September 2007 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Mod_sofia loading fail
by Anthony Minessale 18 Sep '07

18 Sep '07

Also can you make sure you are up to date with the latest trunk when reporting problems because we are in beta now so we need to work with the newest code when we deal with issues. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 ----- Original Message ---- From: Michael Jerris <mike(a)jerris.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, September 14, 2007 9:13:47 AM Subject: Re: [Freeswitch-users] Mod_sofia loading fail Could you post the error messages you get when it tries to load please? On 9/14/07, Yogesh Chaturvedi <ychaturvedi(a)gmail.com > wrote:Hi, i am using lastest code form trunk : http://svn.freeswitch.org/svn/freeswitch/trunk and compiling version 5615 , I compiled successfully code but it do not load mod_sofia: I debug the code and found that it fail at Apr_dso_load () function and do not return status as 0 : But when i compile code (that i am using from last 3 month) has version 4712 , It works fine , what may be the problem: I am using : visual studio 2005 as compiler Microsoft windows 2000 SP4 as OS Thanks Yogesh _______________________________________________ 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 ____________________________________________________________________________________ Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mail&p=summer+activities+for+kids&…

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Using FreeSwitch with sip-Proxy
by Yogesh Chaturvedi 18 Sep '07

18 Sep '07

Hi! I am using FreeSwitch as sipPbx and Opensbc as sip proxy in some prototype, I want to register my FreeSwitch with my sip provider (that is gafachi) and request should go viya SipProxy(Opensbc). My freeswitch settings are: <gateways> <gateway name="gatway ip"> /// account username *required* /// <param name="username" value="username"/> /// auth realm: *optional* same as gateway name, if blank ///  <param name="realm" value="gafachi"/> /// account password *required* /// <param name="password" value="password"/> /// extension for inbound calls: *optional* same as username, if blank /// <param name="extension" value="username"/> /// proxy host: *optional* same as realm, if blank /// <param name="proxy" value="sip proxy ip"/> /// expire in seconds: *optional* 3600, if blank /// <param name="expire-seconds" value="600"/> </gateway> </gateways> As we know realm is the Ip address that goes with username and make request for example username is guest and realm name is gafachi request will be guest@gafachi and proxy is the ip address of sip proxy. When we set Geteway setting in freeswitch configuration file "freeswitch.xml" We gave realm as gafachi ip address and proxy as opensbc machine address . According to this, register request should to gafachi viya sip-proxy, but when we see this wireshark it add proxy ipaddress in register request instead of realm in sip message. As: To:<sip:a2541shRzJKLjtQA@192.168.96.113> So I debugged the source code of FreeSwitch and changed the code in : mod_sofia.c and in config sofia() and chage as : gateway->register_to = switch_core_sprintf(gateway->pool, "sip:%s@%s", username, proxy); To gateway->register_to = switch_core_sprintf(gateway->pool, "sip:%s@%s", username, realm); and its working fine and request go through proxy and secrate code captured by OpenSbc. Buit i think it is wrong to edit source code: when I used Asterisk and give register request in sip.conf (Asterisk conf file) like register=>username@gafachi:password:username@opensbcip and everything seems working fine. Also In this case it sends REGISTER sip:opensbcip (request uri) and username@opensbcip(To uri) so in this case opensbc takes it as it is going to register with opensbc and doesn't forward the request to gafachi. I'll be thankful for any guidance on what's going wrong or should I set something else somewhere. Thanks in Advance! Regards, Yogesh

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Re: [Freeswitch-users] Sofia reporting 480 - maximum number of sessions
by Anthony Minessale 17 Sep '07

