Quote: | Send Freeswitch-users mailing list submissions to freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)To subscribe or unsubscribe via the World Wide Web, visit
http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersor, via email, send a message with subject or body 'help' to
freeswitch-users-request@lists.freeswitch.org (freeswitch-users-request@lists.freeswitch.org)You can reach the person managing the list at freeswitch-users-owner@lists.freeswitch.org (freeswitch-users-owner@lists.freeswitch.org)
When replying, please edit your Subject line so it is more specificthan "Re: Contents of Freeswitch-users digest..."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: 1Date: Wed, 4 Feb 2009 10:52:45 -0600From: Brian West <brian@freeswitch.org (brian@freeswitch.org)>
Subject: Re: [Freeswitch-users] SIP AuthenticationTo: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)Message-ID: <7DAC91F6-2FD3-464D-AA81-321EBCADC8C0@freeswitch.org (7DAC91F6-2FD3-464D-AA81-321EBCADC8C0@freeswitch.org)>
Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yesWhat client is this? I also notice we receive port 3458 and reply to port 1059.../bOn Feb 4, 2009, at 10:17 AM, Ali Al-Rubaie wrote:
| Quote: | What I have noted is that the client does not send the values for > "cnonce" and "nc" in the response. I'm not sure ifthis is the > reason, however how this problem can be solved?
| Quote: | Thanks,>> Ali------------------------------Message:
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2Date: Wed, 4 Feb 2009 09:41:07 -0800From: Michael Collins <msc@freeswitch.org (msc@freeswitch.org)>Subject: Re: [Freeswitch-users] origainate through sofia gateway
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)Message-ID: <87f2f3b90902040941r61d669aaie949aa7cc8578a9a@mail.gmail.com (87f2f3b90902040941r61d669aaie949aa7cc8578a9a@mail.gmail.com)>
Content-Type: text/plain; charset=ISO-8859-1I'll make sure the substance of this is in the wiki and I'll look forreferences to the deprecated way and remove those.-MCOn Wed, Feb 4, 2009 at 6:09 AM, Anthony Minessale
<anthony.minessale@gmail.com (anthony.minessale@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
| Quote: | removed.>> the first arg to session.originate is either undefined or a *different*> session (the a leg)>> session1 = new Session();> session1.originate(undefined,>
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"{ignore_early_media=true}user/1008@192.168.1.122 (1008@192.168.1.122)");>>session1.setVariable("effective_caller_id_number","fixed0248b");
| Quote: | | Quote: | //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,
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"sofia/gateway/halonet/0225490317");>> Anyway this whole code is depricated in favor of this:>> session1 = newSession("{ignore_early_media=true}user/1008@192.168.1.122 (1008@192.168.1.122)");
| Quote: | 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 = newSession("{ignore_early_media=true}user/1008@192.168.1.122 (1008@192.168.1.122)");
if
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(session1.ready()) {> session1.setVariable("effective_caller_id_number","fixed0248b");> session1.execute("bridge","sofia/gateway/halonet/0225490317");
| Quote: | }>> or down to this one line of code that will setup the call detached fromthe> script and exit.>> var result = apiExecute("originate",>"{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008@192.168.1.122 (1008@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@192.168.1.122 (1008@192.168.1.122)> bridge:sofia/gateway/halonet/0225490317 inline");
| Quote: | | Quote: | | Quote: | On Wed, Feb 4, 2009 at
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2:51 AM, Jacek Sokulski<jsokulski@dotsystems.pl>> wrote:>>>> We have tried setting both effective_caller_id_number and>> origination_caller_id_number:>>>>
| Quote: | | Quote: | 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:
| Quote: | | Quote: | session1 = new Session();>>session1.originate(session1,"user/1003@192.168.1.122 (1003@192.168.1.122)",15);//local>> if(session1.ready()) {
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session1.execute("execute_extension","00930691688627XML>> default");//external>> }>>>> so the external call goes through the dialplan.>> It does not work if both calls are external. One possible solution
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could>>
be>> to pass the originating call through dialplan (loopback?) but we havenot>> managed>> to figure out how to do it.>>>> Thanks>> Jacek>>>> Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner pisze:
| Quote: | | Quote: | | Quote: | Oops! Well, fortunately I don't use that voip provideranymore (nor the>> > script).>> >>> > Thanks Brian.>> >>> > Nicolas>> >
On Tue, Feb 3, 2009 at 2:25 PM, Brian West<brian@freeswitch.org (brian@freeswitch.org)> wrote:>> > > YOU should NEVER use this method or call setCallerData at
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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 onthe A-Leg:>> > >
http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number>> > >>> > > If you're originating you use this:
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
Subject: [Freeswitch-users] Recording background music and voice is out of syncTo: <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>Message-ID:
<0B02E756F603CC409EB553879B090CC80A23EBB5@HPEXCHVS01.exchange.airg>Content-Type: text/plain; charset="us-ascii" What I did was the following: First, I sent the
playback command: call-command: executeexecute-app-name: playbackexecute-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 0.5second behind the expected result.
