March 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch Video Conferencing required
by Naveen Khanna 03 Mar '15

03 Mar '15

HI, Suggest video conference solution with Freeswitch that can be supported. Need minimum 16 party video conference sessions with streaming server & multicast capability. Regards, Naveen Khanna

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Javascript, session_in_hangup_hook and api_hangup_hook
by Roman_ 03 Mar '15

03 Mar '15

Hi, I have been trying to run a javascript on hangup in my dialplan, where I need access to the session in the script, but I cannot get it to work. Searching the mailing list also has not revealed anything pertinent (except that it *should* work for at least lua and javascript). My (partial) dialplan looks as follows: <action application="set" data="session_in_hangup_hook=true"/> <action application="set" data="api_hangup_hook=jsrun /etc/freeswitch/scripts/script.js"/> <action application="bridge" data="..."/> The javascript gets called reliably, however, the session object is not available. If I try to access it, I get an error: 2015-03-02 22:19:22.260371 [ERR] script.js:7 Exception: TypeError: Object #<error> has no method 'getVariable' (near: "var total_billed = session.getVariable("nibble_total_billed");") So it seems the session object isn't actually a session, but some error? Just printing the session object to the console will output "false". Is there something I am missing here? Has anybody managed to use the session object in a javascript script in the api hangup hook? I am using FreeSWITCH Version 1.4.15-1~64bit (-1 64bit) (the debian packages). Any help would be greatly appreciated. Thanks and best regards, Roman -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Javascript-session-in-hangup-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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mod_callcenter not updating
by Ali Jibran 03 Mar '15

03 Mar '15

I am using FreeSWITCH (Version 1.5.15b git 556cb5c 2015-02-05 00:55:29Z 64bit). For some reason "reload mod_callcenter" doesn't update when I edit callcenter.conf.xml. It only updates agents being added but the ones that have been removed do not go away. Is it some bug?

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FreeSWITCH Week in Review (Master Branch) February 21st-27th
by Ken Rice 02 Mar '15

02 Mar '15

New Post on freeswitch.org from Kathleen check it out at http://ift.tt/1DLKdYZ FreeSWITCH Week in Review (Master Branch) February 21st-27th Hello, again. This passed week in the FreeSWITCH master branch we had 11 commits. The features for this week are: updating mod_verto to proxy additional variables and the ability to force URL refresh in mod_http_cache. Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-7312 Update mod_verto to proxy additional variables FS-7323 Add ability to force URL refresh in mod_http_cache using {refresh=true} parameter that can be prefixed to a URL to force refresh when using http:// https:// file formats or the http_get API. And added http_remove_cache API call to manually expire a cached URL. Improvements in cross platform build supports: FS-6520 Fix for libv8 build issue using MSVC 2013 The following bugs were squashed: FS-7307 Fixed buffering issue when recording calls in native format FS-7126 Fixed coredump when calling the translate application FS-7314 Fix for configure error caused by a broken openssl 1.0.2 includes FS-7313 Fix for coredump when passing invalid params to the vm_fsdb_msg_email api in mod_voicemail FS-7322 Fix for issues building on centos 5 and others distributions with older autotools FS-6758 Fixed issue with hold dropping calls on Skinny Cisco 7961G

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freeswitch say application currency in multipal language
by bhavik patel 02 Mar '15

02 Mar '15

Hi all, I use free switch and i want to play sounds file like if user has credit in USD then doller.wav file play and EUR then another file will be play. Currently it play doller.wav by default in /usr/local/freeswitch/sounds/currency/en/doller.wav but i want to play EUR so how can this possible. Is that any easy way to do this thing in multi language currency play in say application. i use this syntax in my free-switch dial plan $dialstring = "<action application=\"say\" data=\"en CURRENCY PRONOUNCED $credit_balance\"/>"; Thanks In advance... -- Thanks, Bhavik Patel

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ICE SDP Params
by Rob Moore 02 Mar '15

