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Skeleton Configuration?
by Bryan Smart 05 Jul '11

05 Jul '11

Is a truly skeletal configuration available? As far as I can tell, the choice right now is to start with no configuration files at all, or else to install a sample set that is bursting with stuff. So far, I've installed the samples, and tried to strip out the parts that are obviously not needed by me: The dialplan is the best example that I can think of: all the exts that replace functions on Snom/Linksys phones, record/play video, demo substitute ring tone, call groups, add/del groups, dynamically create numerous types of conferences, call features like redial/hold/etc, the exts for the demo IVR, and so on. I know enough to know that I don't want those on a live system, but I don't know enough to know if it is safe to remove parts. Will I break refer or deflect if I get rid of the "refer" ext? Is "unloop" only necessary as a safety feature if I'm connecting sessions to SIP URIs outside of the local FS? ? Can I drop "external-sip-uri" if users won't be calling external URIs, or will that completely block FS from connecting to external URIs in all cases? I think I can dump features.xml altogether. There are an internal and external SIP profile. If I'm only hosting a voice application, I think I can dump the internal profile and just run the default external profile on 5060, but there are a huge amount of options in there that look as if they're setup to cover cases where the external profile is used to connect to ITSPs, which I won't. There are so many configuration files that I'm not comfortable with blowing the whole directory tree away and recreating a raw basics version of each of them. While I don't feel paranoid about a specific part of the security of the full samples install, my rule is always that I try to not run anything that I don't at least somewhat understand. There seem to be so many little demos and optional features that I know that I by no means understand the implications of what they could be used to do in combination with what I'll add later. It seems safer to start from the bare basics, and only add what is needed, but, for now, I must start with the samples, and try to strip away as much as I can without breaking FS. For me, I'm using FS as a platform for running a voice app, so I need little, if any, PBX type features. Some people that might wish to use FS as a PBX might not want all of the demo bits, though. An absolute minimum set of configuration files, as well as a set containing the essentials for a app platform or PBX would be useful, if they don't already exist somewhere. Bryan

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Video is not display
by sc_zhangming 05 Jul '11

05 Jul '11

hi: if use fs_cli originate user/70001 700002 agent is not vedio if i use x-lite 4 input number the call have vedio my dialplan configuration information: <condition field="destination_number" expression="^(7[0-4][0-9][0-9][0-9][0-9])$"> <action application="set" data="bypass_media=true"/> <action application="bridge" data="user/$1@${domain_name}"/> <action application="answer"/> </condition> </extension> log: 2011-07-07 19:04:31.427458 [DEBUG] sofia.c:4669 Remote SDP: v=0 o=- 12954249923203125 3 IN IP4 10.108.226.238 s=CounterPath X-Lite 4.0 c=IN IP4 10.108.226.238 t=0 0 a=ice-ufrag:2e4e41 a=ice-pwd:1af7090a62cdba6f63b4b6bd9fd01472 m=audio 64370 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=candidate:1 1 UDP 659136 10.108.226.238 64370 typ host a=candidate:1 2 UDP 659134 10.108.226.238 64371 typ host m=video 63720 RTP/AVP 34 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1;CIF=1;VGA=2 a=candidate:1 1 UDP 659136 10.108.226.238 63720 typ host a=candidate:1 2 UDP 659134 10.108.226.238 63721 typ host 2011-07-07 19:04:31.427458 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2011-07-07 19:04:31.427458 [DEBUG] sofia_glue.c:2690 Already using PCMU 2011-07-07 19:04:31.427458 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf send payload to 101 2011-07-07 19:04:31.427458 [DEBUG] sofia_glue.c:4626 Video Codec Compare [H263:34]/[H263:34] 2011-07-07 19:04:31.427458 [DEBUG] sofia.c:5121 Processing updated SDP 2011-07-07 19:04:36.870143 [DEBUG] switch_channel.c:2535 (sofia/internal/sip:700001@10.108.226.222:39206) Callstate Change ACTIVE -> HANGUP 2011-07-07 19:04:36.870143 [NOTICE] sofia.c:537 Hangup sofia/internal/sip:700001@10.108.226.222:39206 [CS_HIBERNATE] [NORMAL_CLEARING] 2011-07-07 19:04:36.870143 [DEBUG] switch_channel.c:2551 Send signal sofia/internal/sip:700001@10.108.226.222:39206 [KILL] sc_zhangming(a)sina.com.cn

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MSVC 2010 Compile broken
by Jan Berger 03 Jul '11