Thanks,Daniel -----Original Message-----From: freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)[mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] On Behalf Of
Brian WestSent: February 3, 2009 6:36 PMTo: freeswitch-users@lists.freeswitch.org (freeswitch-users@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 andRecord aren't async so
you'll need to show us how you're doingthis. Also don't hijack threads you hit replay on the one "Re: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC" as the subject, deleted
thesubject and started a new body. That hijacks the thread and that cancause your problem to go ignored in some cases if people aren'tinterested in the thread topic depending on how their reader threads the
emails. Please click new message and type freeswitch- users@lists.freeswitch.org (users@lists.freeswitch.org)in and then input your subject and body to start a new thread.
Thanks,Brian WestFreeSWITCH.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
| Quote: | user's voice and the background music is about 0.5 second out of sync.> I also tried
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to use uuid_displace instead of playback, but I got the > same result. _______________________________________________Freeswitch-users mailing listFreeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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------------------------------Message: 4Date: Wed, 4 Feb 2009 23:26:14 +0530From: "Gopalakrishnan A.N" <saigop@gmail.com (saigop@gmail.com)>Subject: Re: [Freeswitch-users] Q931 decoding Update
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)Message-ID:
<2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com (2ea4d47e0902040956v75c5472foa4649c50b7340484@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,-------------- next part --------------An HTML attachment was scrubbed...URL:
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------------------------------Message: 5Date: Wed, 04 Feb 2009 09:59:48 -0800From: Chav Paskov <chavpaskov@shaw.ca (chavpaskov@shaw.ca)>Subject: [Freeswitch-users] mod_limit
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)Message-ID: <4989D794.1010805@shaw.ca (4989D794.1010805@shaw.ca)>
Content-Type: text/plain; charset=ISO-8859-1; format=flowedHi ,is it possible to use mod_limit in case if the end point is not registered / gateway for
example/.RegardsChav------------------------------Message: 6Date: Wed, 4 Feb 2009 10:06:52 -0800From: Michael Collins <msc@freeswitch.org (msc@freeswitch.org)>
Subject: Re: [Freeswitch-users] mod_limitTo: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)Message-ID: <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0@mail.gmail.com (87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0@mail.gmail.com)>
Content-Type: text/plain; charset=ISO-8859-1On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov <chavpaskov@shaw.ca (chavpaskov@shaw.ca)> wrote:> Hi ,> is it possible to use mod_limit in case if the end point is not
| Quote: | 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 solutionwhich is why I am asking.)
|
-MC> Regards> Chav>> _______________________________________________> Freeswitch-users mailing list>
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Subject: Re: [Freeswitch-users] mod_limitTo: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)Message-ID: <4989E480.1080105@shaw.ca (4989E480.1080105@shaw.ca)>
Content-Type: text/plain; charset=ISO-8859-1; format=flowedMichael Collins wrote:> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov <chavpaskov@shaw.ca (chavpaskov@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>>>> _______________________________________________
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.RegardsChav------------------------------Message: 8Date: Wed, 4 Feb 2009 11:05:09 -0800From: Michael Collins <msc@freeswitch.org (msc@freeswitch.org)>
Subject: Re: [Freeswitch-users] mod_limitTo: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)Message-ID: <87f2f3b90902041105l50f51f08t230bab8d69eefb4e@mail.gmail.com (87f2f3b90902041105l50f51f08t230bab8d69eefb4e@mail.gmail.com)>
Content-Type: text/plain; charset=ISO-8859-1On Wed, Feb 4, 2009 at 10:54 AM, Chav Paskov <chavpaskov@shaw.ca (chavpaskov@shaw.ca)> wrote:> Michael Collins wrote:
| Quote: | | Quote: | On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov <chavpaskov@shaw.ca (chavpaskov@shaw.ca)>wrote:>>>>> Hi
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,>>> 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?
