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Call Waiting not working!!!!!
by Irshad Ahamed 25 Jan '08

25 Jan '08

Hi, I am working on Freeswitch@ rev 7309. I remember with earlier revision call waiting was working fine. But now on getting the second incoming call FXS is returning 502 (BAD gateway error). Was there any change on FXS side pertaining to number of calls allowed or something??? Freeswitch Log: 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() sofia/default/6502@192.168.2.51:5060 Execute set(dialed_ext=6601) 2008-01-25 15:48:17 [DEBUG] mod_dptools.c:442 set_function() SET [dialed_ext]=[6601] 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() sofia/default/6502@192.168.2.51:5060 Execute set(hangup_after_bridge=true) 2008-01-25 15:48:17 [DEBUG] mod_dptools.c:442 set_function() SET [hangup_after_bridge]=[true] 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() sofia/default/6502@192.168.2.51:5060 Execute set(bypass_media=false) 2008-01-25 15:48:17 [DEBUG] mod_dptools.c:442 set_function() SET [bypass_media]=[false] 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() sofia/default/6502@192.168.2.51:5060 Execute bridge(openzap/1/1) 2008-01-25 15:48:17 [ERR] mod_openzap.c:833 channel_outgoing_channel() No channels available 2008-01-25 15:48:17 [ERR] switch_ivr_originate.c:827 switch_ivr_originate() Cannot Create Outgoing Channel! cause: DESTINATION_OUT_OF_ORDER 2008-01-25 15:48:17 [DEBUG] switch_ivr_originate.c:1226 switch_ivr_originate() Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 2008-01-25 15:48:17 [INFO] mod_dptools.c:1325 audio_bridge_function() Originate Failed. Cause: DESTINATION_OUT_OF_ORDER 2008-01-25 15:48:17 [NOTICE] mod_dptools.c:1352 audio_bridge_function() Hangup sofia/default/6502@192.168.2.51:5060 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2008-01-25 15:48:17 [DEBUG] switch_channel.c:1317 switch_channel_perform_hangup() Kill sofia/default/6502@192.168.2.51:5060 [KILL] 2008-01-25 15:48:17 [DEBUG] switch_core_session.c:643 switch_core_session_signal_state_change() Kill sofia/default/6502@192.168.2.51:5060 [BREAK] 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:413 switch_core_session_run() sofia/default/6502@192.168.2.51:5060 Running State Change CS_HANGUP 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:439 switch_core_session_run() (sofia/default/6502@192.168.2.51:5060) State HANGUP 2008-01-25 15:48:17 [DEBUG] mod_sofia.c:299 sofia_on_hangup() Channel sofia/default/6502@192.168.2.51:5060 hanging up, cause: DESTINATION_OUT_OF_ORDER 2008-01-25 15:48:17 [DEBUG] mod_sofia.c:332 sofia_on_hangup() Responding to INVITE with: 502 2008-01-25 15:48:17 [DEBUG] sofia.c:183 sofia_event_callback() event [nua_i_state] status [502][Call state] session: sofia/default/6502@192.168.2.51:5060 2008-01-25 15:48:17 [DEBUG] sofia.c:1430 sofia_handle_sip_i_state() Channel sofia/default/6502@192.168.2.51:5060 entering state [terminated] 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:45 switch_core_standard_on_hangup() Standard HANGUP sofia/default/6502@192.168.2.51:5060, cause: DESTINATION_OUT_OF_ORDER 2008-01-25 15:48:17 [DEBUG] switch_core_session.c:701 switch_core_session_thread() Session 3 (sofia/default/6502@192.168.2.51:5060) Locked, Waiting on external entities 2008-01-25 15:48:17 [NOTICE] switch_core_session.c:707 switch_core_session_thread() Session 3 (sofia/default/6502@192.168.2.51:5060) Ended 2008-01-25 15:48:17 [NOTICE] switch_core_session.c:709 switch_core_session_thread() Close Channel sofia/default/6502@192.168.2.51:5060 [CS_HANGUP] 2008-01-25 15:48:36 [DEBUG] zap_analog.c:665 process_event() EVENT [ONHOOK][1:1] STATE [UP] 2008-01-25 15:48:36 [DEBUG] zap_analog.c:691 process_event() Changing state on 1:1 from UP to DOWN 2008-01-25 15:48:36 [DEBUG] zap_analog.c:351 zap_analog_channel_run() Executing state handler on 1:1 for DOWN 2008-01-25 15:48:36 [DEBUG] mod_openzap.c:1037 on_fxs_signal() got FXS sig [STOP] 2008-01-25 15:48:36 [NOTICE] mod_openzap.c:1115 on_fxs_signal() Hangup OpenZAP/777 [CS_EXECUTE] [NORMAL_CLEARING] 2008-01-25 15:48:36 [DEBUG] switch_channel.c:1317 switch_channel_perform_hangup() Kill OpenZAP/777 [KILL] 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:643 switch_core_session_signal_state_change() Kill OpenZAP/777 [BREAK] 2008-01-25 15:48:36 [DEBUG] zap_analog.c:649 zap_analog_channel_run() ANALOG CHANNEL thread ended. 