October 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

sip profile - accept-blind-auth vs
by Daniel-Constantin Mierla 14 Oct '12

14 Oct '12

Hello, in the past I used to set: <param name="auth-calls" value="false"/> in the sip profile in order to skip user authentication for calls. Lately I started to play a bit with 1.2 stable branch and seems that setting auth-calls to false is no longer doing what I expected, calls being challenged for user authentication. Setting instead the accept-blind-auth to false got me what I wanted, like:  <param name="accept-blind-auth" value="true"/> But from the comment (checked the wiki as well, but has the same text) is a bit unclear what is the real purpose for it. Isn't auth-calls=false supposed to accept calls without user authentication anymore? For this particular case, I play some announcements, like 'user not available', and should work also for calls coming from outside. The access is restricted by IP address ACL, allowing SIP traffic only from my Kamailio instance. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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conference.freeswitch.org native ipv6
by Brian West 13 Oct '12

13 Oct '12

sip:888@[2606:d900:0:24:c8fe:caff:fefe:cafe]:5060 Can everyone with V6 test this out? -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire http://freeswitchcookbook.com http://freeswitchbook.com T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410

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Multicast on Freeswitch
by Aleksei Zjabkin 13 Oct '12

13 Oct '12

Hi! I'm working on a project of integration radios into VoIP network. Main task is SIP to Radio intercom using multicast streams. I already have a working solution, based on Cisco E&M and twisted pair WAVE. Since Wave is a little bit costly solution, I want to implement intercom using freeswitch. Main problem is that I'm not developer and just a network guy and I need help in implementation. I had some experience with multicast steams on Asterisk and I believe that freeswitch is more flexible than asterisk. First and most important question is about how is multicast implemented on freeswitch. Here is an example from the samples included with freeswitch: <extension name="rtp_multicast_page"> <condition field="destination_number" expression="^pagegroup$|^7243"> <action application="answer"/> <action application="esf_page_group"/> </condition> </extension> expression="^pagegroup$|^7243"> - what does this expression mean? Does it mean that dialer must be a member of a pagegroup? At the beggining I need to setup freeswitch to stream a SIP call, so applications like VLC could listen it. Best Regards, Aleksei Zjabkin IT lahendused \ IT solutions GSM +372 58055975 ------------------------- Telegrupp AS Tartu mnt. 63, 10115 Tallinn Eesti \ Estonia

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FRIDAY FREE FOR ALL IS BACK!!@#!
by Ken Rice 12 Oct '12

12 Oct '12

What is it? The Friday Free For All Where is it? The FreeSWITCH Community Conference Bridge of course... How do you get there? You have many options... Via the PSTN +1-919-386-9900 Via SIP sip:888@conference.freeswitch.org (this is Both IPv4 and IPv6 enabled) Via the web http://conference.freeswitch.org (we¹re still working on fixing up the v6 here) What time? 4PM EST... Does that mean you have to wait til then? No go ahead and call in whenever you are ready Hang out start a converstation.... Whats this weeks topic? Whatever we want it to be... -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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problems with SLA on polycom phones
by KEN KELLY-TURNER 12 Oct '12

12 Oct '12

Hi, I'm new to freeswitch and I'm trying to set up 2 polycom phones running 4.0.2 firmware to share a line. I followed the wiki pages http://wiki.freeswitch.org/wiki/Shared_Line_Appearance and However, whenever I pick up an incoming call or try to make an outgoing call on one of the shared lines, the other line doesn't detect that the line is in use and I get a massive dump of 481 and 482 errors dozens of times. I have no idea what is causing this behaviour, any help would be greatly appreciated send 854 bytes to udp/[10.10.8.81]:5060 at 02:03:54.850984: ------------------------------------------------------------------------ NOTIFY sip:1000@10.10.8.81 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.1;rport;branch=z9hG4bKcHKrvF7SKSp1j Max-Forwards: 70 From: <sip:1000@10.10.11.1:5060>;tag=AWBzYjyr5AtS To: "ppcBarge2" <sip:1000@10.10.11.1>;tag=FC20B986-C3012B3F Call-ID: d753ed55-db70fc4e-bddaf407(a)10.10.8.81 CSeq: 4053008904 NOTIFY Contact: <sip:1000@10.10.11.1:5060;transport=udp> Call-Info: <sip:10.10.11.1>;appearance-index=*;appearance-state=idle User-Agent: FreeSWITCH-mod_sofia/1.2.3 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: call-info Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3600 Content-Length: 0 ------------------------------------------------------------------------ recv 405 bytes from udp/[10.10.8.81]:5060 at 02:03:54.864355: ------------------------------------------------------------------------ SIP/2.0 482 Loop Detected Via: SIP/2.0/UDP 10.10.11.1;rport;branch=z9hG4bKcHKrvF7SKSp1j From: <sip:1000@10.10.11.1:5060>;tag=AWBzYjyr5AtS To: "ppcBarge2" <sip:1000@10.10.11.1>;tag=FC20B986-C3012B3F CSeq: 2147483647 NOTIFY Call-ID: d753ed55-db70fc4e-bddaf407(a)10.10.8.81 Event: call-info User-Agent: PolycomSoundPointIP-SPIP_550-UA/4.0.2.11307_0004f2149566 Accept-Language: en Content-Length: 0 ------------------------------------------------------------------------ Thanks, Ken

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Compiling mod_h323 on Windows
by Shishko 12 Oct '12

12 Oct '12

Hi, could anybody give me basic instructions on how to build mod_h323 on Windows using Visual Studio 2010? Thanks

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sofia recover not working in latest GIT
by Vik Killa 12 Oct '12

12 Oct '12

Testing HA using sofia recover on latest GIT (e1fe289) and it appears the 'sofia recover' no longer works correctly. Following this tutorial (which I wrote and know worked on previous GIT version) http://wiki.freeswitch.org/wiki/Freeswitch_HA#Floating_IP_method_on_Debian_… The calls appear to be recovered when 'sofia recover' is used but there is no audio and if one party hangs up, the other leg is not terminated. Could anyone out there test and verify the 'sofia recover' is not working in latest GIT? Thanks

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How to handle SIP_REFER transfer in Bridged Call
by Subhash 12 Oct '12

12 Oct '12

Hi all, I need to handle blind transfer in VoIP call using REFER. Initial call is bridged using bridge application dialplan, post bridge either of the leg should be able to do blind transfer. How do i create a dialplan which can do this functionality. NOTE: Here FreeSWITCH only bridges the call, either of the two legs can do blind transfer. -- Thanks, Subhash.

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PocketSphinx
by Andy Ayers 12 Oct '12

12 Oct '12

Hi, I have users calling into an ivr and speaking an alphanumeric job reference number like 'DT1234E'. Is it possible to use pocketsphinx to translate this into the actual string? Could anyone supply the necessary javascript or dialplan to do this? Any help much appreciated. Thanks Andy

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FreeSWITCH Websites are ipv6 native… w00t!
by Brian West 12 Oct '12

12 Oct '12

The bulk of our websites are now v6 capable and have published AAAA records. Please report any issues you see to me please. Thanks, -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire http://freeswitchcookbook.com http://freeswitchbook.com T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410

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