May 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Recording all calls with good quality.
by Евгений 10 May '12

10 May '12

Hi. I see you are recording in stereo. Both channels has horrible quality? I had horrible quality in one channel when it was two interfaces looking in the same network. Evginey I've tried the following codec settings, set in vars.xml. The first and third lines are the defaults upon install. <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/>  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/> My CPU load shows fine, even with 2 clients connected to an external (secondary) switch, through this one. I truly appreciate the help. Brandon McGinty-Carroll On Sun, May 06, 2012 at 06:50:44PM +0300, Avi Marcus wrote: > AFAIK, gsm is lossy, which is why it doesn't sound good. > wav, however, should sound great. What codec are your calls in? Are you > recording many calls? > Check the load average while you are recording. If it's greater than the > number of CPUs then probably your hard drives can't keep up. There are > several options then, first let see if we can diagnose the issue. > > -Avi > > > On Sun, May 6, 2012 at 5:46 PM, <brandon.mcginty(a)gmail.com> wrote: > > > Good day. > > I'm trying to record all calls sent through my Freeswitch PBX. > > The problem is that the quality is horrible. I've tried wav and GSM > > formats, and both are les than satisfactory. > > I'm looking for something around the quality of the recent recordings of > > the Freeswitch conference. (I checked on the wiki, and couldn't find > > anything through search.) > > Any information on the FS conference format or my configuration would be > > appreciated. > > I've got: > >  > > <condition field="destination_number" expression="^(.+)$" break="never"> > > <action application="set" data="RECORD_TITLE=Recording > > ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> > > <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> > > <action application="set" data="RECORD_STEREO=true"/> > > <action application="record_session" > > data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > > </condition> > > > > Sincerely, > > Brandon McGinty-Carroll > > > > > > _________________________________________________________________________ > > 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 > > > _________________________________________________________________________ > 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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Call intercept
by Miha 10 May '12

10 May '12

Hi, I need a little help about call intercept. I have read wiki (http://wiki.freeswitch.org/wiki/Callgroup_intercept) but still having few problems. 1. I have Opensips which works like load_balancer and registrar. Behind Opensips I have two FS server, which are for load_balacing. 2. I need to set a call intercept thing. I done it like it is written on FS wiki but this does not work as I have Opensips. I have put this in public dialplan: <action application="set" data="called_party_callgroup=${user_data(${dialed_ext}@${domain} var callgroup)}"/> <action application="db" data="insert/last_dial/${called_party_callgroup}/${uuid}"/> After I try to intercept the call this fails as in database is written different UUID. CAll goes like this: Phone-openips-FS-Openips-Phone Thank you for all your help! MIha

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User-specific gateways
by Lesley Pervis 09 May '12

09 May '12

Is this supposed to work? For me, when the user registers, the gateway does not register. There's no indication in either the FreeSWITCH or sofia logs that an attempt is even contemplated. http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#User-Specific_Gate…

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Re: [Freeswitch-users] freeswitch and voipstunt
by Fernando Di Nucci 09 May '12

09 May '12

That is to say I'm hopeless ? :-(

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mod_sms send() not working as expected
by John Gathm 09 May '12

09 May '12

Hi, I am trying to make mod_sms work, it seems that with the configuration I give to freeswitch, it does not forward sms (though it might not be related to mod_sms but perhaps chatplan ?) works : -two extentions can call each other -the default example chatplan works with the autoreply ( Action reply(Hello, you said: ${_body} ), I receive the response SMS. however, when I just wan to use the "send" action ( <action application="send"/> ) it does not work, (*.*.* in ip's are only replaced by me in the mail for privacy, are full and correct in logs) 2012-05-08 23:15:38.784507 [INFO] mod_sms.c:300 Processing text message 1234->1005 in context default 2012-05-08 23:15:38.784507 [CONSOLE] mod_xml_curl.c:318 XML response is in /tmp/b953861e-9963-11e1-b83f-5d2cc73f3eee.tmp.xml Chatplan: 1005 parsing [default->demo] continue=false Chatplan: 1005(a)176.*.*.* Regex (PASS) [demo] to(1005(a)176.*.*.*) =~ /^(.*)$/ break=on-false Chatplan: 1005(a)176.*.*.* Action send() 2012-05-08 23:15:38.784507 [DEBUG] sofia_presence.c:208 Can't find registered user 1005(a)176.*.*.* 2012-05-08 23:15:38.784507 [WARNING] sofia_presence.c:4021 Not sending message to ourselves! although my clients are registered properly (and can call each other) 2012-05-08 22:44:20.944505 [WARNING] sofia_reg.c:1442 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1234(a)176.*.*.*] from ip 78.*.*.* 2012-05-08 22:44:21.165051 [CONSOLE] mod_xml_curl.c:318 XML response is in /tmp/5a2cb204-995f-11e1-b7fb-5d2cc73f3eee.tmp.xml 2012-05-08 22:44:21.184514 [CONSOLE] mod_xml_curl.c:318 XML response is in /tmp/5a300bc0-995f-11e1-b7fc-5d2cc73f3eee.tmp.xml 2012-05-08 22:44:21.244503 [WARNING] sofia_reg.c:1442 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1005(a)176.*.*.*] from ip 78.*.*.* 2012-05-08 22:44:21.324503 [CONSOLE] mod_xml_curl.c:318 XML response is in /tmp/5a463dd2-995f-11e1-b7fe-5d2cc73f3eee.tmp.xml 2012-05-08 22:44:21.384540 [CONSOLE] mod_xml_curl.c:318 XML response is in /tmp/5a4ff886-995f-11e1-b7ff-5d2cc73f3eee.tmp.xml trying to send a sms to myself does not work either, I also get "can't find registered user " even when I'm the sender and logged. Thanks in advance for any help. Regards, J.