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:186 audio_bridge_thread() read: OpenZAP/777 Bad Frame.... Bubye! 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:358 switch_core_session_receive_message() Kill OpenZAP/777 [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:224 audio_bridge_thread() Kill sofia/default/777(a)sipx.oneconv.com [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:227 audio_bridge_thread() BRIDGE THREAD DONE [OpenZAP/777] 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:693 switch_ivr_multi_threaded_bridge() sofia/default/777(a)sipx.oneconv.com State Change CS_LOOPBACK -> CS_RESET 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:643 switch_core_session_signal_state_change() Kill sofia/default/777(a)sipx.oneconv.com [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:413 switch_core_session_run() OpenZAP/777 Running State Change CS_HANGUP 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:439 switch_core_session_run() (OpenZAP/777) State HANGUP 2008-01-25 15:48:36 [DEBUG] mod_openzap.c:418 channel_on_hangup() OpenZAP/777 CHANNEL HANGUP 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:45 switch_core_standard_on_hangup() Standard HANGUP OpenZAP/777, cause: NORMAL_CLEARING 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:701 switch_core_session_thread() Session 1 (OpenZAP/777) Locked, Waiting on external entities 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:180 audio_bridge_thread() write: OpenZAP/777 Bad Frame....[160] Bubye! 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:358 switch_core_session_receive_message() Kill sofia/default/777(a)sipx.oneconv.com [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:224 audio_bridge_thread() Kill OpenZAP/777 [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:227 audio_bridge_thread() BRIDGE THREAD DONE [sofia/default/777(a)sipx.oneconv.com] 2008-01-25 15:48:36 [NOTICE] switch_ivr_bridge.c:257 audio_bridge_on_loopback() Hangup sofia/default/777(a)sipx.oneconv.com[CS_RESET] [NORMAL_CLEARING] 2008-01-25 15:48:36 [NOTICE] switch_core_session.c:707 switch_core_session_thread() Session 1 (OpenZAP/777) Ended 2008-01-25 15:48:36 [NOTICE] switch_core_session.c:709 switch_core_session_thread() Close Channel OpenZAP/777 [CS_HANGUP] 2008-01-25 15:48:36 [DEBUG] switch_channel.c:1317 switch_channel_perform_hangup() Kill sofia/default/777(a)sipx.oneconv.com[KILL] 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:643 switch_core_session_signal_state_change() Kill sofia/default/777(a)sipx.oneconv.com [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:413 switch_core_session_run() sofia/default/777(a)sipx.oneconv.com Running State Change CS_HANGUP 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:439 switch_core_session_run() (sofia/default/777(a)sipx.oneconv.com) State HANGUP 2008-01-25 15:48:36 [DEBUG] mod_sofia.c:299 sofia_on_hangup() Channel sofia/default/777(a)sipx.oneconv.com hanging up, cause: NORMAL_CLEARING 2008-01-25 15:48:36 [DEBUG] mod_sofia.c:325 sofia_on_hangup() Sending BYE to sofia/default/777(a)sipx.oneconv.com 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:45 switch_core_standard_on_hangup() Standard HANGUP sofia/default/777(a)sipx.oneconv.com, cause: NORMAL_CLEARING 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:701 switch_core_session_thread() Session 2 (sofia/default/777(a)sipx.oneconv.com) Locked, Waiting on external entities 2008-01-25 15:48:36 [NOTICE] switch_core_session.c:707 switch_core_session_thread() Session 2 (sofia/default/777(a)sipx.oneconv.com) Ended 2008-01-25 15:48:36 [NOTICE] switch_core_session.c:709 switch_core_session_thread() Close Channel sofia/default/777(a)sipx.oneconv.com [CS_HANGUP] Thanks in advance... BR, Irshad.

