- FreeSWITCH-users - FreeSWITCH Mailing Lists

segfaults after 24-36 hours
by Ivan C Myrvold 07 Aug '07

07 Aug '07

My Freeswitch never runs more than 48 hours, and then segmentation faults. Here are the messages before it segfaults: http://pastebin.freeswitch.org/3120 Running version 1.0.5500 on OS X 10.4.10 Ivan

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NI-2 pri support for openzap.
by Michael Jerris 06 Aug '07

06 Aug '07

I have had several people state interest in adding NI-2 national pri support to the pri stack in openzap. Can those interested in sponsoring such work please contact me privately. Thanks Mike

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using Freeswitch to make calls with a sip account
by Sauro Cesaretti 06 Aug '07

06 Aug '07

Hello, I'm a new user here and I'd like to know if it's possible to use freeswitch to make normal calls with one sip account. I'd like to use a console application to make calls with the sip account and I was wondering if freeswitch could work for this task. If yes, how can I configure it? Thanks in advance for all information. Best regards, Sauro

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Re: [Freeswitch-users] Dial
by Anthony Minessale 03 Aug '07

03 Aug '07

<condition field="destination_number" expression="8337"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="somewhere"/> <action application="javascript" data="noanswer.js"/> </condition> This will make the execution stop when a sucessful bridge was over or continue if the attempt to originate a call is not sucessful so if it works it will hangup and if not it will move on. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER: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:213-799-1400 ----- Original Message ---- From: Jonas Gauffin <jonas.gauffin(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, August 2, 2007 1:55:25 PM Subject: Re: [Freeswitch-users] Dial Wont the javascript be called when the destination of the bridge answers? On 8/2/07, Bernhard Suttner <suttner(a)comdasys.com> wrote: > this should work (sleep in milliseconds) > > <condition field="destination_number" expression="8337"> > <action application="bridge" data="somewhere"/> > <action application="sleep" data="20000"/> > <action application="javascript" data="noanswer.js"/> > </condition> > > > Am Donnerstag, den 02.08.2007, 17:52 +0200 schrieb Jonas Gauffin: > > Hello > > > > Is it possible to add a delayed action that only is executed if the > > previous action was not successful? > > > > for instance: > > > > <condition field="destination_number" expression="8337"> > > <action application="bridge" data="somewhere"/> > > <action application="javascript" delay="20" data="noanswer.js"/> > > </condition> > > > > //Jonas > > > > _______________________________________________ > > 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 > _______________________________________________ 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 ____________________________________________________________________________________ Need a vacation? Get great deals to amazing places on Yahoo! Travel. http://travel.yahoo.com/

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Dial
by Jonas Gauffin 02 Aug '07

02 Aug '07

Hello Is it possible to add a delayed action that only is executed if the previous action was not successful? for instance: <condition field="destination_number" expression="8337"> <action application="bridge" data="somewhere"/> <action application="javascript" delay="20" data="noanswer.js"/> </condition> //Jonas

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DNS Error
by Dinesh Dialani 02 Aug '07

02 Aug '07

Hi, I am trying to make call through enum lookup and using bind 9.4.1 as a dns server. My zone file look like this : $ORIGIN . $TTL 21600 ; 6 hours 91.e164.arpa IN SOA ns1.91.e164.arpa. hostmaster.ns1.91.e164.arpa. ( 2005022250 ; serial 10800 ; refresh (3 hours) 3600 ; retry (1 hour) 604800 ; expire (1 week) 86400 ; minimum (1 day) ) NS ns1.91.e164.arpa. A 192.168.96.57 $TTL 60 ; 1 minute NAPTR 100 10 "u" "SIP+E2U" "!^.*$!sip:100@192.168.96.122!i" . $ORIGIN 91.e164.arpa. 1 NAPTR 100 10 "u" "E2U+SIP" "!^.*$!sip:100@192.168.96.122!i" . athir A 192.168.96.57 here i have two xlite phone register with freeswitch and i want to make call between them using enum. I have given name to them as 1 and 2 now i want to call to 1 using 2. Freeswitch give me error dns error and do not resolve 1.91.e164.arpa. Even when i test the naptr record using nslookup utility of bind it give me right answer but freeswitch fail to resolve the address. Thanks Yogesh

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Compilation errors win32
by Jonas Gauffin 02 Aug '07

