July 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

(no subject)
by anthony kanlo 22 Jul '08

22 Jul '08

Hello freeswitch users. I have a serious problem of registering my softphones with freeswitch. I have jus installed the application and i was trying out the sample examples and the following error messages. 2008-05-11 23:49:23 [WARNING] sofia_reg.c:892 sofia_reg_parse_auth() can't find user [1005 at 10.3.2.34] 2008-05-11 23:49:23 [DEBUG] sofia_reg.c:482 sofia_reg_handle_register() send challange for [1005 at 10.3.2.34] 2008-05-11 23:50:16 [WARNING] sofia_reg.c:892 sofia_reg_parse_auth() can't find user [stanley at 10.3.2.34] 2008-05-11 23:50:16 [DEBUG] sofia_reg.c:482 sofia_reg_handle_register() send challange for [stanley at 10.3.2.34] 2008-05-11 23:50:55 [DEBUG] sofia_reg.c:482 sofia_reg_handle_register() send challange for [sye at 10.3.2.34] can you send me some sample solution to this problems.

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FreeSWITCH uses old IP address
by Ivan C Myrvold 22 Jul '08

22 Jul '08

I have a NAT problem with my FreeSWITCH which shows up only when my public address have changed AFTER I have started FreeSWITCH. I don't know if this is a small bug in FreeSWITCH, therefore I want to get your opinion before I post a Jira ticket: I have an incoming SIP call from a registered SIP provider This trace is after I have forced my router to get a new public address. My routers old public address is 85.165.194.100 My routers new public address is 85.165.204.52 http://pastebin.freeswitch.org/4927 This trace is after I have restarted FreeSWITCH: My routers public address is still 85.165.204.52 http://pastebin.freeswitch.org/4928 The first trace is normal up to line 69, when FreeSWITCH is sending the old IP address in the Contact field. The same in lines 96, 98, 102 and 111, where the old IP address are used in the Via field, Contact field and Remote-Party-ID etc. FreeSWITCH is using the new IP address in the To field in all messages. Why is FreeSWITCH using both the old and new IP address? Ivan

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Current state of 1.0.1
by jay binks 22 Jul '08

22 Jul '08

I realize 1.0.1 is not a scheduled release, and that there is no set release date.. however Im hanging out for it to release on my production systems ( need some of the bug fixes ) I realize I COULD just use SVN head, but for a matter of a few days I decided to wait.. its now been a few weeks :) ( which is a bit of a pain, but I understand... and appreciate the reasons ) anyways... just wanting to get a feel for where things are at with 1.0.1 and see if I should bite the bullet and just install a SVN copy... ( but I just don't feel right about doing that in a production environment ) Thanks anyways. Jay

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originating a call from JS
by John Wehle 21 Jul '08

21 Jul '08

Still fighting with our System 25 voice mail ports. Turns out that sending DTMF to control the message waiting lights only works when done while receiving a dialtone from the PBX ... you can't just send the tones at the end of a voice mail call. I have some JavaScript which looks like: function mwi_on (ext) { mwi_session = new Session(); mwi_session.setCallerData ("caller_id_name", mwi_caller_name); mwi_session.setCallerData ("caller_id_number", mwi_caller_number); mwi_session.originate (session, mwi_device + "/#90" + ext, 15); mwi_session.execute("sleep", "1000"); mwi_session.hangup ("NORMAL_CLEARING"); } ... session.execute ("voicemail", "default " + domain + " " + to); mwi_on (to); ... which sometimes works fine and other times fails with a trace similar to: [NOTICE] switch_channel.c:533 switch_channel_set_name() New Channel OpenZAP/4:4/#90482 [9d2980b4-7557-dd11-8fb6-001fc6ab49e2] [WARNING] zap_zt.c:356 zt_open() Echo training not available for 4:4 [NOTICE] switch_ivr_originate.c:1263 switch_ivr_originate() Hangup OpenZAP/4:4/#90482 [CS_CONSUME_MEDIA] [NO_ANSWER] [WARNING] mod_spidermonkey.c:2824 session_originate() Cannot Create Outgoing Channel! [openzap/4/A/#90482] [ERR] inline:1 mod_spidermonkey() You must call the session.originate method before calling this method! [NOTICE] switch_core_state_machine.c:157 switch_core_standard_on_execute() Hangup OpenZAP/1:1/551 [CS_EXECUTE] [NORMAL_CLEARING] [NOTICE] switch_core_session.c:754 switch_core_session_thread() Session 4 (OpenZAP/1:1/551) Ended [NOTICE] switch_core_session.c:756 switch_core_session_thread() Close Channel OpenZAP/1:1/551 [CS_HANGUP] [NOTICE] switch_core_session.c:754 switch_core_session_thread() Session 5 (OpenZAP/4:4/#90482) Ended ... the result of handling a voice mail from openzap span 1 extension 551 (a System 25 voice mail port) and attempting to original a call using openzap span 4 (a System 25 loopstart line) to extension #90482 which, if successful, causes 482's message waiting light to go on (the DTMF signals from the System 25 indicated to the JavaScript program that the voice mail message was for ext 482). So what's the correct, reliable way to pick up an openzap line, wait for dial tone, and dial some digits / send some DTMF? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Weekly docs call time: survey
by Michael S Collins 21 Jul '08