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Newbie question: can Freeswitch create subconferences on-the-fly & let a moderator speak to all of them?
by Brian Burt 24 Jan '08

24 Jan '08

Short version: can Freeswitch create subconferences on-the-fly & let a moderator speak to all of them? Long version: We lead live experiential learning workshops, where we are constantly saying "get a partner and share about x" and then "now get in groups of 4/6/8" to share about that and then finally we come back to the "big room" where we take a few shares from the big group. In particular we would like to place participants in small groups where they interact with each other, but all groups can hear the facilitator. I'd like to do something similar over the phone, and am wondering if a programmer could fairly easily accomplish this in Freeswitch. It's critical that the presenter still be able to speak to everyone, even when they are in small subconferences. Thinking that Asterisk could do it, I wrote up some requirements below. They're definitely not all "must-haves". I heard from several Asterisk experts that the "one-way" nature of this requirement (subconference hears presenter but not the other way around) meant that Asterisk really wasn't suited for it. Below are the full requirements. Do people think they're do-able? Easily? What platforms would allow that, especially open-source? ps. If there's a video it's likely always simply the presenter delivering that. Ideally would have ability to connect with POTS but if this can be done in a computer-only method (is there still such a thing?) I'd still love to know about it. ---------------------------------------- -- ===================== Brian Burt, Founding Partner Financial Circles Empowerment in Community www.FinancialCircles.com 415-308-4258 mobile (best) 510-251-1189 office brianDELETE-ME(a)FinancialCircles.com ===================== - - - "Financial Circles teaches people to master money so that it serves their dreams. We offer courses and an active network where people wake up to possibility, master concrete financial and business skills, and take action on what truly matters to them." - - -

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'outbound' IVR with JavaScript - garbage collection issues
by tuhl＠ix.netcom.com 24 Jan '08

24 Jan '08

My application dials 100's of numbers, waits for user to answer, plays a .wav file, and waits for some DTMF. I'm using the Javascript interface to do all this now (actually it's a Perl script which launches JS by sending jsrun commands through the RPC XML interface). But at high call rates, I'm running into JS garbage collection issues fairly quickly. At 10 calls per second, after about 300 calls it looks like JS GC kicks in and runs for like 45 seconds, during which time I can't launch any more calls with JS (RPC XML accepts them, but then they are getting Q'd up inside the spidermonkey code, and eventually some of them fail because GC is taking so long, etc). Is JS the best way to do this application, and if so, any ideas how I workaround the GC? If not, what is the alternative (where I would still have access to the simple programming interface that JS gives me, as far as all the session. functions like waitForAnswer, streamFile, getDigits). Tom =============== tuhl(a)ix.netcom.com

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Re: [Freeswitch-users] 'outbound' IVR with JavaScript - garbage collection issues
by Anthony Minessale 24 Jan '08