| Quote: | | Quote: | (mod_limit may or may not work, but there might be another solution>> which is why I am asking.)>>>> -MC>>>>>>> Regards>>> Chav
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Freeswitch-users mailing list>> Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
| Quote: | 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
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all of this but I know that a number of our users are doing thingslike this currently. Can you hop on to the IRC channel? #freeswitch onirc.freenode.net. A lot of people there can help with
this one.-MC (IRC: mercutioviz)> Regards> Chav>> _______________________________________________> Freeswitch-users mailing list> Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
Freeswitch-users mailing listFreeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
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kerrada2003 at yahoo.com
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| Posted: Mon Feb 09, 2009 10:12 am Post subject: [Freeswitch-users] SIP Authentication |
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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,
| Quote: | | Quote: | Message: 2Date: Thu, 5 Feb 2009 10:46:54 -0600From: Anthony Minessale <anthony.minessale@gmail.com>Subject: Re: [Freeswitch-users] SIP AuthenticationTo: freeswitch-users@lists.freeswitch.orgMessage-ID: <191c3a030902050846o60047c30pa2890707eae386d6@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 exercisetheoption which we have opted to
always do =DThere is no way in the code to disable sending it because we prefer the moresecure version of SIP auth.So again it's a bug in the client for not following the protocol. It wouldbe considered a feature in FreeSWITCH to support limping for the sake ofthis broken client and we currently do not have any plans for implementingthis feature.On Thu, Feb 5, 2009 at 10:34 AM, Ali Al-Rubaie<kerrada2003@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 respondwith the> "cnonce" parameter. The parameter "qop" is optional inDigest Auth. So the> question here is that, can we configure FreeSWITCH so that it will notsend> "qop" in the challenge?>> Thanks!>> --- On *Wed,
2/4/09, freeswitch-users-request@lists.freeswitch.org <> freeswitch-users-request@lists.freeswitch.org>* wrote:>> From: freeswitch-users-request@lists.freeswitch.org <> freeswitch-users-request@lists.freeswitch.org>> Subject: Freeswitch-users Digest, Vol 32, Issue 39> To: freeswitch-users@lists.freeswitch.org> Date: Wednesday, February 4, 2009, 2:05 PM>> Send Freeswitch-users mailing list submissions to> freeswitch-users@lists.freeswitch.org>> To subscribe or unsubscribe via the World Wide Web, visit> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> or, via email, send a message with subject or body 'help' to> freeswitch-users-request@lists.freeswitch.org>> You can reach the person managing the list at> freeswitch-users-owner@lists.freeswitch.org>> When replying, please edit your
Subject line so it is more specific> than "Re: Contents of Freeswitch-users digest...">>> 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@freeswitch.org>> Subject: Re: [Freeswitch-users] SIP Authentication> To: freeswitch-users@lists.freeswitch.org> Message-ID:
<7DAC91F6-2FD3-464D-AA81-321EBCADC8C0@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 notsure 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@freeswitch.org>> Subject: Re: [Freeswitch-users] origainate through sofia gateway> To: freeswitch-users@lists.freeswitch.org>
Message-ID:> <87f2f3b90902040941r61d669aaie949aa7cc8578a9a@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 lookfor> references to the deprecated way and remove those.> -MC>> On Wed, Feb 4, 2009 at 6:09 AM, Anthony Minessale> <anthony.minessale@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@192.168.1.122");> >>
| Quote: | | Quote: | session1.setVariable("effective_caller_id_number","fixed0248b");> >> > //once you have session1 when you originate session2 you passsession1 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@192.168.1.122");> > if (session1.ready()) {> >>session1.setVariable("effective_caller_id_number","fixed0248b");> > session2 = newSession("sofia/gateway/halonet/0225490317",> session1);> > }> >> > and could be further refactored down to this:> >> > session1 = new>
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Session("{ignore_early_media=true}user/1008@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 detachedfrom> the> > script and exit.> >> > var result = apiExecute("originate",> >>"{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008@192.168.1.122> > bridge:sofia/gateway/halonet/0225490317 inline");> >> > if you dont care about the result and want to exit even before thecall is> > completed.> >> > var result = apiExecute("bgapi", "originate>
| Quote: | | Quote: | {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008@192.168.1.122> > bridge:sofia/gateway/halonet/0225490317 inline");> >> >> >> > On Wed, Feb 4, 2009 at> 2:51 AM, Jacek Sokulski> <jsokulski@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 thecase> >> when we originate two calls, local and external, is as follow:> >>> >> session1 = new Session();>
| Quote: | session1.originate(session1,"user/1003@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 possiblesolution> could> >>> be> >> to pass the originating call through dialplan (loopback?) but wehave> not> >> managed> >> to figure out how to do it.> >>> >> Thanks> >> Jacek> >>> >> Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brennerpisze:> >> > Oops! Well, fortunately I don't use that voip provider> anymore (nor the> >> > script).>
| Quote: | | Quote: | | Quote: | | Quote: | Thanks Brian.> >> >> >> > Nicolas> >> >> >> > On Tue, Feb 3, 2009 at 2:25 PM, Brian West> <brian@freeswitch.org> wrote:> >> > > YOU should NEVER use this method or call setCallerDataat> all you> >> > > should use the correct methods to override thecallerid.> >> > >> >> > > If its a B-Leg born from an A-Leg you use these on theon> 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
