March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How does mod_directory get users information?
by Robert Hadley 19 Mar '10

19 Mar '10

Hello, Where does mod_directory get the user information it uses to put into the database in creates? Thanks, Robert

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Re: [Freeswitch-users] rtp timeout issue
by Madovsky 19 Mar '10

19 Mar '10

Found the iptables lines that caused problem : #### #Prevent external packets from using loopback -A INPUT -i br0 -s 127.0.0.1 -j DROP -A FORWARD -i br0 -s 127.0.0.1 -j DROP -A INPUT -i br0 -d 127.0.0.1 -j DROP -A FORWARD -i br0 -d 127.0.0.1 -j DROP ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, March 19, 2010 1:53 PM Subject: Re: [Freeswitch-users] rtp timeout issue bingo.... :), grrrr, now I need to read all my iptables :( ----- Original Message ----- From: Anthony Minessale To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, March 19, 2010 1:44 PM Subject: Re: [Freeswitch-users] rtp timeout issue try stopping iptables for 1 call to make sure it is not related to that. On Fri, Mar 19, 2010 at 12:29 PM, Madovsky <infos(a)madovsky.org> wrote: yes I did it also. the issue is only noticeable if I use 2 softphones on the same public IP where there is FS. I tried from another PC (even through nat) and it works fine... Arrgh, I'm lost ----- Original Message ----- From: Anthony Minessale To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, March 19, 2010 1:17 PM Subject: Re: [Freeswitch-users] rtp timeout issue you should also unblock the rtp port range 16384-32768 udp On Fri, Mar 19, 2010 at 12:05 PM, Madovsky <infos(a)madovsky.org> wrote: with bypass-media=true everything is ok, but I need to pass media through FS, I missed some settings but don't know where... I use default configs, inverted the internal and external port and commented out all nat params. Thanks ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, March 19, 2010 12:29 PM Subject: rtp timeout issue I set FS internal on port 5080 and external 5060. 2 softphones on public IP, so no nat. the sip invite is ok, but there's a media timeout.. port 5080 and 5060 are opened on my firewall must I set any nat params anyway ? Thanks Franck _______________________________________________ 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 -- 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 iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 ------------------------------------------------------------------------ _______________________________________________ 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 _______________________________________________ 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 -- 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 iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 ---------------------------------------------------------------------------- _______________________________________________ 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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Re: [Freeswitch-users] rtp timeout issue
by Madovsky 19 Mar '10

19 Mar '10

with bypass-media=true everything is ok, but I need to pass media through FS, I missed some settings but don't know where... I use default configs, inverted the internal and external port and commented out all nat params. Thanks ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, March 19, 2010 12:29 PM Subject: rtp timeout issue I set FS internal on port 5080 and external 5060. 2 softphones on public IP, so no nat. the sip invite is ok, but there's a media timeout.. port 5080 and 5060 are opened on my firewall must I set any nat params anyway ? Thanks Franck

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rtp timeout issue
by Madovsky 19 Mar '10

19 Mar '10

I set FS internal on port 5080 and external 5060. 2 softphones on public IP, so no nat. the sip invite is ok, but there's a media timeout.. port 5080 and 5060 are opened on my firewall must I set any nat params anyway ? Thanks Franck

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FS Best practices
by Todd 19 Mar '10

19 Mar '10

Hey List, still kinda new to this whole deal, looking for some advice. We are building a website that needs to interact with FS. Not looking for specific instructions just opinions about the best practices for doing things like click to dial from a web page, calling CDR records to web applications to display customers stats and general web integration. Does this all need to be done using scripting languages like Lua or Perl, or can you directly integrate FS with a website in some other manner. Again, just getting into all this, but having a blast learning FS, config and power rock! Anyhow, some places or things I should start looking into would be greatly appreciated. Thanks List!

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make errors - mod_cepstral
by Aloysius Lloyd 19 Mar '10

19 Mar '10

Hi All, I am trying to enable flite & cepstral on a existing FreeSWITCH installation. I follow the following steps 1. build directory cp modules.con.in modules.conf 2. edit modules.conf remove the comments from the following asr_tts/mod_flite asr_tts/mod_cepstral 3../bootstrap.sh 4. ./configure 5. make I get the following error make[5]: *** [mod_cepstral.lo] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_cepstral-all] Error 1 make[2]: *** [all-recursive] Error 1 How can I successfully enable the mod_cepstral and flite. Thanks Lloyd

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Newbie problem with Multiple Tenants and mod_xml_curl
by lloyd thomas 19 Mar '10