02 Mar '15

Hi All, After a recent upgrade of Freeswitch I've been seeing some unusual behaviour in codec choice being made by our clients hardware during late negotiation calls. No other changes have taken place that could have caused this unusual behaviour so we assume that it must be something the newer version of freeswitch is doing in this situation. Comparing traces before and after the upgrade, the SDP in the final 200 OK's from our freeswitch now contains more parameters relating to ICE and other source specific attributes. I expect these additional params are what is upsetting our clients hardware. I've since attempted to disable all forms of NAT management in Freeswitch in an attempt to get rid of these extra SDP attributes but none of them seem to have had any effect: Setting the following in the sip Profile to disable stun / NAT. http://wiki.freeswitch.org/wiki/Sofia.conf.xml#stun-auto-disable http://wiki.freeswitch.org/wiki/Sofia.conf.xml#stun-enabled ensuring the following are set to the local ip of the server <X-PRE-PROCESS cmd="set" data="external_sip_ip=localIP "/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=localIP "/> Does anyone have any suggestions on how I can remove these additional SDP params? I've included an example good and bad 200 ok in case I've missed anything else that's obvious. Many thanks Rob Bad 200 ok: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.18.4.78;branch=z9hG4bK0ba4.7a1f3a47e3eb3a2a4e8e13fd7dda8ab8.0 Via: SIP/2.0/UDP 10.9.138.26:5060;rport=5060;branch=z9hG4bK9ivhm3hvpf5c7vrsu2gh7ou4o2 Record-Route: <sip: 172.18.4.78;r2=on;lr=on;rtpi=1;nat=yes> Record-Route: <sip:10.16.0.3;r2=on;lr=on;rtpi=1;nat=yes> From: <sip:8448224432@10.150.35.124>;tag=130e56d9-dcc0-4483-9c8e-edf93ab1fb9a-33986537 To: <sip:00353876721075@10.9.138.9>;tag=r3ZZc86S529mD Call-ID: b87ef500-4f012e3b-1f738-7c23960a(a)10.150.35.124 CSeq: 101 INVITE Contact: <sip:mod_sofia@172.18.4.251:5060> User-Agent: Aeriandi Tel Server Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 841 x-nt-location: -1 v=0 o=FreeSWITCH 1425007009 1425007010 IN IP4 172.18.4.251 s=FreeSWITCH c=IN IP4 172.18.4.251 t=0 0 a=msid-semantic: WMS hGYoorbKxEnXtlapBPoffcf3QQg7ijgq m=audio 19630 RTP/SAVPF 18 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fingerprint:sha-256 18:6A:21:5F:BF:02:8B:52:29:96:85:6B:05:99:B2:9D:C7:B3:26:DB:F9:32:5A:90:51:62:E8:21:3E:C2:23:C3 a=rtcp-mux a=rtcp:19630 IN IP4 172.18.4.251 a=ssrc:3641993291 cname:Q4E4mGFKbSxu4yJv a=ssrc:3641993291 msid:hGYoorbKxEnXtlapBPoffcf3QQg7ijgq a0 a=ssrc:3641993291 mslabel:hGYoorbKxEnXtlapBPoffcf3QQg7ijgq a=ssrc:3641993291 label:hGYoorbKxEnXtlapBPoffcf3QQg7ijgqa0 a=ice-ufrag:eYAj0GcHniFvsLL8 a=ice-pwd:Gg3SMysNgP8bdIwhwXqnttUH a=candidate:1204810811 1 udp 659136 172.18.4.251 19630 typ host generation 0 a=ptime:20 good 200 ok: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.18.4.78;branch=z9hG4bKef6a.6cc24c86d42011b141c967c04c23e077.0 Via: SIP/2.0/UDP 10.9.138.26:5060;rport=5060;branch=z9hG4bKa2v9mvku4854npk24c4tn51vv2 Record-Route: <sip:172.18.4.78;r2=on;lr=on;rtpi=1;nat=yes> Record-Route: <sip:10.16.0.3;r2=on;lr=on;rtpi=1;nat=yes> From: <sip:8448224432@10.151.35.123>;tag=130e56d9-dcc0-4483-9c8e-edf93ab1fb9a-33990014 To: <sip:07940845322@10.9.138.9>;tag=Fy3a1Zv7Kt6Xm Call-ID: f59cb480-4f014116-1f782-7b23970a(a)10.151.35.123 CSeq: 101 INVITE Contact: <sip:mod_sofia@172.18.4.254:5060> User-Agent: Aeriandi Tel Server Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 224 x-nt-location: -1 v=0 o=FreeSWITCH 1425014601 1425014602 IN IP4 172.18.4.254 s=FreeSWITCH c=IN IP4 172.18.4.254 t=0 0 m=audio 16858 RTP/AVP 18 0 8 101 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20

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Getting FS to place Outbound calls
by T Fred Farmington 02 Mar '15