02 Aug '07

I get the following errors/warnings with the latest trunk. 1>switch_ivr_play_say.c 1>..\..\src\switch_ivr_play_say.c(1173) : error C2039: 'samples' : is not a member of 'switch_speech_handle' 1> c:\projects\cpp\FreeSwitch\src\include\switch_module_interfaces.h(358) : see declaration of 'switch_speech_handle' 1>..\..\src\switch_ivr_play_say.c(1384) : error C2039: 'samples' : is not a member of 'switch_speech_handle' 1> c:\projects\cpp\FreeSwitch\src\include\switch_module_interfaces.h(358) : see declaration of 'switch_speech_handle' 1>..\..\src\switch_ivr_play_say.c(1385) : error C2039: 'samples' : is not a member of 'switch_speech_handle' 1> c:\projects\cpp\FreeSwitch\src\include\switch_module_interfaces.h(358) : see declaration of 'switch_speech_handle' 1>..\..\src\switch_ivr_play_say.c(1403) : error C2039: 'samples' : is not a member of 'switch_speech_handle' 1> c:\projects\cpp\FreeSwitch\src\include\switch_module_interfaces.h(358) : see declaration of 'switch_speech_handle' 1>..\..\src\switch_ivr_play_say.c(1403) : warning C4047: 'function' : 'int' differs in levels of indirection from 'switch_memory_pool_t *' 1>..\..\src\switch_ivr_play_say.c(1403) : warning C4024: 'switch_core_timer_init' : different types for formal and actual parameter 4 1>..\..\src\switch_ivr_play_say.c(1403) : error C2198: 'switch_core_timer_init' : too few arguments for call 1>switch_xml.c 1>..\..\src\switch_xml.c(68) : fatal error C1083: Cannot open include file: 'dirent.h': No such file or directory

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Re: [Freeswitch-users] ringnack does not working
by Anthony Minessale 02 Aug '07

02 Aug '07

ring_ready is a dialplan application that sends the protocol specific message to indicate ringing. In the case of SIP a 180 Ringing. ringback is a channel variable you can set to artificially generate a tone or play an audio file to an originating channel while it waits to be connected to another call. pre_answer is a dialplan application that will send the protocol specific message to indicate early media. In the case of SIP a 183 Progress So, if you call into freeswitch with sip and the first entry in your dialplan is ring_ready followed by a bridge to some other destination when it hits ring_ready it will send "180 Ringing" back to your phone so it can generate the ringing sound. If instead you use the set application to set ringback to a tone spec or audio file followed by a call to pre_answer to establish an early media connection followed by a call to bridge to another dest, then the core will generate this audio locally and send it back to your phone. Again, this occurs during early media meaning the call has not been answered but the SIP has negotiated a media path in advance for this type of pre-answer audio indication. SIP has no promise that early media must be supported so some switches and devices opt to not support it meaning you may not be able to hear any audio until the call is officially answered which would keep you from hearing the artificial ringback at all. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER: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:213-799-1400 ----- Original Message ---- From: Farrukh Ahmed <f4fahmed(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, August 2, 2007 4:35:20 AM Subject: [Freeswitch-users] ringnack does not working Dear Sir, I was trying to get ringback working but its not working. I used exactly what docs said. can some one tell me what does it means http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready ? i could not find any thing regarding this on fs example config files but its on wiki my FS Configuration File FYI, http://pastebin.freeswitch.org/3155 Best Regards. -- Färrükh Ähmëd ____________________________________________________________________________________ Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mail&p=graduation+gifts&cs=bz

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Status of OpenZap and Digium cards?
by Robert La Ferla 02 Aug '07

02 Aug '07

Anyone get a TDM card working with OpenZap and FreeSwitch? Some basic instructions would be valuable...

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Broadvoice config?
by Robert La Ferla 02 Aug '07

02 Aug '07

Anyone have Broadvoice working w/FS? If so, how do you configure it? I tried this but it doesn' work (see error below) <gateway name="sip.broadvoice.com">  <param name="username" value="8885551212"/>      <param name="password" value="SECRET"/>               </gateway> ERRORS 2007-07-22 00:35:40 [NOTICE] sofia_reg.c:78 sofia_reg_check_gateway() registering sip.broadvoice.com 2007-07-22 00:35:40 [DEBUG] sofia.c:71 sofia_event_callback() event [nua_r_register] status [503][DNS Error] session: n/a 2007-07-22 00:35:40 [WARNING] sofia_reg.c:569 sofia_reg_handle_sip_r_register() Registration Failed with status 503

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