17 Sep '07

you can try it though I would have thought closing the script would also close the sockets but it could easily be the problem if those socks are not closing or you are not hanging up the channels before you exit. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 ----- Original Message ---- From: David Knell <dave(a)3c.co.uk> To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, September 17, 2007 2:53:42 PM Subject: Re: [Freeswitch-users] Sofia reporting 480 - maximum number of sessions This *might* be it. My external scripts uses this one as a framework: http://svn.freeswitch.org/svn/freeswitch/trunk/scripts/socket/sock.pl - and leaves one zombie process for each time it's called. I imagine that the errors which I saw were a result of tables being filled up and a new session/socket/thread/something not being able to be created. My proposed fix is to add (from fs.pl) $SIG{CHLD} = sub {$fs->disconnect(); die "done"}; which ought to both kill off the zombies and relieve me of making sure that I call $fs->disconnect on every occasion when the call-handling thread could exit. Makes sense, or am I barking up a blind alley? Cheers -- Dave Unfortunately not. All that was there, though, were the error reports as below, one for each incoming call - I'll capture the console next time it happens. Is there any other debug info that'd be useful? Cheers -- Dave do you have a copy of the console log of this problem with the verbose errors? Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 ----- Original Message ---- From: David Knell <dave(a)3c.co.uk> To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, September 17, 2007 1:01:10 PM Subject: [Freeswitch-users] Sofia reporting 480 - maximum number of sessions Hi - I've a little calling-card application which uses FreeSWITCH as an engine via mod_event_socket. Every so often, it stops accepting incoming calls; the error shown on the console is that Sofia's got its maximum number of sessions active. This isn't down to: - sockets being left open; - there being too many sessions (status and sofia status both showed 12 sessions last time it happened, and my max_sessions is set to 1000); Any ideas? Cheers -- Dave _______________________________________________ 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 Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. _______________________________________________ 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 -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 8692 0677 M: 07773 800623 http://www.3c.co.uk _______________________________________________ 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 -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 8692 0677 M: 07773 800623 http://www.3c.co.uk ____________________________________________________________________________________ Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more! http://tv.yahoo.com/collections/3658

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Re: [Freeswitch-users] Sofia reporting 480 - maximum number of sessions
by Anthony Minessale 17 Sep '07

17 Sep '07

do you have a copy of the console log of this problem with the verbose errors? Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 ----- Original Message ---- From: David Knell <dave(a)3c.co.uk> To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, September 17, 2007 1:01:10 PM Subject: [Freeswitch-users] Sofia reporting 480 - maximum number of sessions Hi - I've a little calling-card application which uses FreeSWITCH as an engine via mod_event_socket. Every so often, it stops accepting incoming calls; the error shown on the console is that Sofia's got its maximum number of sessions active. This isn't down to: - sockets being left open; - there being too many sessions (status and sofia status both showed 12 sessions last time it happened, and my max_sessions is set to 1000); Any ideas? Cheers -- Dave _______________________________________________ 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 a deal? Find great prices on flights and hotels with Yahoo! FareChase. http://farechase.yahoo.com/

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Mod_sofia loading fail
by Yogesh Chaturvedi 14 Sep '07

14 Sep '07

Hi, i am using lastest code form trunk : http://svn.freeswitch.org/svn/freeswitch/trunk and compiling version 5615 , I compiled successfully code but it do not load mod_sofia: I debug the code and found that it fail at Apr_dso_load () function and do not return status as 0 : But when i compile code (that i am using from last 3 month) has version 4712 , It works fine , what may be the problem: I am using : visual studio 2005 as compiler Microsoft windows 2000 SP4 as OS Thanks Yogesh

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Re: [Freeswitch-users] 2 SIP Lines, Difficulty Originating Call
by Chris Danielson 14 Sep '07

