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Re: [Freeswitch-users] unable to establish call to pstn
by Anthony Minessale 05 Apr '07

05 Apr '07

Did you use the syntax: sofia/gateway/<name of gateway>/<destination uri> 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: Helmut Kuper <helmut.kuper(a)ewetel.de> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, April 5, 2007 3:26:30 AM Subject: Re: [Freeswitch-users] unable to establish call to pstn -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I compiled and installed rev. 4853. Unfortunately there is no change in INVITE message. There is still "sip:mod_sofia@212.6.123.118:5060" in invites contact field. regards helmut Anthony Minessale schrieb: > try svn revision 4837 > > I added code to make the contact be the same as the register when > using a gateway for an invite. > > 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 > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGFLKI4tZeNddg3dwRAsKsAJ9eb2m7YUMusT1/yv0yGeacPzo/jwCffLpz 0jwZu/77gONdByzFsaCpM6o= =XfiE -----END PGP SIGNATURE----- _______________________________________________ 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 ____________________________________________________________________________________ Get your own web address. Have a HUGE year through Yahoo! Small Business. http://smallbusiness.yahoo.com/domains/?p=BESTDEAL

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OSP Users and Developers
by Brian West 05 Apr '07

05 Apr '07

Is their anyone out here on the list that would be interested in a consulting job related to OSP and FreeSWITCH? Contact me off List. Thanks, Brian West

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Requesting quote
by EdPimentl 05 Apr '07

05 Apr '07

We are looking for a quote for a FS project with the following requirements: FreeSwitch Dial Plan: Support Sip Trunking (In-Out) with 3rd party SIP Provider Support Asterisk via AIX and SIP Support VoiceMail with individual and group Support Conferences - Create Adhoc and scheduled and assigned PIN , (creation of conf call via call back feature, ala jajah) Support Accepting SIP URI(one or more), e164 number(one or more) and establish a call ( another call back feature) Support Accepting 1 or more e164 numbers and Conference call pin and establish the call Support dialing out one or more number and at a given date and time and establishing a conference call Support use a schedule to know when to dial out and deliver a message(log if it is ring no answer,, busy or AMD full and call back) Support Integration with OpenFire for GoogleTalk - XMPP interoperability Support unified communications rules and determine how to route call based upon Support ITAD / ISN Routing Support Lease Cost Routing -------------------------------------- Please note this Asterisk syntax.. and it will require a equivalent FS dial plan.. S= the Mobile# or ID of person and "N" is the priority to parse the rule.. AYA_EVENT = anywhereyouare event exten => s,n,AYA_CAS(${ARG3},${CALLERIDNUM},${CALLERIDNAME},${EMAILID},${GTALKUSERID}) exten => s,n,Verbose( AYA_EVENT is ${AYA_EVENT}) exten => s,n,GotoIf($[${AYA_EVENT} = ACCEPT ]?accept:) exten => s,n,GotoIf($[${AYA_EVENT} = DECLINE ]?decline:) exten => s,n,GotoIf($[${AYA_EVENT} = REDIRECT ]?redirect:) exten => s,n,GotoIf($[${AYA_EVENT} = VMAIL ]?voicemail:) exten => s,n,GotoIf($[${AYA_EVENT} = CONFERENCE ]?conference:) exten => s,n,GotoIf($[${AYA_EVENT} = SMS ]?sms:) exten => s,n,GotoIf($[${AYA_EVENT} = PRESENCE ]?IM:) Thanks in advance, -E

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Re: [Freeswitch-users] unable to establish call to pstn
by Anthony Minessale 05 Apr '07

05 Apr '07

try svn revision 4837 I added code to make the contact be the same as the register when using a gateway for an invite. 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: Helmut Kuper <helmut.kuper(a)ewetel.de> To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, April 4, 2007 8:45:15 AM Subject: [Freeswitch-users] unable to establish call to pstn -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I use freeswitch as a gateway to PSTN. I'm not able anymore to establish calls from softphone registered to freeswitch and PSTN extension since sofia config changed from registrations to gateways. The registration works fine with PSTN gateway provider (at least I get an OK message). But when I try to call a PSTN extension, I got first a "Proxy Authentication Request" message, then a DECLINE message from PSTN GW. When I compare the REGISTER Message from freeswitch to PSTN GW with the INVITE message, I found that the contact in message header are not the same. In register it is "sip:032223930016@212.6.123.118:5060", in invite it is "sip:mod_sofia@212.6.123.118:5060" (see register-contact.png and invite-contact.png). I think this is the reason why. is there a way in sofia config to change this? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGE6vr4tZeNddg3dwRAlW/AJ49wbETmQvcm0/yKimEHCmdaZ2OMwCggDWP 7dQWRFC0uKoUd/sObG6PgQw= =Uwau -----END PGP SIGNATURE----- _______________________________________________ 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 ____________________________________________________________________________________ Get your own web address. Have a HUGE year through Yahoo! Small Business. http://smallbusiness.yahoo.com/domains/?p=BESTDEAL