02 Mar '15

I have made some progress in learning FreeSWITCH, but now the next hurdle. I have my in-house softphones connecting to each other and I have an in-house SIP hard-phone connecting to the various in-house extensions. And I can place a call to my 'outside' number which utilizes the SIP line to connect to one of my in-house extensions. That is progress! Now I want to get one of my in-house extensions to be able to connect to an 'outside' number via my single inbound/outbound SIP line. I have followed the advice found on the web, but it is not working. 1. My Firewall is open to port 5060 2. Within the directory: conf\sip_profiles\external I have created a new XML file velocity_outbound.xml and within it I configured a gateway Since my SIP line provider indicates that: No SIP authentication is required. I set the parameters as follows <include> <gateway name="velocity-outbound"> <param name="username" value=""/>  <param name="password" value="" />  <param name="proxy" value="<my SIP line IP address here>"/> <param name="register" value="false"/>   <param name="register-transport" value="udp"/> </gateway> </include> 3. In the directory: conf\dialplan\default\ I created a new file: outbound_via_velocity.xml in which I defined what to do: <include> <extension name="domestic.velocity.short"> <condition field="${toll_allow}" expression="domestic"/> <condition field="destination_number" expression="^(\d{10})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/velocity-outbound/$1"/> </condition> </extension> <extension name="domestic.velocity"> <condition field="${toll_allow}" expression="domestic"/> <condition field="destination_number" expression="^(\d{11})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/velocity-outbound$1"/> </condition> </extension> </include> I attempt to place an outside call and I only get a BUSY. I look at the freeswitch.log and I see that the new gateway file is accessed: Action bridge(sofia/gateway/velocity-outbound/<number dialed here>) Later in the log I see (sofia/external-ipv6/<number dialled here>) State Change CS_INIT -> CS_ROUTING Followed by: sofia/external-ipv6/<number dialled here> entering state [calling][0] [DEBUG] sofia.c:6403 Channel sofia/external-ipv6/<number dialled here> entering state [terminated][503] [NOTICE] sofia.c:7286 Hangup sofia/external-ipv6/<number dialled here> [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] [DEBUG] switch_channel.c:3222 Send signal sofia/external-ipv6/<number dialled here> [KILL] This is repeated a few times and finally (sofia/internal/1001@<my FS IP address here>:5060) Locked, Waiting on external entities (sofia/internal/1001@<my FS IP address here>:5060) Ended (sofia/internal/1001@<my FS IP address here>:5060) Running State Change CS_DESTROY (sofia/external-ipv6/<number dialled here>) Ended (sofia/external-ipv6/<number dialled here>) State DESTROY going to sleep (Obviously due to the MANY lines of info in the log, I am only showing a few of them here) I am not yet familiar enough with 'interpreting' what the log is trying to tell me with the exception of: Something Did Not Work. Have I not created and/or configured something wrong in order to get my call out working? Any other suggestions/advice? Thanks, TF

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freeswitch + flashphoner web call server
by jdayola＠spingine.com 02 Mar '15

02 Mar '15

Hi Guys, Have anyone tried using flashphoner web call server? I already have a freeswitch and a flashponer webcallserver running. My problem is how to make freeswitch use the flashphoner webcallserver. Which freeswitch xml file do I need to edit to make use of the flashphoner webcallserver. Im using a flex phone client for my flex desktop application. Do you have any suggestions on the best approach for sip to rtmp gateway? Regards, jdayola

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Mod_shout libfacc support
by Eloy Coto Pereiro 02 Mar '15

02 Mar '15

Hi, I'm trying to play a libfaac[0] files into my Freeswitch, but I can't get it working. Play mp3 files work ok, but encoding with libfacc doesn't work. This is my config in mod_shout: <action application="playback" data="shout://localhost/suite.mp4"/> In the other hand, I tried mod_vlc to play mp4 files, but It didn't work too. I compiled from sources, and tried with debian backports too. <action application="playback" data="vlc://http://localhost/suite.m4a"/> In both cases http request work, reply 200 ok and with data. Play mp3 with both modules work ok. In the other hand, mod_shout and mod_vlc are loaded correctly, and module_exists return always True. Any idea? Is libfaac supported? [0] https://trac.ffmpeg.org/wiki/Encode/AAC#libfaac Regards

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supporting multiple dtmf protocols
by ik 01 Mar '15

01 Mar '15

Hello, I have the misfortune of forced to use a telco named Bezeq - the biggest telco in Israel. The SIP trunk they provide is very problematic. The one that I cannot overcome is that sometimes doing the call, it switches DTMF between Inband and rfc2833, and sometimes the whole DTMF sending is either inband or rfc2833 doing the entire call. I cannot make them to be more stable in this matter (I have tried talking with them), and I cannot replace them (I have tried to do so as well). How can I deal with such mess ? Thanks, Ido

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