24 Jan '08

looking at the C code in mod_spidermonkey I can see that session.originate and session.waitForAnswer/Media did not suspend the garbage collector. Most often, when the host system spidermonkey is embedded in calls something that will block in the C code we should use special code around it to disable garbage collection. It's sort of like a mutex only it's backwards where you want to unlock it while you do anything that blocks and re-lock it when you resume. Since these 3 function can delay execution of more javascript I added code to revision 7343 to do this. Please test. Also, even if it works, you can still be more efficient by doing what Mike suggested and use the new bgapi FSAPI call together with the originate FSAPI call to originate the call in the core then execute the script. bgapi originate {ignore_early_media=true}sofia/default/user(a)domain.com &javascript(myscript.js) This will launch your script with the session already created. If you omit bgapi the xmlrpc call will block until the call is established, if it's present it will run in it's own thread to establish the call and fire and event to indicate the results. 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: "tuhl(a)ix.netcom.com" <tuhl(a)ix.netcom.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, January 23, 2008 7:55:23 PM Subject: [Freeswitch-users] 'outbound' IVR with JavaScript - garbage collection issues My application dials 100's of numbers, waits for user to answer, plays a .wav file, and waits for some DTMF. I'm using the Javascript interface to do all this now (actually it's a Perl script which launches JS by sending jsrun commands through the RPC XML interface). But at high call rates, I'm running into JS garbage collection issues fairly quickly. At 10 calls per second, after about 300 calls it looks like JS GC kicks in and runs for like 45 seconds, during which time I can't launch any more calls with JS (RPC XML accepts them, but then they are getting Q'd up inside the spidermonkey code, and eventually some of them fail because GC is taking so long, etc). Is JS the best way to do this application, and if so, any ideas how I workaround the GC? If not, what is the alternative (where I would still have access to the simple programming interface that JS gives me, as far as all the session. functions like waitForAnswer, streamFile, getDigits). Tom =============== tuhl(a)ix.netcom.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 ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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Re: [Freeswitch-users] Hello from Scotland - Python - error importing module
by Traun Leyden 24 Jan '08

24 Jan '08

You'll want to change your dialplan to <extension name="500"> <condition field="destination_number" expression="^500$"> <action application="python" data="test"/> </condition> </extension> and make sure you put your test.py script somewhere on the PYTHONPATH by: 1. Copying or symlinking it to the appropriate site-packages directory 2. Copying test.py to /tmp, adding /tmp to your PYTHONPATH environment variable, and restarting freeswitch so it has the correct PYTHONPATH. (replace /tmp with your favorite directory) See http://wiki.freeswitch.org/wiki/Mod_python#PYTHONPATH for more details. > Hi, > > I've previously had a lot of Windows based SIP/IVR expreience with a > variety of technologies but am now looking at alternative architectures for > some forthcoming projects so have been doing some evaluation on platforms > for producing some IVR type solutions initially for inhouse hosting but > longer term to scale possibly on Amazon EC2. > > I'm also interested in contributing to the project if I settle on > Freeswitch. > > Initially speed of development / application tuning for new proof of > concepts is critical and the combination of Freeswitch / Python seems to fit > the bill (not withstanding my lack of significant experience with either) > > To test this out I have set up a box running CentOS 5, built Freeswitch > (with mod python) and configured the switch. > > Freeswitch works great for SIP calls between wifi handsets, so far so good > and a very painless set up, impressive ! > > Dialplan is modified as below, > > > <extension name="500"> > <condition field="destination_number" expression="^500$"> > <action application="python" data="test.py"/> > </condition> > </extension> > > test.py as below (borrowed from WIKI) > > from freeswitch import * > > def handler(uuid): > session = PySession(uuid) > session.answer() > console_log("INFO", "Hello, world\n") > session.hangup("1") > > is in python2.5/site-packages as is freeswitch.py > > however an incoming call to 500 generates the following > > 2008-01-22 21:46:12 [DEBUG] mod_dialplan_xml.c:273 dialplan_hunt() > sofia/default/1001@192.168.0.12:5060 State Change CS_RING -> CS_EXECUTE > 2008-01-22 21:46:12 [DEBUG] switch_core_session.c:643 > switch_core_session_signal_state_change() Kill > sofia/default/1001@192.168.0.12:5060 [BREAK] > 2008-01-22 21:46:12 [DEBUG] switch_core_state_machine.c:413 > switch_core_session_run() sofia/default/1001@192.168.0.12:5060 Running > State Change CS_EXECUTE > 2008-01-22 21:46:12 [DEBUG] switch_core_state_machine.c:452 > switch_core_session_run() (sofia/default/1001@192.168.0.12:5060) State > EXECUTE > 2008-01-22 21:46:12 [DEBUG] mod_sofia.c:185 sofia_on_execute() > sofia/default/1001@192.168.0.12:5060 SOFIA EXECUTE > 2008-01-22 21:46:12 [DEBUG] switch_core_state_machine.c:130 > switch_core_standard_on_execute() Standard EXECUTE > 2008-01-22 21:46:12 [DEBUG] switch_core_state_machine.c:144 > switch_core_standard_on_execute() sofia/default/1001@192.168.0.12:5060 > Execute python(test.py) > 2008-01-22 21:46:12 [NOTICE] mod_python.c:86 eval_some_python() Invoking > py module: test.py > 2008-01-22 21:46:12 [ERR] mod_python.c:106 eval_some_python() Error > importing module > 2008-01-22 21:46:12 [DEBUG] mod_python.c:180 eval_some_python() > Threadstate mod_python.c swap-out! > 2008-01-22 21:46:12 [NOTICE] switch_core_state_machine.c:201 > switch_core_standard_on_execute() Hangup sofia/default/1001@192.168.0.12:5060 > [CS_EXECUTE] [NORMAL_CLEARING] > > This suggests to me that the the dialplan is working correctly and we are > trying to execute test.py, which from playing with file names I am fairly > happy it is finding test.py , but beyond that I'm pretty well stuck for > tonight. > > Any one got any thoughts as to where I might be going wrong. > > Many Thanks > > Steve > >