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Re: [Freeswitch-users] freeswitch and voipstunt
by Fernando Di Nucci 09 May '12

09 May '12

freeswitch@internal> sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia@192.168.30.30:5080 RUNNING (0) external::voipstunt gateway sip:*********@sip.voipstunt.com TRYING (retry: 46s) external::pstn gateway sip:81@192.168.30.176:5062 NOREG external::Messagenet gateway sip:*******@sip.messagenet.it:5061 REGED 192.168.30.30 alias internal ALIASED internal profile sip:mod_sofia@192.168.30.30:5060 RUNNING (0) ================================================================================================= 2 profiles 1 alias pstn is a ht-503 (working) and also messagenet is working. @Michael Collins: move the gateway to the internal profile means putting the file in internal folder ? Please be patient ! I tried to register X-lite 4 but no luck, the only registration that works seems their own softphone...

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Re: [Freeswitch-users] freeswitch and voipstunt
by Fernando Di Nucci 09 May '12

09 May '12

First of all thank you for your time. According to your suggestions I tried: <include> <gateway name="voipstunt"> <param name="username" value="*********"/> <param name="password" value="*********"/> <param name="realm" value="voipstunt.com"/> <param name="proxy" value="sip.voipstunt.com"/> <param name="expire-seconds" value="800"/> <param name="register" value="true"/> <param name="retry-seconds" value="60"/> <param name="caller-id-in-from" value="false"/> <param name="context" value="public"/> </gateway> </include> after issuing at fs_cli console: /debug 9 sofia profile external siptrace on I get ------------------------------------------------------------------------ 2012-05-09 19:18:39.041881 [WARNING] sofia_reg.c:405 Timeout Registering voipstunt 2012-05-09 19:18:40.041894 [WARNING] sofia_reg.c:429 voipstunt Failed Registration [0], setting retry to 60 seconds. send 645 bytes to udp/[77.72.169.134]:5060 at 17:18:40.065594: ------------------------------------------------------------------------ REGISTER sip:sip.voipstunt.com;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.30.30:5080;rport;branch=z9hG4bK2XryvegBXNejD Max-Forwards: 70 From: <sip:*********@sip.voipstunt.com;transport=udp>;tag=2crj3HatDt40g To: <sip:********@sip.voipstunt.com;transport=udp> Call-ID: 35f3320b-4bd0-414a-8324-0a98c411f6c6 CSeq: 27955864 REGISTER Contact: <sip:gw+voipstunt@192.168.30.30:5080;transport=udp;gw=voipstunt> Expires: 800 User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ repeated many times. I'm new to freeswitch, so if a have to do something different, please let me know. Thank you

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freeswitch and voipstunt
by Fernando Di Nucci 09 May '12

09 May '12

Hi all, I have an account at voipstunt. Everything's fine using their softphone, but in their website http://www.voipstunt.com/en/sipp.html they claim it is possible to use their service from a sip device. I tried this in freeswitch: <include> <gateway name="voipstunt"> <param name="username" value="*********"/> <param name="password" value="*********"/> <param name="proxy" value="sip.voipstunt.com:5060"/> <param name="realm" value="sip.voipstunt.com:5060"/> <param name="register-transport" value="udp"/> <param name="expire-seconds" value="800"/> <param name="register" value="true"/> <param name="retry-seconds" value="60"/> <param name="context" value="public"/> </gateway> </include> but freeswitch can't register. Any help appreciated, thank you

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No ESL events received
by Bernard Fluixa 09 May '12

09 May '12

Hi, I am running an ESL C program that collects Freeswitch events in a thread (esl_thread_create_detached(EventCollectorThread, switch_handle) ) following an "originate" command with the "return_ring_ready=true" option set, run by another process. When I call a soft phone, I receive all events (after the esl_events(switch_handle, ESL_EVENT_TYPE_PLAIN, "ALL") execution) and everything works as expected. However, that same program does NOT receive any Freeswitch events when I call a real phone number through my VOIP provider (flowroute) although Freeswitch console shows that the call has been answered (Channel [sofia/external/+19999999999] has been answered). What am I missing? Thank you Bernard

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Freeswitch And 2 Network I/Fs
by Jerry Richards 09 May '12

09 May '12

Is there an example of configuring Freeswitch to support both eth0 and eth1, when they are connected to different networks? I know I can create two sip_profiles, which works fine for registration, but in the past, I've had issues with Freeswitch not sending media to the right interface (or at least, not knowing how to control that). Thanks, Jerry

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