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Compile Freeswitch on OS X
by Ivan C Myrvold 04 Apr '07

04 Apr '07

How can I build freeswitch on OS X now? A few weeks back i could svn checkout a new version and run ./configure etc, but now I have to run ./bootstrap, and my MacBook Pro complains about the automake version. Ivan

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Re: [Freeswitch-users] session.originate question
by Anthony Minessale 04 Apr '07

04 Apr '07

It's all the same as when you do it in the dialplan. If you set the ringback channel var to the name of a file or a ttml script it will play it while the B leg originates providing that the inbound leg is passed to the new session's originate method: session.setVariable("ringback", "%(2000, 4000, 440.0, 480.0)"); //or //session.setVariable("ringback", "/path/to/proper_sample_rate.wav"); new_session = new Session(); new_session.originate(session, "sofia/foo(a)bar.com"); 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: Tamas Cseke <cstomi.levlist(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, April 4, 2007 4:29:38 AM Subject: [Freeswitch-users] session.originate question Hello, I'd like to make an ivr, what ask for a number, and call it. I try to write a js script. As i see, the new session originate method return when the called party answer the call. I need to play a soundfile back to the caller on 180 reply, what inform the call is ringing. Is it possible somehow? If with js not, with anything else ? (mod_bridgecall or something) (Maybe if would useful an other callback function setAnswerHook() ?) Thanks any help, Tamas Anthony Minessale wrote: >please update to 4773 or better and see the code examples >in the commit log. > >http://fisheye.freeswitch.org/changelog/FreeSWITCH/?cs=4773 > >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: Tamas Cseke <cstomi.levlist(a)gmail.com> >To: freeswitch-users(a)lists.freeswitch.org >Sent: Saturday, March 24, 2007 5:34:13 AM >Subject: [Freeswitch-users] spidermonkey- session done->script terminate > >Hello, > >I need help using session object, please help me. >"Note: While the JavaScript application will continue to execute after a >hangup is detected the session object will be gone and any attempt to >access it will cause the JavaScript to terminate. Save off any >variables you might need for post hangup processing and use your local >variables instead." > >How can i avoid the script termination, is it possible to somehow safely check if the session hungup? > >exactly i'd like to call an user, if don't answer call another, i can't use simply mod_bridgecall because > >i know the destinations just in runtime, i try use fetchURLHash to determine the next user, when one fails. > > > >var queue = argv; >var hash = new Array(); > >while (true) { > >// fetch first agent > >fetchURLHash("http://localhost:8000/lockagent/"; + queue + "/"+session.uuid, "hash"); >var agent = hash["agent"]; > > >//call agent > >//... > new_session = new Session(session, > "sofia/devil/" + agent + "%192.168.1.93", //dest > "xml",//dialplan > "default",//context > > "",//cid_name > ""), //cid_num > "",//network_addr > "",//ani > > "",//anii > "",//rdnis > "",//username > 10);//to > > > // i need help in this condition > > // on timeout, new_session is done, and if i try access its properties, script terminate > > if (session.ready() && new_session.ready()) { > > bridge(session, new_session); > fetchURLHash(" >http://localhost:8000/unlockagent/"; + agent + "/" + session.uuid, "hash"); > break; > > } > else { > // should try again, but script terminated > > console_log("notice","timeout, try again...\n"); > //wait a bit > } > fetchURLHash("http://localhost:8000/unlockagent/ >" + agent + "/" + session.uuid, "hash"); >} > >Thanks any help, >Tamas > >_______________________________________________ >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 > > > > > > > > >____________________________________________________________________________________ >Be a PS3 game guru. >Get your game face on with the latest PS3 news and previews at Yahoo! Games. >http://videogames.yahoo.com/platform?platform=120121 > > >------------------------------------------------------------------------ > >_______________________________________________ >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 ____________________________________________________________________________________ Get your own web address. Have a HUGE year through Yahoo! Small Business. http://smallbusiness.yahoo.com/domains/?p=BESTDEAL

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unable to establish call to pstn
by Helmut Kuper 04 Apr '07

04 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I use freeswitch as a gateway to PSTN. I'm not able anymore to establish calls from softphone registered to freeswitch and PSTN extension since sofia config changed from registrations to gateways. The registration works fine with PSTN gateway provider (at least I get an OK message). But when I try to call a PSTN extension, I got first a "Proxy Authentication Request" message, then a DECLINE message from PSTN GW. When I compare the REGISTER Message from freeswitch to PSTN GW with the INVITE message, I found that the contact in message header are not the same. In register it is "sip:032223930016@212.6.123.118:5060", in invite it is "sip:mod_sofia@212.6.123.118:5060" (see register-contact.png and invite-contact.png). I think this is the reason why. is there a way in sofia config to change this? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGE6vr4tZeNddg3dwRAlW/AJ49wbETmQvcm0/yKimEHCmdaZ2OMwCggDWP 7dQWRFC0uKoUd/sObG6PgQw= =Uwau -----END PGP SIGNATURE-----

