- FreeSWITCH-users - FreeSWITCH Mailing Lists

What am I looking for?
by Karl Schmidt 01 Jul '13

01 Jul '13

I am looking to migrate out of an old KX-T61610 system. I plan to support up to 16 old analog phone lines - so I'm looking for something I can plug into my server provides 16 analog extension connections. ( Best would be something that supported the old Panasonic KX-t phones - but I doubt that exists.) -------------------------------------------------------------------------------- Karl Schmidt EMail Karl(a)xtronics.com Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 Gumption is 99% of success. kps --------------------------------------------------------------------------------

4 4

What is rtp-autofix-timing
by Mino Haluz 01 Jul '13

01 Jul '13

Hi, can somebody explain me what parameter rtp-autofix-timing does in fact? I had to set this parameter to false in order to avoid warnings with asynchronous ptime not supported error, but I cant get any information about this on wiki. Thanks, M

2 1

Re: [Freeswitch-users] Webrtc video
by Phil Dunks 01 Jul '13

01 Jul '13

Hi Anthony No luck I'm afraid. I tried the uuid_video_refresh command and there was no effect. Phil > > From: Anthony Minessale <anthony.minessale(a)gmail.com> > Subject: Re: [Freeswitch-users] Webrtc video > Date: 28 June 2013 16:48:36 BST > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > Its a matter of understanding how vp8 transitions are handled. The codec is very picky about the data and probably changing the source does causes some errors. There are some rtcp signals to manipulate the video that we may need to add. > > When it freezes try sending the command uuid_video_refresh <uuid> on the leg who is sending the video and see if it has any effect. > > > > On Fri, Jun 28, 2013 at 10:12 AM, Phil Dunks <phil.dunks(a)netdev.co.uk> wrote: > Hi FS Users > > I made a simple web application using sipML5, which connects directly to 1.4.0+git~20130623T182400Z~2f08e40fce, and goes straight into a conference. > The audio works fine, but am experiencing problems with the video freezing or not being present. > I tried out https://webrtc.freeswitch.org/webrtc/portal.html > Enabled video, and dialed 3300 for the local 48K conference. > Seeing the same problem there. Audio fine. Video fine for first member, but goes wrong when second person joins. > Should this work for me, or is the webrtc conference video not quite ready yet? > Thanks, > Phil > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

1 0

Odp: Regex Help
by Mariusz Czulada 01 Jul '13

01 Jul '13

try: ^((\d{10})\*(\d{6})\*(1|3|12)\*)$ you should get then: $1 - everything $2 - first 10-digit part $3 - second 6-digit number $4 - third - last part - extension

3 2

Regex Help
by Vishal Kakkar 01 Jul '13

01 Jul '13

Hi all, I am trying to use regex in an dial plan to match - *10-DigitNumber*6-DigitNumber*X* * *where X can be = 1 or 3 or 12 * ** i.e. 1234567890*123456*1* 1234567890*123456*3* 1234567890*123456*12* I formed following to match first two- ^\d{10}[*]\d{6}[*][13][*]$ And this to match third one ^\d{10}[*]\d{6}[*][1][2][*]$ But i am unable to make single regex that would pass all three using OR or anything.. I want to use this regex in play_n_get_digits. Please Help -Vishal

1 0

Lost Call - New call using same port
by Mr Nathan Downes 01 Jul '13

01 Jul '13

Hi, I know this is an endpoint/network issue but want to confirm this behaviour is RFC friendly (infrastructure provider questioning it) Call from ATA (NEC Fibre ONT) to FS to PSTN. ATA calls a PSTN number and is put on hold from PaBX on that end. ATA either stops sending media or the BYE is lost (either can be assumed). Because B leg is on hold media is received and FS continues sending to A leg but receives no RTP back. 30 mins or so pass and FS attempts to end A leg. During this time ATA makes another call, sends invite using same port as before, wireshark ends up showing 2 streams going back to ATA, one from new call and one from "lost" other call. Issue this causes is NEC ONT attempts to play the 2 streams of incoming G711a creating interesting audio. Assuming because b leg is still active it is waiting for the 1800 sec session timer to expire? And if the call was to a person and not on hold music they would realise call isn't working and hang up, causing the B leg to end and FS to close A leg anyway? Thanks Nathan