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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@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@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@hotmail.com<MSN%3Aanthony_minessale@hotmail.com>> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com>> > IRC: irc.freenode.net #freeswitch> >> > FreeSWITCH Developer Conference> > sip:888@conference.freeswitch.org<sip%3A888@conference.freeswitch.org>> >>
iax:guest@conference.freeswitch.org/888> > googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org>> > pstn:213-799-1400> >> > _______________________________________________> > Freeswitch-users mailing list> > Freeswitch-users@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@airg.com>> Subject: [Freeswitch-users] Recording background music and voice is> out of sync> To: <freeswitch-users@lists.freeswitch.org>> Message-ID:>
<0B02E756F603CC409EB553879B090CC80A23EBB5@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 about0.5> second behind the expected result.>> Thanks,> Daniel>>> -----Original Message-----> From: freeswitch-users-bounces@lists.freeswitch.org>
[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of> Brian West> Sent: February 3, 2009 6:36 PM> To: freeswitch-users@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@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 atthe> > 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 ofsync.>> > I also tried> to use uuid_displace instead of playback, but I got the> > same result.>>> _______________________________________________> Freeswitch-users mailing list> Freeswitch-users@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/>>> -------------- next part --------------> An HTML attachment was scrubbed...> URL:>http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/2d1124e2/attachment-0001.html>>> ------------------------------>> Message: 4> Date: Wed, 4 Feb 2009 23:26:14 +0530> From: "Gopalakrishnan A.N" <saigop@gmail.com>> Subject: Re: [Freeswitch-users] Q931 decoding Update> To: freeswitch-users@lists.freeswitch.org> Message-ID:>> <2ea4d47e0902040956v75c5472foa4649c50b7340484@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,> -------------- next part
--------------> An HTML attachment was scrubbed...> URL:>http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/74f6434a/attachment-0001.html>>> ------------------------------>> Message: 5> Date: Wed, 04 Feb 2009 09:59:48 -0800> From: Chav Paskov <chavpaskov@shaw.ca>> Subject: [Freeswitch-users] mod_limit> To: freeswitch-users@lists.freeswitch.org> Message-ID: <4989D794.1010805@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@freeswitch.org>>
Subject: Re: [Freeswitch-users] mod_limit> To: freeswitch-users@lists.freeswitch.org> Message-ID:> <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0@mail.gmail.com>> Content-Type: text/plain; charset=ISO-8859-1>> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov <chavpaskov@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@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@shaw.ca>> Subject: Re: [Freeswitch-users] mod_limit> To: freeswitch-users@lists.freeswitch.org> Message-ID: <4989E480.1080105@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@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 todo?> >> (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@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@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 thetraffic> 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@freeswitch.org>> Subject: Re: [Freeswitch-users] mod_limit> To: freeswitch-users@lists.freeswitch.org> Message-ID:>
<87f2f3b90902041105l50f51f08t230bab8d69eefb4e@mail.gmail.com>> Content-Type: text/plain; charset=ISO-8859-1>> On Wed, Feb 4, 2009 at 10:54 AM, Chav Paskov <chavpaskov@shaw.ca>wrote:> > Michael Collins wrote:> >> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov<chavpaskov@shaw.ca>> wrote:> >>> >>> Hi> ,> >>> is it possible to use mod_limit in case if the end point isnot> >>> registered / gateway for example/.> >>>> >>> >> Could you add some detail to this question? What are you tryingto do?> >> (mod_limit may or may not work, but there might be anothersolution> >> which is why I am asking.)> >>> >> -MC> >>> >>> >>> Regards> >>> Chav>
are allowed to send calls toFS,> > but i want to be able to enforce "capacity" policy on the> traffic> > coming from any one of them depending on total termination capacityon> > my termination end.> > Let say GW 1 has to be limited to make 10 simultaneous calls whileGW2> > could make up to 30 and so on.>> I'm sure that this is possible. I don't personally have a way totest> 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@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@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@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 IIFreeSWITCH http://www.freeswitch.org/ClueCon http://www.cluecon.com/AIM: anthmMSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com>GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com>IRC: irc.freenode.net #freeswitchFreeSWITCH Developer Conferencesip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org>iax:guest@conference.freeswitch.org/888googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org>pstn:213-799-1400-------------- next part --------------An HTML attachment was
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anthony.minessale at g...
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| Posted: Mon Feb 09, 2009 10:40 am Post subject: [Freeswitch-users] SIP Authentication |
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See this post:
http://www.mail-archive.com/freeswitch-dev@lists.freeswitch.org/msg00926.html
On Mon, Feb 9, 2009 at 9:08 AM, Ali Al-Rubaie <kerrada2003@yahoo.com (kerrada2003@yahoo.com)> wrote:
| Quote: | 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,
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400 |
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