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Re: [Freeswitch-users] spidermonkey- session done->script terminate
by Anthony Minessale 04 Apr '07

04 Apr '07

please update to 4773 or better and see the code examples in the commit log. http://fisheye.freeswitch.org/changelog/FreeSWITCH/?cs=4773 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: Tamas Cseke <cstomi.levlist(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Saturday, March 24, 2007 5:34:13 AM Subject: [Freeswitch-users] spidermonkey- session done->script terminate Hello, I need help using session object, please help me. "Note: While the JavaScript application will continue to execute after a hangup is detected the session object will be gone and any attempt to access it will cause the JavaScript to terminate. Save off any variables you might need for post hangup processing and use your local variables instead." How can i avoid the script termination, is it possible to somehow safely check if the session hungup? exactly i'd like to call an user, if don't answer call another, i can't use simply mod_bridgecall because i know the destinations just in runtime, i try use fetchURLHash to determine the next user, when one fails. var queue = argv; var hash = new Array(); while (true) { // fetch first agent fetchURLHash("http://localhost:8000/lockagent/" + queue + "/"+session.uuid, "hash"); var agent = hash["agent"]; //call agent //... new_session = new Session(session, "sofia/devil/" + agent + "%192.168.1.93", //dest "xml",//dialplan "default",//context "",//cid_name ""), //cid_num "",//network_addr "",//ani "",//anii "",//rdnis "",//username 10);//to // i need help in this condition // on timeout, new_session is done, and if i try access its properties, script terminate if (session.ready() && new_session.ready()) { bridge(session, new_session); fetchURLHash(" http://localhost:8000/unlockagent/" + agent + "/" + session.uuid, "hash"); break; } else { // should try again, but script terminated console_log("notice","timeout, try again...\n"); //wait a bit } fetchURLHash("http://localhost:8000/unlockagent/ " + agent + "/" + session.uuid, "hash"); } Thanks any help, Tamas _______________________________________________ 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 ____________________________________________________________________________________ Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games. http://videogames.yahoo.com/platform?platform=120121

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Unable to Configure FreeSwitch
by Dinesh Dialani 03 Apr '07

03 Apr '07

Hi All, Can any body tell me how to configure FreeSwitch. I have gone through Getting_Started_Guide link and tried to configure it by making changes in xml configuration files, but didn't get any success. I have compiled FreeSwitch with VisualStudio 2005. I just need to configure the basic functionality of FreeSwitch and want to be able to dial "pa call 1234" as given in Getting_Started_Guide link. Thanks in advance. Dinesh

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Re: [Freeswitch-users] Codec negotiation
by Anthony Minessale 03 Apr '07

03 Apr '07

You have a few options: Early Negotiation (default behaviour): *) When leg A calls FS the codecs will be compared against the configured codecs in the profile config. *) When B sends the invite, the chosen codec in leg A will be pushed to the top of the list regardless of if it's actually chosen in the config. (the other codecs offered to leg A will not appear) Late Negotation (requires param): <param name="inbound-late-negotiation" value="true"/> *) The call will hit the dialplan without looking at the codecs at all. *) The negotiation will take place when leg A is answered. *) This allows you to route the call to a script and examine the sdp and rewrite the acceptable codecs with the "codec_string" channel variable. Early Negotiation + Disable Transcoding: <param name="disable-transcoding" value="true"/> If this sofia profile param is set, all B legs that you invite will contain *only* the codec negotiated by the A leg. Also in either case the variable "codec_string" on the A leg controls what codecs will be offered to the B leg. The variable "absolute_codec_string" is similar except it implies the implicit list of codecs and will disable the addition of the A leg codec to the list as described in the defualt behaviour. 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: James H Thompson <jht(a)lj.net> To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, April 2, 2007 6:20:50 PM Subject: [Freeswitch-users] Codec negotiation I wondering how FreeSwitch does codec negotiation in the following call flow: SIP Endpoint A ----> FreeSwitch ----> SIP Endpoint B For example, if the SIP Endpoint A offers Codecs: G.711u, G.711a, G.729 when FreeSwitch sets up the Call to SIP Endpoint B, how does it determine what list of Codecs to offer to SIP Endpoint B? What I'm looking for is for Freeswitch to always offer the same list of Codecs to SIP Endpoint B as it received from SIP Endpoint A. In this application, having the RTP packets bypass FreeSwitch and go directly between SIP Endpoint A and SIP Endpoint B is not an option since the two SIP Endpoints cannot talk to each other directly. SIP Endpoint A is in public IP space, and SIP Endpoint B is in private IP space. _______________________________________________ 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 ____________________________________________________________________________________ Looking for earth-friendly autos? Browse Top Cars by "Green Rating" at Yahoo! Autos' Green Center. http://autos.yahoo.com/green_center/

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