July 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

effective_caller_id_name & effective_caller_id_number - not working
by Lloyd Aloysius 02 Jul '13

02 Jul '13

Hi All, When sending calls out from gateway the caller id name and number not working. Please find the freeswitch cli log and sip invite trace. Any help is appreciated. *freeswitch cli* EXECUTE sofia/sipinterface_1/barbara(a)alcan.mydomain.caset(effective_caller_id_name=Alcan Cable) 2013-07-01 19:23:21.676990 [DEBUG] mod_dptools.c:1373 sofia/sipinterface_1/ barbara(a)alcan.mydomain.ca SET [effective_caller_id_name]=[Alcan Cable] EXECUTE sofia/sipinterface_1/barbara(a)alcan.mydomain.caset(effective_caller_id_number=9052066900) 2013-07-01 19:23:21.676990 [DEBUG] mod_dptools.c:1373 sofia/sipinterface_1/ barbara(a)alcan.mydomain.ca SET [effective_caller_id_number]=[9052066900] *SIP Trace* INVITE sip:4164750401@alcan.mydomain.ca SIP/2.0 Via: SIP/2.0/UDP 192.168.240.105:5060;branch=z9hG4bK17bfd0ad4 Max-Forwards: 70 Content-Length: 285 To: 4164750401 <sip:4164750401@alcan.mydomain.ca> From: Barbara Ladner <sip:barbara@alcan.mydomain.ca>;tag=02f4abe9bef3a7f Call-ID: 43016333178f7b0882c3767aab886c1d(a)192.168.240.105 CSeq: 1894816474 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Contact: Barbara Ladner <sip:barbara@192.168.240.105:5060;transport=udp> Supported: replaces User-Agent: Aastra 9133i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 v=0 o=MxSIP 0 575681134 IN IP4 192.168.240.105 s=SIP Call c=IN IP4 192.168.240.105 t=0 0 m=audio 3000 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=silenceSupp:off - - - - SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.240.105:5060 ;branch=z9hG4bK17bfd0ad4;received=99.247.64.220;rport=16511 From: Barbara Ladner <sip:barbara@alcan.mydomain.ca>;tag=02f4abe9bef3a7f To: 4164750401 <sip:4164750401@alcan.mydomain.ca> Call-ID: 43016333178f7b0882c3767aab886c1d(a)192.168.240.105 CSeq: 1894816474 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.3b+git~20130628T160527Z~a52a604fbb Content-Length: 0 * * Thanks Lloyd

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ZRTP security bugs
by Ben 01 Jul '13

01 Jul '13

Hi, Does the recent news about the flaws in the ZRTPCPP library (http://blog.azimuthsecurity.com/2013/06/attacking-crypto-phones-weaknesses- in.html) impact the Freeswitch implementation of ZRTP in any way? Appreciate this is fairly fresh news, so in a way it's more of an FYI! Thanks.

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No voicemail after attempted FusionPBX upgrade
by Phil Quesinberry 01 Jul '13

01 Jul '13

Just attempted an upgrade to FusionPBX and it appears to have messed up lua. I completely backed up the FS config (/usr/local/freeswitch/conf) as well as the FusionPBX files and database before attempting this but reverting to the old configuration doesn't fix the problem, despite doing a reloadxml/restarting FS, etc. I think FusionPBX may have made some changes to the scripts directory which I hadn't backed up. Just to be sure, I recompiled with 'make current' but that didn't help either. I did make a snapshot of the VM that FS is running on a day or two ago but I'd like to understand what happened. I now get the following errors in the console output when attempting to enter voicemail: EXECUTE sofia/internal/102@192.168.1.6:5060 set(voicemail_action=check) 2013-07-01 15:10:24.774751 [DEBUG] mod_dptools.c:1373 sofia/internal/102@192.168.1.6:5060 SET [voicemail_action]=[check] EXECUTE sofia/internal/102@192.168.1.6:5060 set(voicemail_profile=default) 2013-07-01 15:10:24.774751 [DEBUG] mod_dptools.c:1373 sofia/internal/102@192.168.1.6:5060 SET [voicemail_profile]=[default] EXECUTE sofia/internal/102@192.168.1.6:5060 lua(app.lua voicemail) 2013-07-01 15:10:24.774751 [ERR] mod_lua.cpp:198 cannot open /usr/local/freeswitch/scripts/resources/functions/file_exists.lua: No such file or directory stack traceback: [C]: in function 'dofile' /usr/local/freeswitch/scripts/resources/config.lua:20: in main chunk [C]: in function 'dofile' /usr/local/freeswitch/scripts/app.lua:28: in main chunk 2013-07-01 15:10:24.774751 [DEBUG] switch_cpp.cpp:1050 sofia/internal/102@192.168.1.6:5060 destroy/unlink session from object 2013-07-01 15:10:24.774751 [NOTICE] switch_core_state_machine.c:262 sofia/internal/102@192.168.1.6:5060 has executed the last dialplan instruction, hanging up. 2013-07-01 15:10:24.774751 [NOTICE] switch_core_state_machine.c:264 Hangup sofia/internal/102@192.168.1.6:5060 [CS_EXECUTE] [NORMAL_CLEARING] Doing a sofia loglevel all 9 or SLA debug doesn't really seem to generate any additional useful info pertaining to this problem, at least not that I can see. Any ideas on what might have happened here? FusionPBX comes in really handy for the more routine admin tasks but in my experience it doesn't appear to upgrade very well. Otherwise, it seems quite polished. This is the first time it's taken out my FS config though.

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How to get callid
by Mino Haluz 01 Jul '13

01 Jul '13

Hi, before briding I want to add to custom header callid of inbound call. <action application="set" data="sip_h_X-Origination-Leg=???"/> what should I put there so that it will be filled with this callid? Thanks M

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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 --------------------------------------------------------------------------------

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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

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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 > >

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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

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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

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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

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