21 Jul '08

FYI, It is painfully obvious that there is no perfect time to hold this call. I would like to survey those who are serious about regularly attending this call and get your input on when this call should take place. I can attend on Tuesdays, Wednesdays, or Thursdays. My preferred day is Thursday but Wednesday is pretty good too. I am on the west coast so I can't really attend if the call is before 7am my time. I can attend as late as 4pm (start) on thursday or as late as 2pm on Wednesday. Please let me know what works for you and we will pick a time that works for as many people as possible. Thanks, MC Sent from my iPhone

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Mod-pocketsphinx Install error
by unknown 21 Jul '08

21 Jul '08

I am trying to make install the latest freeswitch SVN trunk on my OSX system, but having difficulty with the mod-pocketsphinx. I got the following error: *making install mod_pocketsphinx usage: mv [-f | -i | -n] [-v] source target mv [-f | -i | -n] [-v] source ... directory make[6]: *** [/opt/local/freeswitch/grammar/model/wsj1] Error 64 make[5]: *** [install] Error 1 make[4]: *** [mod_pocketsphinx-install] Error 1 make[3]: *** [install-recursive] Error 1 * Could anybody shed some light on this what's wrong with this? I checked my target directory *opt/local/freeswitch/grammar/model/wsj1, nothing is there. Thanks, Chris *

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parking scenario
by jeff sacksteder 21 Jul '08

21 Jul '08

I want to make a parking system slightly different than what comes in the box. The intended sequence would be- 1. User presses programmed button on phone 2. Call is transferred to 1st available park lot/orbit 3. User hears lot/orbit announced 4. User hangs up headset 5. 2nd user picks up call at announced extension Inevitable questions - In step 2, how would I go about accessing the park orbits in js to determine what the state of each is? I want no more than 1 call per lot, so I don't need mod_fifo, unless it's easier to use from js.

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Configuration
by Jair Santos 21 Jul '08

21 Jul '08

Hi all, I just installed FS with RH E5 OS on a Dell PowerEdge R200 Xeon 3065 Processor, 4MB. The installation worked very well and FS is running fine. I am an absolute beginner so I started reading the available docs and checked out the config files. My question is . How can I configure mydomain.com and setup termination that I bought from another provider ? If one could provide the very first steps informing which config file I should change it will be greatly appreciated. thanks in advance for any help Jair Santos

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Inbound Event_Socket SendMsg problems
by Anthony Knight 20 Jul '08

