September 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH 1.4.9 Released!
by Ken Rice 29 Sep '14

29 Sep '14

Hey FreeSWITCHers! The FreeSWITCH Dev Team is happy to announce FreeSWITCH 1.4.9 has been released. This release is a bug fix release. Source Tarball is available at http://files.freeswitch.org/ and packages are available via the Yum and Debian repos. K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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API Execute uuid_transfer
by andretodd＠verizon.net 29 Sep '14

29 Sep '14

3 2

Monstrous Sound Delay, WebRTC, sip-0.6.2, mod_verto
by Алексей kob.kz Целищев 29 Sep '14

29 Sep '14

Hello. I work with browser with WebRTC with FreeSWITCH Version 1.5.14b+git~20140730T044231Z~b1f382c67e~64bit (git b1f382c 2014-07-30 04:42:31Z 64bit). Since answer call until the sound is appeared (ICE connection state changed to "connected" in the log) takes ~ 1216ms, which is monstrous for my problems. Have any ideas on this? STUN, as you can see, is disabled. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.inviteservercontext | new listener added to event connecting sip-0.6.2.js:2655 >>> ||| <<< since answer call : 0 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.eventemitter | emitting event userMediaRequest sip-0.6.2.js:2655 >>> ||| <<< since answer call : 41 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | emitting event userMediaRequest sip-0.6.2.js:2655 >>> ||| <<< since answer call : 41 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.eventemitter | emitting event userMedia sip-0.6.2.js:2655 >>> ||| <<< since answer call : 43 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | emitting event userMedia sip-0.6.2.js:2655 >>> ||| <<< since answer call : 43 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | acquired local media stream sip-0.6.2.js:2655 >>> ||| <<< since answer call : 44 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.inviteservercontext | emitting event connecting sip-0.6.2.js:2655 >>> ||| <<< since answer call : 44 ms. Ожидание соединения: 47 ms. WebRTCCallCenter.js?r=1:212 Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | ICE connection state changed to "checking" sip-0.6.2.js:2655 >>> ||| <<< since answer call : 47 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | ICE Gathering Completed sip-0.6.2.js:2655 >>> ||| <<< since answer call : 47 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | emitting event iceComplete sip-0.6.2.js:2655 >>> ||| <<< since answer call : 48 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | emitting event getDescription sip-0.6.2.js:2655 >>> ||| <<< since answer call : 48 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.transaction.ist | emitting event stateChanged sip-0.6.2.js:2655 >>> ||| <<< since answer call : 50 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.inviteservercontext | emitting event accepted sip-0.6.2.js:2655 >>> ||| <<< since answer call : 51 ms. >>>>> Звонок начат Operator.js?r=3:1903 Wed Sep 17 2014 16:31:21 GMT+0600 | sip.inviteservercontext | no ACK received, attempting to retransmit OK sip-0.6.2.js:2655 >>> ||| <<< since answer call : 60 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.inviteservercontext | no ACK received, attempting to retransmit OK sip-0.6.2.js:2655 >>> ||| <<< since answer call : 80 ms. Wed Sep 17 2014 16:31:22 GMT+0600 | sip.invitecontext.mediahandler | ICE connection state changed to "connected" sip-0.6.2.js:2655 >>> ||| <<< since answer call : 1216 ms. Nuance: if the operator and the server are in the local network and connected to an internal address, then until the sound appeared goes ~ 280ms, which is a stretch of more or less acceptable. But if I connect to the browser through external network (router with port forwarding), then the delay increases by 5 times. In that demo with Verto https://webrtc.freeswitch.org/verto/index.html >>>>>>>>>>>>>>>>Thu Sep 25 2014 16:13:14 GMT+0600 ICE connection state changed to "checking" verto-min.js:41 >>> ||| <<< since answer call: 476 ms. verto-min.js:42 Dialog c46b6eb8-73de-62ce-df07-fb3d002e07eb: state change from trying to active verto-min.js:206 >>>>>>>>>>>>>>>>Thu Sep 25 2014 16:13:15 GMT+0600 ICE connection state changed to "connected" verto-min.js:41 >>> ||| <<< since answer call: 2014 ms. How do I configure Freeswitch and javascript app to sound appeared within 200 ms?

2 1

tel: uri freeswitch
by joe 29 Sep '14

29 Sep '14

2014-09-27 05:52:40.377783 [ERR] sofia_glue.c:983 URL Error! tel: uri's not supported at this time i am not able to set tel uri's in sip_invite_req_uri= during originate pstn outbound call from freeswitch. awaiting positive answer asap. tx

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Shared Presence Sofia profile config
by Nandy Dagondon 29 Sep '14

29 Sep '14

Hello, I'm trying to monitor PRESENCE of remote extension phones. In this wiki https://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Presence it says - - - - - *manage-presence* "Enable presence. If you want to share your presence (see dbname and presence-hosts) set this to "true" on the first profile and enable the shared presence database. Then on subsequent profiles that share presence set this variable to "passive" and enable the shared presence database there as well. - - - - - - If i have the ff profiles: doublenat.xml internal.xml which one is the "first profile" that I should set manage-presence=true? Does parsing of XML config files in alphabetical order? Tks. /Nandy

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Export Var
by Andre 28 Sep '14

28 Sep '14

Can you export the var to the B leg using this code? api.ExecuteString("uuid_setvar ".. Would the syntax be Usage: uuid_setvar <uuid> Export <varname> [value]

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BLF Lamps flashing orange
by Cj B 28 Sep '14

28 Sep '14

Hello list, I have a Cisco 525G and 504G both with side cars. They were working fine for a while (green lights, flash when ringing, etc) and then all of a sudden they are now just blinking orange constantly. I haven't upgraded freeswitch (1.4.6+git~20140603T193516Z~9479729340~32bit). The sidecar still seems to work, I can transfer to other lines, I can dial them, but the lights just stay flashing orange. Here's a packet capture: http://pastebin.com/vDreS2bm Does anyone know what could cause this? Thanks for any help!

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mod_v8 vs mod_lua overhead?
by Avi Marcus 28 Sep '14

28 Sep '14

Hi - I need to do an sql query with some logic and manipulation, and then route a call. I think it's a bit much to do with mod_odbc_query and the dialplan, so I'll be using mod_lua or mod_v8. Does anyone have information about the overhead about running a script in either language? I know mod_lua has been the goto scripting language for low overhead, but how does the newer mod_v8 perform? I'd rather use javascript, but only if there isn't a significant performance difference. Thanks, -Avi Marcus

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Mod_sofia registration records
by Carlos Ruiz Díaz 27 Sep '14

27 Sep '14

Hi all, is there any way for FS to consume registration records from a custom database, for example, location table of Kamailio? Or on a on-demand basis, making a Redis query for example. Thanks, -- Carlos http://caruizdiaz.com

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Mod_translate Issue
by Ave ebock 27 Sep '14

27 Sep '14

Hi, Please im trying to use mod_translate. I have this profile in my translate.conf.xml <profile name="NG"> <rule regex="^0([1-9]\d+)$" replace="234$1"/> </profile> How do i please apply it, such that before calls reach my dialplan, the number is translated accordingly. Any help would be highly appreciated Thank You

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