19 Mar '10

I eventually managed to get mod_xml_curl working for me. Now I trying to be a bit cleverer setting up a multi-tenancy using mod_xml_curl based on the wiki instrucions http://wiki.freeswitch.org/wiki/Multiple_Companies. But I get an error when a phone tries to register. 2010-03-18 21:19:04.869688 [WARNING] sofia_reg.c:1853 Can't find user [ 203(a)192.168.101.102] You must define a domain called '192.168.101.102' in your directory and add a user with the id="203" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2010-03-18 21:19:04.885327 [WARNING] sofia_reg.c:1018 SIP auth failure (REGISTER) on sofia profile 'internal' for [203(a)192.168.101.102] from ip 192.168.91.28 I think I am nearly there,but I suspect there is something wrong with the resulting 'user directory' xml. --------------------------------------------------------------- <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory" description="User Directory"> <domain name="default"> <groups> <group name="default"> <users> <user id="203"> <params> <param name="password" value="$${default_password}"/> <param name="dial-string" value="{presence_id=${dialed_user}@ ${dialed_domain}\}${sofia_contact(${dialed_user}@${dialed_domain})}"/> <param name="vm-password" value="1234"/> <param name="vm-disk-quota" value="30"/> </params> <variables> <variable name="user_context" value="default"/> <variable name="accountcode" value="203"/> <variable name="day_bar" value="3"/> <variable name="night_bar" value="3"/> <variable name="effective_caller_id_name" value="Extension 203"/> <variable name="effective_caller_id_number" value="203"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_number}"/> </variables> </user> </users> </group> </groups> </domain> </section> </document> -------------------------------------------------------------------- I think the Company Directory XML is OK -------------------------------------------------------------------- <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory" description="Company Directory"> <domain name="default"> <params> <param name="dial-string" value="{presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> <variables> <variable name="record_stereo" value="true"/> <variable name="default_gateway" value="$${default_provider}"/> <variable name="default_areacode" value="$${default_areacode}"/> <variable name="transfer_fallback_extension" value="operator"/> <variable name="user_context" value="default"/> </variables> </domain> <domain name="phisystems.dom"> <params> <param name="dial-string" value="{presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> <variables> <variable name="record_stereo" value="true"/> <variable name="default_gateway" value="$${default_provider}"/> <variable name="default_areacode" value="$${default_areacode}"/> <variable name="transfer_fallback_extension" value="operator"/> <variable name="user_context" value="phisystems.dom"/> </variables> </domain> </section> </document> -------------------------------------------------------------- I get the following result when I 'sofia status' Name Type Data State ================================================================================================= internal profile sip:mod_sofia@192.168.101.102:5060 RUNNING (0) phisystems.dom alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) default alias internal ALIASED external profile sip:mod_sofia@192.168.101.102:5080 RUNNING (0) ================================================================================================= Thanks in advance

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ProxyMedia Unable to hear voice
by lakshmanan ganapathy 19 Mar '10

19 Mar '10

Hi all, I've updated to the latest SVN 17048. Here is my dialplan: <include> <context name="default"> <extension name="outbound_soc"> <condition field="destination_number" expression="^.*$"> <action application="ring_ready" /> <action application="set" data="continue_on_fail=true"/> <action application="set" data="bypass_media=true"/> <action application="set" data="ignore_early_media=true"/> <action application="set" data="exec_after_bridge_app=park"/> <action application="socket" data=" 0.0.0.0:8447 async full"/> </condition> </extension> </context> </include> >From a nc session I issued the following commands. connect sendmsg call-command: execute execute-app-name: answer sendmsg call-command: execute execute-app-name: bridge execute-app-arg: user/1010 I (user/1001) is able to hear the voice of the bridged party (user/1010). But user/1010 is unable to hear my voice. If I set the bypass_media=false, then both can hear the voice. I wanted to know why when bypass_media=true, the user/1010 cannot hear my voice. Here is the freeswitch log: http://pastebin.freeswitch.org/12484

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Video pass problem
by Mikhail Krivushin 19 Mar '10

19 Mar '10

Hello! We have problem with pass video over FreeSWITCH. I tshark traf, and see that we have 1280 video packets input, and only 560 passed to B leg. Anyone can point me to right direction? I can send pcap file by request. Most time I have black screen, and then one frame can appear, and I see frozen picture pair minutes, and then other frozen picture. We have not network issues, we have good video, when bypass FS. We have not perfomance troubles to. We have ubuntu 9.10 x64, and powerfull server board. uname: Linux fs 2.6.31-17-generic #54-Ubuntu SMP Thu Dec 10 17:01:44 UTC 2009 x86_64 GNU/Linux config: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=BV32,BV16,speex@8000h @20i,speex@16000h@20i,speex@32000h @20i,PCMU,PCMA,GSM,H264,H263,H263-1998,H263-2000,H261"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=BV32,BV16,speex@8000h @20i,speex@16000h@20i,speex@32000h @20i,PCMU,PCMA,GSM,H264,H263,H263-1998,H263-2000,H261"/> -- С уважением, Кривушин Михаил г. Томск сот. +7 913 865 78 66 icq: 218 744 127 xmpp: KrivushinME(a)jabber.ru skype: mkrivushin

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Caller ID w/ SLA/SCA
by Gabriel Kuri 19 Mar '10

19 Mar '10

Hey guys, is the Caller ID data (number and name) supposed to be passed between the phones when using SLA/SCA ? If you put a call on hold on phone 1, and pick up the call on phone 2, number/name do not show up. Likewise, if you go back to the same call on phone 1, the name and number are not present. Is this normal? Do I smell a bounty ;) ? Cheers, Gabe

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