20 Jul '08

Hi I am having problems using a PHP client (hack using intralanman fs_sock.php) Using the dialplan I send the call to PARK. When I get the CHANNEL_PARK event if I issue: "bgapi uuid_broadcast $uuid /usr/local/freeswitch/socketcode/test.wav\r\n\r\n" (where I grab uuid from unique-id in CHANNEL_PARK event) I get the prompt to play back to me. HOWEVER the following scenarios with SendMsg do not work. If I send "SendMsg 60152374-54ec-11dd-8b83-5bb61f67b40f call_comand: execute execute-app-name: playback execute-app-arg: /usr/local/freeswitch/socketcode/test.wav\r\n\r\n" The reply back is [Content-Type] => command/reply [Reply-Text] => -ERR invalid session id [60152374-54ec-11dd-8b83-5bb61f67b40f [call_comand] => execute [execute-app-name] => playback [execute-app-arg] => /usr/local/freeswitch/socketcode/test.wav] >From the code, this looks like the uuid is not being seperated from the rest of the message when it is being read uuid is within the []? If I add explicit newlines at the end of each line "SendMsg 60152374-54ec-11dd-8b83-5bb61f67b40f\n call_comand: execute\n execute-app-name: playback\n execute-app-arg: /usr/local/freeswitch/socketcode/test.wav\r\n\r\n" I get back [Content-Type] => command/reply [Reply-Text] => +OK and slightly later a PRIVATE_COMMAND event, [Command] => SendMsg d4bb10c6-54ec-11dd-8b83-5bb61f67b40f [Event-Name] => PRIVATE_COMMAND [Core-UUID] => 8f70f40c-5480-11dd-8b83-5bb61f67b40f [FreeSWITCH-Hostname] => standcanada.tknights.com [FreeSWITCH-IPv4] => 72.249.85.158 [FreeSWITCH-IPv6] => 127.0.0.1 [Event-Date-Local] => 2008-07-18 17:13:24 [Event-Date-GMT] => Fri, 18 Jul 2008 17:13:24 GMT [Event-Date-timestamp] => 1216401204843801 [Event-Calling-File] => switch_ivr.c [Event-Calling-Function] => switch_ivr_parse_next_event [Event-Calling-Line-Number] => 460 and in freeswitch.log I see 2008-07-18 17:13:24 [DEBUG] switch_core_state_machine.c:140 switch_core_standard_on_execute() sofia/default/1008(a)freeswitch.tknights.com Execute park() 2008-07-18 17:13:24 [DEBUG] switch_core_session.c:611 switch_core_session_queue_private_event() Kill sofia/default/1008(a)freeswitch.tknights.com [BREAK] 2008-07-18 17:13:24 [ERR] switch_ivr.c:309 switch_ivr_parse_event() Invalid Command! 2008-07-18 17:13:24 [DEBUG] sofia.c:2048 sofia_handle_sip_i_state() Channel sofia/default/1008(a)freeswitch.tknights.com entering state [ready] 2008-07-18 17:13:35 [DEBUG] sofia.c:2048 sofia_handle_sip_i_state() Channel sofia/default/1008(a)freeswitch.tknights.com entering state [terminated] which I think means it is not parsing out the "execute" any suggestions on what I'm doing wrong? I'm on Debian Etch, trunk 9072 Thanks Tony

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Gafachi Origination
by Klaus Teller 19 Jul '08

19 Jul '08

Hi Folks, anybody there to help me setup Gafachi for origination? here is what they propose for Asterisk: [general] static=yes writeprotect=yes [default] exten => _1NXXNXXXXXX,1,SetCallerID(2125551212|a) exten => _1NXXNXXXXXX,2,Dial,SIP/${EXTEN}@gafachi exten => _1NXXNXXXXXX,3,Hangup exten => _011.,1,SetCallerID(2125551212|a) exten => _011.,2,Dial,SIP/${EXTEN}@gafachi exten => _011.,3,Hangup [gafachi-incoming] ; NOTE: this section is for Gafachi Origination (incoming) services. ; DID and/or Toll free origination numbers need to include the "1" prefix ; Replace the 12125551234 with your DID or Toll Free number, and repeat or ; change as needed. exten => 12125551234,1,Dial,Zap/g1 exten => 12125551234,2,Hangup /etc/asterisk/sip.conf [general] port=5060 bindaddr=0.0.0.0 context=default tos=lowdelay disallow=all allow=ulaw register=>a6409sAHZ7088wSG:j0MGA0MktWZEMpgU@sip.gafachi.com ; NOTE: The line below ([gafachi]) can not be changed, otherwise your Asterisk ;system will reject calls, with a "403 Forbidden", from the Gafachi Network. [gafachi] type=friend username=a6409sAHZ7088wSG secret=j0MGA0MktWZEMpgU host=sip.gafachi.com canreinvite=no fromuser=a6409sAHZ7088wSG dtmfmode=rfc2833 context=gafachi-incoming Thanks, Klaus L'Imbecile. -- GMX startet ShortView.de. Hier findest Du Leute mit Deinen Interessen! Jetzt dabei sein: http://www.shortview.de/wasistshortview.php?mc=sv_ext_mf@gmx

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