June 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Call Hangup
by Alessandro 20 Jun '11

20 Jun '11

Hi, I have an active call from the extension A to the extension B. If one of the two extension is disconnected from the network without deregistration the call isn't hung up by the server. The call should be hangup when FS realizes that the extension isn't in the network? Is there a time out to be set somewhere? thanks Regards Alessandro Luppi -- Ing. Alessandro Luppi Software development Seletech srl Via Collodi 8, 20052 Monza (MI) - Italy Tel: +39.039.5962000 - Fax: +39.039.9716905 email: a.luppi(a)seletech.com - Web: www.seletech.com or www.seletech.eu

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Can FS use external user database
by fieldpeak 20 Jun '11

20 Jun '11

Gurus, Can we configure FS to have it to read registeration users information (e.g. account and pwd) for authentication from external database replacing default xml files(under conf\directory\default)? e.g. we have a database which contains user's account and password and was used by other application, we would like to have FS to read this exsiting databse for authentication as well... Thanks. Regards, Charles

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No Voice for default IVR ext 5000
by ankIT WALiA 20 Jun '11

20 Jun '11

Dear all, I have configured an gateway for external profile to receive and make calls. I have added a file in public directory. <include> <extension name="iptel-inbound"> <condition field="destination_number" expression="^(MY_IPTEL_USERNAME)$"> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="ExtNo XML default"/> </condition> </extension> </include> Now, while I am calling to my voip account, I am transferring the call. It is getting transferred to the extension also. But, If I use normal ext number, I am able to talk both ways. But when I use default ext number 5000 for IVR menu. I could not listen to IVR menu. Though, I can see in my logs that the audio file are being played one by one, but I cant listen. Do I need some extra configuration to listen to IVR? There may be a gap in my knowledge. Please enlighten. Thanks Ankit -- Life is like a rose its upto u feel it as its fragrance or thorns

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mod_pocketsphinx
by roy 19 Jun '11

19 Jun '11

Hi, Trying to get mod_pocketsphinx working from the current git tree, FreeSWITCH version: 1.0.head (git-2629a57 2010-06-16 22-42-15 +0200) however, I got this error during startup: <snip> 2010-06-17 11:18:08.032645 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_pocketsphinx.so **/usr/local/freeswitch/mod/mod_pocketsphinx.so: undefined symbol: ngram_model_get_counts** </snip> Lib paths were already defined in /etc/ld.so.conf as: include /usr/local/freeswitch/lib /usr/local/freeswitch/mod # ls -al mod/mod_pocketsphinx.* -rwxr-xr-x 1 root staff 1467 Jun 17 10:49 mod/mod_pocketsphinx.la -rwxr-xr-x 1 root staff 905769 Jun 17 10:49 mod/mod_pocketsphinx.so Was wondering if someone was able to get this working. Manually downloading the current pocketsphinx and sphinxbase and renaming them into 0.4.99 and rebuilding didn't help either. Would appreciate if anyone can point me into the right direction. TIA. Roy

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No more sessions allowed at this time.
by Henk Oegema 18 Jun '11

18 Jun '11

My FS server was working fine until (suddenly) now. When I want to call a mobile phone (via a gateway) I get busy (?) or unavailable (?) and get this message: freeswitch@internal> 2011-06-18 19:23:00.379259 [CRIT] sofia.c:841 No more sessions allowed at this time. There is only one user on my system. (myself) Rgds. Henk.

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FreeSWITCH Cookbook Need: Fax Examples
by Michael Collins 18 Jun '11

18 Jun '11

Hey all, I am working on the FSCB and I could use some of your real-world fax examples. I'm looking for anything like these: Inbound fax to email/PDF Outbound fax examples (scripts, etc. - whatever you do to make it easier for your users to send faxes thru FS) T.38 config examples and reports on symptoms/issues you had to overcome to make it work I will gladly accept any information you have, even if it's just raw dumps of scripts, XML configs, etc. Shoot them to me off-list. Thanks! -MC

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using keypress to stop a recording.
by Wes 18 Jun '11

18 Jun '11

I'm using the following script to record, and I"m hoping to take a user keypress to trigger the end of the recording and then play it back: (the problem is that it stops on ANY keypress, while it seems like it should only break on a "#" keypress) I found this example at: http://wiki.freeswitch.org/wiki/Mod_lua#session:recordFile local numberToCall = 1234 local session = freeswitch.Session("sofia/xxx.xxx.xxx.xxx/"..numberToCall); session:set_tts_parms("flite", "kal"); session:speak("Thank you for using my recording service, press the pound key to stop the recording." ); function onInput(s, type, obj) if (type == "dtmf" and obj['digit'] == '#') then return "break"; end end session:setInputCallback("onInput", ""); session:recordFile("/tmp/luatest.wav"); freeswitch.consoleLog("info", "recording stopped by user keypress \n"); session:speak("your voicehas been recorded, i will play it for you now"); session:streamFile("/tmp/luatest.wav"); session:speak("that's it, goodbye."); freeswitch.consoleLog("info", "hanging up... \n"); session:hangup();

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Help With ClueCon 2011
by Michael Collins 18 Jun '11

18 Jun '11

Hey gang! I am looking to get a few notable names to ClueCon this year and am looking for businesses who are willing to sponsor by purchasing plane tickets or hotel rooms. If your business is in a position to assist then please contact me off list and I will give you the details. Also, if you haven't already signed up for ClueCon then maybe you'd like to call in via our new RTMP phone <http://conference.freeswitch.org/cc2/>! Brian would love to hear from you. :) Thanks, Michael

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Registration problem with default extension
by abderrahmane abdmeziane 17 Jun '11

17 Jun '11

hey,after installing freeswitch from deb package,and using default extension 1000 with default password 1234 in xlite softphone,i can't call another extension just don't ring but in xlite screen i get my user name with my number;a check the output of F9 from fs_cli command,but there is no register extension: ############################################## freeswitch@internal> sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT true DBName sofia_reg_internal Pres Hosts 192.168.2.2,192.168.2.2 Dialplan XML Context public Challenge Realm auto_from RTP-IP 192.168.2.2 Ext-RTP-IP 41.104.52.52 SIP-IP 192.168.2.2 Ext-SIP-IP 41.104.52.52 URL sip:mod_sofia@192.168.2.2:5060 BIND-URL sip:mod_sofia@192.168.2.2:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM CODECS OUT G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Total items returned: 0 ================================================================================================= freeswitch@internal> ################################# any help please?

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Barracuda Networks Donates mod_rtmp for FreeSWITCH (Flash Audio Endpoint)
by Anthony Minessale 17 Jun '11

17 Jun '11

check it out http://slashdot.org/submission/1657690/Barracuda-Networks-Connects-FLASH-To… mark it worthy and rate it http://digg.com/news/entertainment/barracuda_networks_connects_flash_to_sip… digg it -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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