03 Jul '11

Hi, I just downloaded the latest FreeSWITCH and I have not been able to compile it with Visual Studio 2010 so far. Firstly - opening FreeSWITCH.2010.sln gives an error - a missing project. Ignoring that and switching off spandsp I still end up with compiler errors in Sofia. Opening and running FreeSWITCH.2010.express.sln or converting FreeSWITCH.2008.sln gives the same result - Sofia is the first that don't compile. The actual error I get is that http_host_class is undefined and checking with the original Sofia I notice that http_protos.h that should have contained this is empty ??? I assume it's a build step that don't work or have dropped out - any ideas? Jan

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Invitation to connect on LinkedIn
by Jacob Maldonado via LinkedIn 03 Jul '11

03 Jul '11

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get answertime dialstatus after bridge
by salzh 03 Jul '11

03 Jul '11

hi. is there any way to get a call information such as answertime dialstatus and so on after executing 'bridge' application? just like what the application 'dial' do in asterisk.

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how to change conference's caller-id-number at run-time?
by ran zhang 02 Jul '11

02 Jul '11

I want when the bridging conference is inviting someone to start the conference, the caller is shown as the real caller id number rather than whats stored in 'caller-id-number' in conference.conf.xml.

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Inband DTMF set in Event Socket
by rex.alex 02 Jul '11

02 Jul '11

Hello, How to set the DTMF type as inband when I dial through event socket using api originate command? Thanks, Rex -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Inband-DTMF-set-in-Event-Sock… Sent from the freeswitch-users mailing list archive at Nabble.com.

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LUA Script: IGNORE INFO DTMF
by ankIT WALiA 02 Jul '11

02 Jul '11

Hi, I am trying to get the digits pressed using statement in my LUA script "digits = session:playAndGetDigits(3, 5, 3, 7000, "#", prompt, invalid, "\\d+")" The FS is not able to take the digits and giving the invalid response always. 2011-07-02 15:51:30.405443 [WARNING] sofia.c:6715 IGNORE INFO DTMF(7) (This channel was not configured to use INFO DTMF!) I am trying establish a channel between default 1002 ext and custom ext. Am I missing something? Thanks Ankit -- Life is like a rose its upto u feel it as its fragrance or thorns

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Linksys PAP2 tls with freeswitch
by Mohammad Emran 02 Jul '11

02 Jul '11

Hi all, Please help me to configure linksys pap2 tls with freeswitch.i have configured freeswitch as per wiki. Thank you. Sent from my iPad

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REGISTER response sent to the wrong por
by Dave Horton 02 Jul '11

02 Jul '11

I have a phone behind a firewall registering with the internal profile. The address in the Via header is not reachable, and the FS (correctly, IMO) sends the response back to the actual sending IP address. However, rather than sending it to the actual sending udp port, it incorrectly (again, IMO) sends it to the port advertised in the Via header. Of course the message does not get through the firewall, and the phone keeps retransmitting the REGISTER. Is this a bug, or am I missing something? And is there a setting where I can force the behavior that I want (which is: send responses to the sending address:port if the sending address:port is different than the Via header). Trace below.. recv 481 bytes from udp/[66.37.97.53]:1024 at 19:28:12.445081: ------------------------------------------------------------------------ REGISTER sip:65.162.239.204 SIP/2.0 Via: SIP/2.0/UDP 10.93.14.119:5064;branch=z9hG4bK-e73de4bf From: "1002" <sip:1002@65.162.239.204>;tag=7718f2af7f3dcc73o2 To: "1002" <sip:1002@65.162.239.204> Call-ID: 930c7e31-648ae3db(a)10.93.14.119 CSeq: 53622 REGISTER Max-Forwards: 70 Contact: "1002" <sip:1002@10.93.14.119:5064>;expires=3600 User-Agent: Linksys/SPA962-5.2.8(SC) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces ------------------------------------------------------------------------ send 671 bytes to udp/[66.37.97.53]:5064 at 19:28:12.445259: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.93.14.119:5064;branch=z9hG4bK-e73de4bf;received=66.37.97.53 From: "1002" <sip:1002@65.162.239.204>;tag=7718f2af7f3dcc73o2 To: "1002" <sip:1002@65.162.239.204>;tag=t1m9emDKB4trF Call-ID: 930c7e31-648ae3db(a)10.93.14.119 CSeq: 53622 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-25c725c 2011-06-30 18-30-24 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="65.162.239.204", nonce="7e426985-2f7a-40ff-a817-6c167207853f", algorithm=MD5, qop="auth" Content-Length: 0

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