14 Sep '07

I attempted the session.streamFile(...) and was able to successfully hear the wav file. I also attempted calling the session.answer() for a second time within the cf_bridge_call function. Still, I am unable to hear the ringing sound even though the phone that I called is actually ringing. When the phone is answered, the originate appears to fail due to a 60 second timeout, which is reported as a NO_ANSWER. This happens regardless of whether or not the phone I am calling is answered or not. Here is the current execution flow within my Javascript: function cf_bridge_call(phone) { var str = "sofia/gateway/line2/"+phone; session.setVariable("ringback", "%(2000, 4000, 440.0, 480.0)"); session.preAnswer(); /* attemped a session.answer() here in another test. The sound file still played, not no ringing was heard. */ session.streamFile("/usr/local/freeswitch/sounds/you_rang.wav"); session.execute("bridge", str); } session.answer(); session.flushDigits(); /** * Query database and dynamically load a phone number into variable **/ var phoneNo = db_get_phone(); cf_bridge_call(phoneNo); The log output was: 2007-09-08 01:54:35 [DEBUG] switch_ivr_play_say.c:1007 switch_ivr_play_file() done playing file 2007-09-08 01:54:35 [NOTICE] switch_channel.c:385 switch_channel_set_name() New Chan sofia/sip.vonics.net/18881234567 [1f766c60-5de9-11dc-af73-3b3c22977fdb] 2007-09-08 01:54:35 [DEBUG] mod_sofia.c:1431 sofia_outgoing_channel() sofia/sip.vonics.net/18881234567 State Change CS_NEW -> CS_INIT 2007-09-08 01:54:35 [DEBUG] switch_core_session.c:647 switch_core_session_signal_state_change() Kill sofia/sip.vonics.net/18881234567 [BREAK] 2007-09-08 01:54:35 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/sip.vonics.net/18881234567) State INIT 2007-09-08 01:54:35 [DEBUG] mod_sofia.c:92 sofia_on_init() SOFIA INIT 2007-09-08 01:54:35 [DEBUG] sofia.c:71 sofia_event_callback() event [nua_i_state] status [0][INVITE sent] session: sofia/sip.vonics.net/18881234567 2007-09-08 01:54:35 [DEBUG] sofia.c:1028 sofia_handle_sip_i_state() Channel sofia/sip.vonics.net/18881234567 entering state [calling] 2007-09-08 01:54:35 [DEBUG] mod_sofia.c:105 sofia_on_init() sofia/sip.vonics.net/18881234567 State Change CS_INIT -> CS_RING 2007-09-08 01:54:35 [DEBUG] switch_core_session.c:647 switch_core_session_signal_state_change() Kill sofia/sip.vonics.net/18881234567 [BREAK] 2007-09-08 01:54:35 [DEBUG] switch_core_state_machine.c:418 switch_core_session_run() (sofia/sip.vonics.net/18881234567) State RING 2007-09-08 01:54:35 [DEBUG] mod_sofia.c:120 sofia_on_ring() SOFIA RING 2007-09-08 01:54:35 [DEBUG] switch_ivr_originate.c:48 originate_on_ring() sofia/sip.vonics.net/18881234567 State Change CS_RING -> CS_HOLD 2007-09-08 01:54:35 [DEBUG] switch_core_session.c:647 switch_core_session_signal_state_change() Kill sofia/sip.vonics.net/18881234567 [BREAK] 2007-09-08 01:54:35 [DEBUG] switch_core_state_machine.c:596 switch_core_session_run() (sofia/sip.vonics.net/18881234567) State HOLD 2007-09-08 01:54:35 [DEBUG] switch_core_state_machine.c:204 switch_core_standard_on_hold() Standard HOLD 2007-09-08 01:54:35 [DEBUG] switch_ivr_originate.c:619 switch_ivr_originate() Raw Codec Activation Success L16@8000hz 1 channel 20ms 2007-09-08 01:54:35 [DEBUG] switch_ivr_originate.c:667 switch_ivr_originate() Play Ringback Tone [%(2000, 4000, 440.0, 480.0)] 2007-09-08 01:54:35 [DEBUG] sofia.c:71 sofia_event_callback() event [nua_r_invite] status [401][Unauthorized] session: sofia/sip.vonics.net/18881234567 2007-09-08 01:54:35 [DEBUG] sofia_reg.c:691 sofia_reg_handle_sip_r_challenge() Authenticating 'FreeSWITCH' with 'Digest:"sip.vonics.net":16191111111:5874'. 2007-09-08 01:54:35 [DEBUG] sofia.c:71 sofia_event_callback() event [nua_i_state] status [0][INVITE sent] session: sofia/sip.vonics.net/18881234567 2007-09-08 01:54:35 [DEBUG] sofia.c:1028 sofia_handle_sip_i_state() Channel sofia/sip.vonics.net/18881234567 entering state [calling] 2007-09-08 01:55:35 [NOTICE] switch_ivr_originate.c:805 switch_ivr_originate() Hangup sofia/sip.vonics.net/18881234567 [CS_HOLD] [NO_ANSWER] 2007-09-08 01:55:35 [DEBUG] switch_channel.c:1076 switch_channel_perform_hangup() Kill sofia/sip.vonics.net/18881234567 [KILL] 2007-09-08 01:55:35 [DEBUG] switch_core_session.c:647 switch_core_session_signal_state_change() Kill sofia/sip.vonics.net/18582718005 [BREAK] 2007-09-08 01:55:35 [DEBUG] switch_ivr_originate.c:878 