October 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

RTMP + G.729 Codec
by Balamurugan Mahendran 04 Oct '12

04 Oct '12

I am working on rtmp call quality, Is it possible to use g.729 for rtmp? and will it increase the quality? Now I am experiencing Echo + large delay and background noise. Thanks, Bala

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stable mod_opal compilation problem
by Serge S. Yuriev 04 Oct '12

04 Oct '12

Hi, /home/nevian/work/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function 'switch_status_t FSConnection::read_frame(const OpalMediaType&, switch_frame_t**, switch_io_flag_t)': /home/nevian/work/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1208:20: error: 'class OpalCall' has no member named 'IsSwitchingT38' /home/nevian/work/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function 'switch_status_t FSConnection::write_frame(const OpalMediaType&, const switch_frame_t*, switch_io_flag_t)': /home/nevian/work/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1225:19: error: 'class OpalCall' has no member named 'IsSwitchingT38' make[6]: *** [mod_opal.lo] Error 1 make[6]: Leaving directory `/home/nevian/work/freeswitch/src/mod/endpoints/mod_opal' Why mod_opal in STABLE branch requires SVN version of libopal?? IMHO this is incorrect dependency. I know this should be in JIRA but where - build system or mod_opal? -- wbr, Serge

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Help on freeswitch : gtalk-sip conversion
by Sanjay Soni 04 Oct '12

04 Oct '12

Hi, I am trying to get a call working from Gtalk client (Running on my laptop) to the SIP device(Registered as user name 1000). When gtalk client make a call, call is accepted by SIP device but the media is not flowing between Freeswitch and gtalk client. Please note that all the three (Freeswitch machine, gtalk client and SIP device) are in same private LAN inside the firewall. I see that freeswitch is rejecting the local candidate sent by gtalk client and selecting the stun candidate. Why is so ? How can I stop it from doing this. If it could select local candidate I feel call could establish. Thanks Sanjay

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GSMOPEN sms
by Sameer Khan 04 Oct '12

04 Oct '12

Hi, I can't see incoming sms in event plain message I'm using huawei e1550, Is there anything else needed apart from default config?

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proxy_media breaks T38 passthru
by Colin Mason 04 Oct '12

04 Oct '12

I am using Cisco SPA2102's for T38 faxing into a corporation with 20 fax machines. I have a freeswitch box that acts as a router to route SIP from our carrier network into our customers private network. This is the only function this box performs. The profiles on this box have: <param name="inbound-late-negotiation" value="true"/> <param name="t38-passthru" value="true"/> Faxes complete 99% of the time. If I enable <action application="set" data="proxy_media=true"/> right before the bridge it completely breaks T38 faxing. Success rate of 0%. Does anyone know why this is? Thanks, Colin

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Google Voice inboaund call to FS
by Sanjay Soni 04 Oct '12

04 Oct '12

Hi, I am trying to get 'Normal PSTN--call--GV#--forwarded to Gmail configured on FS-forwarded to sip phone' set-up working. I want to know whether this method even possible using Freeswitch OR there are some issues which needs to be sorted ? Because I see here http://wiki.freeswitch.org/wiki/Google_Voice in section 'Test Notes' that It does not work. The small answer I need here is one of these - A) It works, You are doing some mistake. Re-check your configs. B) Its just not possible. Google does not allow forwarding to other clients other than gmail logged in through browser. C) Its just not possible. FS does not have required code in there. What works for me is -> Computer (other's gmail account)-audio call-Gmail configured on FS--sip phone.

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Weekly FreeSWITCH Conference Call
by Ken Rice 03 Oct '12

03 Oct '12

Hey Guys, we¹re getting started on the normal wed conf call... Join us at sip:888@conference.freeswitch.org Please note there was a DNS change on this several days ago, if you are not getting in, flush your DNS cache, and try again K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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Maintenance Note: FreeSWITCH.org Sites
by Michael Collins 03 Oct '12

03 Oct '12

Hello all! Just a note: this coming Friday night at 9PM CST we will be performing some maintenance on a number of sites that inhabit the freeswitch.org domain. This includes wiki.freeswitch.org and www.freeswitch.org. The maintenance probably won't last more than an hour or two. Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Endpoint Registration - User name and Authorization user name
by Lloyd Aloysius 03 Oct '12

03 Oct '12

Hi All: By default in freeswitch user id is use for *User name* and *Authorization user name *in the Endpoint Configuration. Can we setup different User Name and Authorization user name? Thanks Lloyd

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Inbound DID call on Digium PRI
by mahesh katta 03 Oct '12

03 Oct '12

Hi List, I am using Digium PRU card E1 express. In My PRI line has 100 DID's enabled. But when I call on DID number not getting incomming call but I am getting call on pilot/boarding number. can you suggest me is there any parameter to enable for this PRI Card ? Thanking you. Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 222, Arunvihar,Sector-28, Noida 201301 GSM +91.99993 45699 | Phone +91.12.0431.0581 Web http://www.buzzworks.com

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