1 0

call_timeout and lua in dialplan
by Vincent Xia 01 Jul '13

01 Jul '13

in a dialplan extension, i have <action application="set" data="call_timeout=30" /> then <action application="lua" data="bridge.lua ${effective_caller_id_number} $1"> in which call bridging is to be processed, after this, is <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}" /> to transfer the call to voicemail. the purpose is to test no-answer till timeout and see if the call is transferred as planned. but nothing happened untill 60 secs after the dial, which is the default value of call_timeout. in my understanding, when lua takes over, the channel variable call_timeout is no longer effective and will not affect what happens in lua script, so i tried session:execute("set", "call_timeout=30"); in bridge.lua, but the problem is still there. how to make call_timeout work with lua, could anyone figure this out?

2 2

music on hold problem with Sangoma A200 board on FS
by Tamer Higazi 30 Jun '13

30 Jun '13

Hi people! I am running FreeSWITCH Version 1.5.2b+git~20130618T211908Z~9cf05e3595 (git 9cf05e3 2013-06-18 21:19:08Z) and I have a problem with music on hold with my Sangoma A200 board, and I don't know how to solve it. Everytime, the other party puts me on hold, I don't hear anything and the connection is dropped. Any ideas?! For any ideas solving this problem, I woudl kindly thank you. Tamer My freetdm.conf.xml looks like this: configuration name="freetdm.conf" description="Freetdm Configuration"> <settings> <param name="debug" value="0"/> <param name="hold-music" value="$${hold_music}"/> <param name="enable-analog-option" value="call-swap"/> <param name="enable-analog-option" value="3-way"/> </settings> <config_profiles> </config_profiles> <analog_spans> <span name="FXS">  <param name="dialplan" value="XML"/> <param name="context" value="default"/>     </span> </analog_spans> </configuration> and the console output is here: 2013-06-27 16:44:13.286747 [DEBUG] switch_channel.c:1746 (sofia/external/061312000) Callstate Change ACTIVE -> HELD 2013-06-27 16:44:13.286747 [DEBUG] switch_core_session.c:1141 Send signal FreeTDM/1:1/0061312000 [BREAK] 2013-06-27 16:44:13.286747 [DEBUG] switch_core_session.c:861 Send signal FreeTDM/1:1/0061312000 [BREAK] 2013-06-27 16:44:13.336605 [DEBUG] switch_ivr.c:612 FreeTDM/1:1/0061312000 Command Execute playback(local_stream://moh) EXECUTE FreeTDM/1:1/0061312000 playback(local_stream://moh) 2013-06-27 16:44:13.336605 [WARNING] mod_local_stream.c:469 Unknown source moh, trying 'default' 2013-06-27 16:44:13.336605 [ERR] mod_local_stream.c:478 Unknown source default 2013-06-27 16:44:13.336605 [DEBUG] switch_core_session.c:861 Send signal FreeTDM/1:1/0061312000 [BREAK] 2013-06-27 16:44:13.336605 [DEBUG] switch_ivr_bridge.c:388 Send signal sofia/external/061312000 [BREAK] 2013-06-27 16:44:13.536666 [DEBUG] switch_core_media.c:2723 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:10:64000] 2013-06-27 16:44:13.536666 [DEBUG] switch_core_media.c:1748 Already using PCMA 2013-06-27 16:44:13.536666 [DEBUG] switch_core_media.c:2885 Set 2833 dtmf send payload to 101 2013-06-27 16:44:13.536666 [DEBUG] switch_core_media.c:3885 Audio params are unchanged for sofia/external/061312000. 2013-06-27 16:44:13.536666 [DEBUG] switch_core_media.c:3895 sofia/external/061312000 Setting audio receive payload in Re-INVITE to 8 2013-06-27 16:44:13.536666 [DEBUG] sofia.c:6286 Processing updated SDP 2013-06-27 16:44:13.536666 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/061312000 [BREAK] 2013-06-27 16:44:13.546766 [DEBUG] sofia.c:5745 Channel sofia/external/061312000 entering state [completed][200] 2013-06-27 16:44:13.606614 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/061312000 [BREAK] 2013-06-27 16:44:13.606614 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/061312000 [BREAK] 2013-06-27 16:44:13.606614 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/061312000 [BREAK] 2013-06-27 16:44:13.616646 [DEBUG] sofia.c:5745 Channel sofia/external/061312000 entering state [ready][200] 2013-06-27 16:44:23.186805 [DEBUG] ftmod_wanpipe.c:1678 [s1c1][1:3] read wanpipe event 5 2013-06-27 16:44:23.186805 [DEBUG] ftmod_wanpipe.c:1462 [s1c1][1:3] Got wanpipe ONHOOK 2013-06-27 16:44:23.186805 [DEBUG] ftmod_wanpipe.c:1685 [s1c1][1:3] Ignoring event for now 2013-06-27 16:44:23.936625 [DEBUG] ftmod_wanpipe.c:1661 [s1c1][1:3] Diff since last event = 752 ms, delivering ONHOOK now 2013-06-27 16:44:23.936625 [DEBUG] ftmod_analog.c:999 [s1c1][1:3] Received event [ONHOOK] in state [UP] 2013-06-27 16:44:23.936625 [DEBUG] ftmod_analog.c:1052 [s1c1][1:3] Changed state from UP to DOWN 2013-06-27 16:44:23.946747 [DEBUG] ftmod_analog.c:638 [s1c1][1:3] Completed state change from UP to DOWN in 6 ms 2013-06-27 16:44:23.946747 [DEBUG] ftmod_analog.c:644 [s1c1][1:3] Executing state handler on 1:1 for DOWN 2013-06-27 16:44:23.946747 [DEBUG] mod_freetdm.c:2322 got FXS sig [STOP] 2013-06-27 16:44:23.946747 [NOTICE] mod_freetdm.c:2414 Hangup FreeTDM/1:1/0061312000 [CS_EXECUTE] [NORMAL_CLEARING] 2013-06-27 16:44:23.946747 [DEBUG] switch_channel.c:3135 Send signal FreeTDM/1:1/0061312000 [KILL] 2013-06-27 16:44:23.946747 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/1:1/0061312000 [BREAK] 2013-06-27 16:44:23.946747 [DEBUG] ftdm_io.c:3180 [s1c1][1:3] DTMF debug is already disabled 2013-06-27 16:44:23.946747 [DEBUG] ftdm_io.c:3212 [s1c1][1:3] No need to disable input dump 2013-06-27 16:44:23.946747 [DEBUG] ftdm_io.c:3243 [s1c1][1:3] No need to disable output dump 2013-06-27 16:44:23.946747 [DEBUG] ftdm_io.c:6798 Cleared call with id 12 2013-06-27 16:44:23.946747 [DEBUG] ftdm_io.c:2985 [s1c1][1:3] channel done 2013-06-27 16:44:23.946747 [DEBUG] ftmod_analog.c:972 [s1c1][1:3] ANALOG CHANNEL 1:1 thread ended. 2013-06-27 16:44:23.946747 [WARNING] ftdm_io.c:4286 [s1c1][1:3] cannot read from channel that is not open 2013-06-27 16:44:23.946747 [WARNING] mod_freetdm.c:792 Failed to read from channel FreeTDM/1:1/0061312000 device 1:1! 2013-06-27 16:44:23.946747 [DEBUG] switch_ivr_bridge.c:642 BRIDGE THREAD DONE [FreeTDM/1:1/0061312000] 2013-06-27 16:44:23.946747 [DEBUG] switch_ivr_bridge.c:667 Send signal sofia/external/061312000 [BREAK] 2013-06-27 16:44:23.956746 [DEBUG] switch_ivr_bridge.c:642 BRIDGE THREAD DONE [sofia/external/061312000] 2013-06-27 16:44:23.956746 [DEBUG] switch_ivr_bridge.c:667 Send signal FreeTDM/1:1/0061312000 [BREAK] 2013-06-27 16:44:23.956746 [NOTICE] switch_ivr_bridge.c:730 Hangup sofia/external/061312000 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2013-06-27 16:44:23.956746 [DEBUG] switch_channel.c:3135 Send signal sofia/external/061312000 [KILL] 2013-06-27 16:44:23.956746 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/061312000 [BREAK] 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:481 (sofia/external/061312000) State EXCHANGE_MEDIA going to sleep 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:416 (sofia/external/061312000) Running State Change CS_HANGUP 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:679 (sofia/external/061312000) State HANGUP 2013-06-27 16:44:23.956746 [DEBUG] mod_sofia.c:463 Channel sofia/external/061312000 hanging up, cause: NORMAL_CLEARING 2013-06-27 16:44:23.956746 [DEBUG] mod_sofia.c:515 Sending BYE to sofia/external/061312000 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:48 sofia/external/061312000 Standard HANGUP, cause: NORMAL_CLEARING 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:679 (sofia/external/061312000) State HANGUP going to sleep 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:692 (sofia/external/061312000) Callstate Change HELD -> HANGUP 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:447 (sofia/external/061312000) State Change CS_HANGUP -> CS_REPORTING 2013-06-27 16:44:23.956746 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/061312000 [BREAK] 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:416 (sofia/external/061312000) Running State Change CS_REPORTING 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:764 (sofia/external/061312000) State REPORTING 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:92 sofia/external/061312000 Standard REPORTING, cause: NORMAL_CLEARING 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:764 (sofia/external/061312000) State REPORTING going to sleep 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:441 (sofia/external/061312000) State Change CS_REPORTING -> CS_DESTROY 2013-06-27 16:44:23.956746 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/061312000 [BREAK] 2013-06-27 16:44:23.956746 [DEBUG] switch_core_session.c:1549 Session 24 (sofia/external/061312000) Locked, Waiting on external entities 2013-06-27 16:44:23.956746 [DEBUG] switch_ivr_bridge.c:1531 FreeTDM/1:1/0061312000 skip receive message [UNBRIDGE] (channel is hungup already) 2013-06-27 16:44:23.956746 [NOTICE] switch_core_session.c:1567 Session 24 (sofia/external/061312000) Ended 2013-06-27 16:44:23.956746 [NOTICE] switch_core_session.c:1571 Close Channel sofia/external/061312000 [CS_DESTROY] 2013-06-27 16:44:23.956746 [DEBUG] switch_core_session.c:2739 FreeTDM/1:1/0061312000 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:478 (FreeTDM/1:1/0061312000) State EXECUTE going to sleep 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:416 (FreeTDM/1:1/0061312000) Running State Change CS_HANGUP 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:679 (FreeTDM/1:1/0061312000) State HANGUP 2013-06-27 16:44:23.956746 [DEBUG] mod_freetdm.c:544 [1:1] FreeTDM/1:1/0061312000 CHANNEL HANGUP ENTER 2013-06-27 16:44:23.956746 [DEBUG] mod_freetdm.c:569 Device [1:1] is no longer attached to FreeTDM/1:1/0061312000. Nothing to do. 2013-06-27 16:44:23.956746 [DEBUG] mod_freetdm.c:640 [1:1] FreeTDM/1:1/0061312000 CHANNEL HANGUP EXIT 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:48 FreeTDM/1:1/0061312000 Standard HANGUP, cause: NORMAL_CLEARING 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:679 (FreeTDM/1:1/0061312000) State HANGUP going to sleep 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:692 (FreeTDM/1:1/0061312000) Callstate Change ACTIVE -> HANGUP 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:566 (sofia/external/061312000) Callstate Change HANGUP -> DOWN 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:447 (FreeTDM/1:1/0061312000) State Change CS_HANGUP -> CS_REPORTING 2013-06-27 16:44:23.956746 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/1:1/0061312000 [BREAK] 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:416 (FreeTDM/1:1/0061312000) Running State Change CS_REPORTING 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:764 (FreeTDM/1:1/0061312000) State REPORTING 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:569 (sofia/external/061312000) Running State Change CS_DESTROY 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:579 (sofia/external/061312000) State DESTROY 2013-06-27 16:44:23.956746 [DEBUG] mod_sofia.c:373 sofia/external/061312000 SOFIA DESTROY 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:99 sofia/external/061312000 Standard DESTROY 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:579 (sofia/external/061312000) State DESTROY going to sleep 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:92 FreeTDM/1:1/0061312000 Standard REPORTING, cause: NORMAL_CLEARING 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:764 (FreeTDM/1:1/0061312000) State REPORTING going to sleep 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:441 (FreeTDM/1:1/0061312000) State Change CS_REPORTING -> CS_DESTROY 2013-06-27 16:44:23.956746 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/1:1/0061312000 [BREAK] 2013-06-27 16:44:23.956746 [DEBUG] switch_core_session.c:1549 Session 23 (FreeTDM/1:1/0061312000) Locked, Waiting on external entities 2013-06-27 16:44:23.956746 [NOTICE] switch_core_session.c:1567 Session 23 (FreeTDM/1:1/0061312000) Ended 2013-06-27 16:44:23.956746 [NOTICE] switch_core_session.c:1571 Close Channel FreeTDM/1:1/0061312000 [CS_DESTROY] 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:566 (FreeTDM/1:1/0061312000) Callstate Change HANGUP -> DOWN 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:569 (FreeTDM/1:1/0061312000) Running State Change CS_DESTROY 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:579 (FreeTDM/1:1/0061312000) State DESTROY 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:99 FreeTDM/1:1/0061312000 Standard DESTROY 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:579 (FreeTDM/1:1/0061312000) State DESTROY going to sleep freeswitch@internal>