switch_ivr_originate() Originate Cancelled by originator termination Cause: 19 [NO_ANSWER] 2007-09-08 01:55:35 [INFO] mod_dptools.c:1087 audio_bridge_function() Originate Failed. Cause: NO_ANSWER Thanks again for all the help. Kind Regards, Chris Danielson Anthony Minessale wrote: > you can find out by playing a file in the script > to make sure you have an audio path. > > session.preAnswer(); > session.streamFile("/some.wav"); > > If that doesn't work the inbound device may not support early media so > try session.answer(); instead to compare what happens when you send a > 200 ok instead of a 183 progress > > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > > ----- Original Message ---- > From: Chris Danielson <chris(a)maxpowersoft.com> > To: freeswitch-users(a)lists.freeswitch.org > Sent: Thursday, September 6, 2007 11:50:29 PM > Subject: Re: [Freeswitch-users] 2 SIP Lines, Difficulty Originating Call > > Thanks again for all the help. My apologies for not replying sooner, > I was out of town for a week and needed some extra time to research > and try some of the things you had mentioned. Everything that is > brought up in this thread will get documented by me within the wiki. > > So far I have tried both techniques within my JavaScript code that > Anthony mentioned. Here is one example: > > function cf_bridge_call(phone) { > var str = "sofia/gateway/line2/"+phone; > session.setVariable("ringback", "%(2000, 4000, 440.0, 480.0)"); > session.preAnswer(); > session.execute("bridge", str); > } > cf_bridge_call("18881234567"); > > When I make this call I still do not hear the ringing and notice that > the console renders: > nta: received 180 Ringing for INVITE (87950735) > nta: 180 Ringing was discarded > > When the phone is actually answered, I still cannot hear the person on > the other side speaking too. > > Is this possibly due to the fact that I am having a NAT issue? Or > could this be that my carrier is doing something out of the norm? And > finally, should I implement this strictly as a dialplan within the > default_context.xml file and avoid handling this functionality within > JavaScript? > > Kind Regards, > Chris Danielson > > Anthony Minessale wrote: >> You should choose one or the other not both. >> One is inband ringing generated by FreeSWITCH and the other >> is just the sip message telling the phone to indicate ringinig. >> Here is a description I wrote on an earlier posting here that someone >> may want to stick in the wiki. >> >> ---snip >> >> 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 >> 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:213-799-1400 >> >> >> ----- Original Message ---- >> From: Tamas Cseke <cstomi.levlist(a)gmail.com> >> To: chris(a)maxpowersoft.com; freeswitch-users(a)lists.freeswitch.org >> Sent: Monday, August 27, 2007 2:34:22 AM >> Subject: Re: [Freeswitch-users] 2 SIP Lines, Difficulty Originating Call >> >> Chris Danielson írta: >> > If any one can help me, I am having two issues. Basically, while my >> > javascript is executing a single inbound SIP call the script >> > conditionally gets to a point where the end-user can originate a >> call to >> > an external phone number. Keep in mind that I have two sip lines >> > defined as gateways supplied by my carrier vonics.net. Upon >> calling the >> > originate method, the call is actually placed and the destination >> phone >> > actually begins to ring. At this point I still cannot hear the >> ringing >> > on the original session that was originated by my inbound call. Also, >> > the originate method times out regardless of whether or not the >> > destination "phone" number was answered or not. >> > >> > So my two issues are: >> > 1) Why does the originate method start a call, the destination phone >> > called actually rings, but always times out, regardless of whether or >> > not the phone was answered? >> > 2) When calling the originate method, I never hear the phone >> ringing on >> > my inbound connection. >> > >> You should call ring_ready app, if you want to hear the ringing afaik. >> >> /* set ringback tone */ >> session.setVariable("ringback", "%(2000, 4000, 440.0, >> 480.0)"); >> session.execute("ring_ready", ""); >> >> I found these pages, maybe you can find more about it on wiki. >> http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready >> >> Hope this help! >> >> Regards, >> Tamas