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Hello from Scotland - Python - error importing module
by Steven Brown 24 Jan '08

24 Jan '08

Hi, I've previously had a lot of Windows based SIP/IVR expreience with a variety of technologies but am now looking at alternative architectures for some forthcoming projects so have been doing some evaluation on platforms for producing some IVR type solutions initially for inhouse hosting but longer term to scale possibly on Amazon EC2. I'm also interested in contributing to the project if I settle on Freeswitch. Initially speed of development / application tuning for new proof of concepts is critical and the combination of Freeswitch / Python seems to fit the bill (not withstanding my lack of significant experience with either) To test this out I have set up a box running CentOS 5, built Freeswitch (with mod python) and configured the switch. Freeswitch works great for SIP calls between wifi handsets, so far so good and a very painless set up, impressive ! Dialplan is modified as below, <extension name="500"> <condition field="destination_number" expression="^500$"> <action application="python" data="test.py"/> </condition> </extension> test.py as below (borrowed from WIKI) from freeswitch import * def handler(uuid): session = PySession(uuid) session.answer() console_log("INFO", "Hello, world\n") session.hangup("1") is in python2.5/site-packages as is freeswitch.py however an incoming call to 500 generates the following 2008-01-22 21:46:12 [DEBUG] mod_dialplan_xml.c:273 dialplan_hunt() sofia/default/1001@192.168.0.12:5060 State Change CS_RING -> CS_EXECUTE 2008-01-22 21:46:12 [DEBUG] switch_core_session.c:643 switch_core_session_signal_state_change() Kill sofia/default/1001@192.168.0.12:5060 [BREAK] 2008-01-22 21:46:12 [DEBUG] switch_core_state_machine.c:413 switch_core_session_run() sofia/default/1001@192.168.0.12:5060 Running State Change CS_EXECUTE 2008-01-22 21:46:12 [DEBUG] switch_core_state_machine.c:452 switch_core_session_run() (sofia/default/1001@192.168.0.12:5060) State EXECUTE 2008-01-22 21:46:12 [DEBUG] mod_sofia.c:185 sofia_on_execute() sofia/default/1001@192.168.0.12:5060 SOFIA EXECUTE 2008-01-22 21:46:12 [DEBUG] switch_core_state_machine.c:130 switch_core_standard_on_execute() Standard EXECUTE 2008-01-22 21:46:12 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() sofia/default/1001@192.168.0.12:5060 Execute python(test.py) 2008-01-22 21:46:12 [NOTICE] mod_python.c:86 eval_some_python() Invoking py module: test.py 2008-01-22 21:46:12 [ERR] mod_python.c:106 eval_some_python() Error importing module 2008-01-22 21:46:12 [DEBUG] mod_python.c:180 eval_some_python() Threadstate mod_python.c swap-out! 2008-01-22 21:46:12 [NOTICE] switch_core_state_machine.c:201 switch_core_standard_on_execute() Hangup sofia/default/1001@192.168.0.12:5060 [CS_EXECUTE] [NORMAL_CLEARING] This suggests to me that the the dialplan is working correctly and we are trying to execute test.py, which from playing with file names I am fairly happy it is finding test.py , but beyond that I'm pretty well stuck for tonight. Any one got any thoughts as to where I might be going wrong. Many Thanks Steve

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HD Telephony, Secure Audio
by Anthony Minessale 23 Jan '08