1 0

Return raw file.
by Mimiko 30 Jun '13

30 Jun '13

Hello. I want to use mod_xml_rpc, or something else to return raw audio files which a recorded on file system using lua script to verify credentials and select file. For now I enable mod_xml_rpc and created a simple lua script: local f=assert(io.open("/path/file.mp3","rb")) io.write( f:read("*all")) f:close() Using http://freeswitch:ip:8080/api/luarun?script.lua gives "+OK", but not the file content. Using http://freeswitch:ip:8080/api/lua?script.lua gives error. Instead I change: local f=assert(io.open("/path/file.mp3","rb")) stream:write( f:read("*all")) f:close() gives nothing thru mod_xml_rpc, but on console outputs the content of the file if file is text. If file is mp3, the output is just some rando three characters. How to get the whole content fro an audio file using lua and some access method? -- Mimiko desu.

2 2

unknown event 22: 988 Incomplete offer/answer
by Mino Haluz 30 Jun '13

30 Jun '13

Hi, I have two 2 FS connected this way: caller(g729) - FS1 - FS2 - calee(g711) Everything works fine, unless I set proxy_media on FS2. And I want to set it because I do not want to transcode twice (g729-linear-g729 and g729-linear-g711). So calee rings, sends 200OK to FS2 with correct SDP v=0 o=- 197090 197090 IN IP4 x.x.x.x s=- c=IN IP4 x.x.x.x t=0 0 m=audio 16392 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv m=audio 0 RTP/AVP 4 101 13 and FS2 changes SDP this way and sends to FS1 v=0 o=- 157558 157558 IN IP4 x.x.x.x s=- c=IN IP4 x.x.x.x t=0 0 m=audio 16386 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 m=audio 0 RTP/AVP 4 96 13 a=rtpmap:96 /0 FS1, writes to log: sofia.c:1468 nua_i_media_error: unknown event 22: 988 Incomplete offer/answer and sends bye to callee and 500 No session set by user to caller. What should I do in this case? Is there any problem with SDP format? Thanks, M

1 0