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Re: [Freeswitch-users] distance of RTP packets
by Anthony Minessale 13 Sep '07

13 Sep '07

nobody has ever reported such an issue so I assume you have some sort of particular problem that relates to what you are exactly doing. Maybe collect some details and try the beta version which is 1000 revisions higher than what you are using? Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 ----- Original Message ---- From: "Sluschny, Thomas" <Thomas.Sluschny(a)siemens.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, September 13, 2007 9:48:26 AM Subject: [Freeswitch-users] distance of RTP packets Hi all, i use an older version of FreeSwitch for some time now (trunk 4560) and i want to ask wether this problem is solved now: i have huge jitter between the RTP packets, e. g. with G.711@20ms i see 3 or 4 packets every 75 ms. Even sometimes i see packet every 90ms. It seems to be machine dependend, i use always WinXP but on different machines: workstation, Laptop and so on. Anyone else have these problems? I loaded a newer binary version for windows for a couple of weeks but it crashed, so i don't now if this problem is solved now. thanks in advance, thomas sluschny ____________________________________________________________________________________ Check out the hottest 2008 models today at Yahoo! Autos. http://autos.yahoo.com/new_cars.html

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SIP INFO based DTMF detection problem and remedy.
by Keith Laaks 09 Sep '07

09 Sep '07

Hi All, With Sofia talking to a gw using SIP INFO for DTMF transport, I was not able to get any DTMF based features (e.g. IVR) to work. In the traces I saw the following when I pressed a key on the phone (in this example, key=4): **************************************************************************** ********************************** I send 489 bytes to udp/[44.33.22.11]:5060 at 16:43:54.656234: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 44.33.22.11:5060;rport=5060;branch=z9hG4bK848e4e821f09dcf2847c25da87f5296b From: <sip:27118091700@44.33.22.11:5060>;tag=GR52RWG346-34 To: "unknown" <sip:878050000@99.88.77.66>;tag=7pBSryH1mjBUg Call-ID: a666e4af-d996-122a-9e99-000a9d08b6ea CSeq: 4 INFO User-Agent: FreeSWITCH(mod_sofia) Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition Content-Length: 0 ------------------------------------------------------------------------ nta: sent 200 OK for INFO (4) nua(0x815c130): event i_info 200 OK nua: nua_event: entering [33m2007-09-09 18:43:54 [DEBUG] sofia.c:1643 sofia_handle_sip_i_info() Unknown INFO Recieved: sip:27118091700@44.33.22.11:5060[signal=4 duration=100] **************************************************************************** ********************************** Looking through the code, I found in sofia.c: **************************************************************************** ********************************** void sofia_handle_sip_i_info(nua_t *nua, sofia_profile_t *profile, nua_handle_t *nh, switch_core_session_t *session, sip_t const *sip, tagi_t tags[]) { //placeholder for string searching char *signal_ptr; //Try and find signal information in the payload signal_ptr = strstr(sip->sip_payload->pl_data, "Signal="); //See if we found a match if (signal_ptr) { struct private_object *tech_pvt = NULL; switch_channel_t *channel = NULL; char dtmf_digit[2] = { 0, 0 }; //Get the channel channel = switch_core_session_get_channel(session); Etc.etc. **************************************************************************** *********************************** Note that the line "signal_ptr = strstr(sip->sip_payload->pl_data, "Signal=");" has the 'S' in 'Signal=' capitalized. But from the above traces it appears sofia sends the INFO string through to freeswitch in lower case. That's why freeswitch was not 'getting' the DTMF correctly. After just changing this to lower case and recompiling - works perfectly! Not sure, but maybe the code needs to be ammended to check for both variants? Would be interesting to know if this issue impacted any other freeswitch users - could also be an issue with my voip connection talking to kit not compling with latest standards. Let me know if this was of any use. Best Regads Keith

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Re: [Freeswitch-users] 2 SIP Lines, Difficulty Originating Call
by Anthony Minessale 07 Sep '07

07 Sep '07

You should choose one or the other not both. One is inband ringing generated by FreeSWITCH and the other is just the sip message telling the phone to indicate ringinig. Here is a description I wrote on an earlier posting here that someone may want to stick in the wiki. ---snip 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 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:213-799-1400 ----- Original Message ---- From: Tamas Cseke <cstomi.levlist(a)gmail.com> To: chris(a)maxpowersoft.com; freeswitch-users(a)lists.freeswitch.org Sent: Monday, August 27, 2007 2:34:22 AM Subject: Re: [Freeswitch-users] 2 SIP Lines, Difficulty Originating Call Chris Danielson írta: > If any one can help me, I am having two issues. Basically, while my > javascript is executing a single inbound SIP call the script > conditionally gets to a point where the end-user can originate a call to > an external phone number. Keep in mind that I have two sip lines > defined as gateways supplied by my carrier vonics.net. Upon calling the > originate method, the call is actually placed and the destination phone > actually begins to ring. At this point I still cannot hear the ringing > on the original session that was originated by my inbound call. Also, > the originate method times out regardless of whether or not the > destination "phone" number was answered or not. > > So my two issues are: > 1) Why does the originate method start a call, the destination phone > called actually rings, but always times out, regardless of whether or > not the phone was answered? > 2) When calling the originate method, I never hear the phone ringing on > my inbound connection. > You should call ring_ready app, if you want to hear the ringing afaik. /* set ringback tone */ session.setVariable("ringback", "%(2000, 4000, 440.0, 480.0)"); session.execute("ring_ready", ""); I found these pages, maybe you can find more about it on wiki. http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready Hope this help! Regards, 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 ____________________________________________________________________________________ Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. http://smallbusiness.yahoo.com/webhosting

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registering a sip softphone
by Santiago Gimenez Ocano 06 Sep '07

06 Sep '07

Hello I have been trying to register my SIP softphone, but y could not get it What I have to do?? Regards, Santiago _________________________________________________________________ Visita MSN Latino Entretenimiento: ¡música, cine, chismes, TV y más...! http://latino.msn.com/entretenimiento/

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