23 Jan '08

Now that we have 16k and Secure RTP working abd a RC1, it's time to spread the word a little. I cross-posted my story at digg http://digg.com/tech_news/HD_Telephony_HIP_or_Hype Check it out and digg it! 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 ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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Re: [Freeswitch-users] dialplan and move temporarily
by Anthony Minessale 23 Jan '08

23 Jan '08

if you want to do that, Just use the "transfer" application. <action application="transfer" data="6000"/> you don't even need to bother with any sip then. 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: Jonas Gauffin <jonas.gauffin(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, January 23, 2008 2:10:30 AM Subject: Re: [Freeswitch-users] dialplan and move temporarily Can I make freeswitch to use the dialplan for the redirect number too? That is, to run the dialplan again, but for the new number. (and by doing so, i will be able to provide a new extension instead of a complete sip uri) On Jan 22, 2008 3:10 PM, Anthony Minessale <anthmct(a)yahoo.com> wrote: > > Yes you can use the "redirect" application to > send a channel a protocol specific redirect response. > > In the case of a SIP channel it will be a 302 and > the arg should be the full sup uri (including sip:) > > <action application="redirect" data="sip:user@domain.com"/> > > you can wrap this in > <condition field="source" expression="mod_sofia" break="never"></condition> > > if you want to only send it on sip channels. > > > > 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: Jonas Gauffin <jonas.gauffin(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Sent: Tuesday, January 22, 2008 6:46:50 AM > Subject: [Freeswitch-users] dialplan and move temporarily > > Hello > > Can I in some way specify that an extension have moved temporarily > (status 302) in the dialplan? > I'm using my own server to provide FS with dialplans, and some times > the user have forwarded their extension(s) using my web interface. > > Regards, > 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 > > > ________________________________ > Looking for last minute shopping deals? Find them fast with Yahoo! Search. > _______________________________________________ > 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 ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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Re: [Freeswitch-users] dialplan and move temporarily
by Anthony Minessale 23 Jan '08

23 Jan '08

Yes you can use the "redirect" application to send a channel a protocol specific redirect response. In the case of a SIP channel it will be a 302 and the arg should be the full sup uri (including sip:) <action application="redirect" data="sip:user@domain.com"/> you can wrap this in <condition field="source" expression="mod_sofia" break="never"></condition> if you want to only send it on sip channels. 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: Jonas Gauffin <jonas.gauffin(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, January 22, 2008 6:46:50 AM Subject: [Freeswitch-users] dialplan and move temporarily Hello Can I in some way specify that an extension have moved temporarily (status 302) in the dialplan? I'm using my own server to provide FS with dialplans, and some times the user have forwarded their extension(s) using my web interface. Regards, 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 ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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Snom Contacts and FreeSWITCH Interop
by Brian West 23 Jan '08

23 Jan '08

Does anyone have a way to contact a real person at Snom to get some feedback on these bugs? I have four bugs one of which they have actually confirmed (Ticket#2007123010000023) and said will be fixed but I have three more that I have reported but haven't received any response on so I re- reported them today... they were previously in the respond on Ticket#2007123010000023 but its been a while since I received anything: Ticket#2008012310000047 RFC 2833 doesn't work when SRTP is enabled. (Works great on Polycom) Ticket#2008012310000038 SDP bug related to RTP/SAVP vs RTP/AVP in sdp and not accepting SAVP when you should for SRTP. (Polycom works great on this one too offers RTP/SAVP and RTP/AVP) Ticket#2008012310000029 Phone lies about its true port number in registration and subscription when using TLS. Will say one thing when it really means another. (Again polycom does this correctly) Example the phone registers to FreeSWITCH using TLS you get this in the contact <sip:1000@10.0.1.241:2048;transport=tls;line=uwohhel1> <839>:netstat -na | grep 10.0.1.241 tcp 0 0 10.0.1.250:5061 10.0.1.241:2051 ESTABLISHED Notice they aren't the same port? This will never work and requires many hacks to make this work correctly and we shouldn't have too, it should correctly report its port on register and subscribe. I would like to get these issues resolved in the next firmware release. If someone from Snom can call me.. I'm in Central Time (CST) my cellphone is listed at the bottom of this email. Thanks, Brian West FreeSWITCH.org 918-424-9278

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