July 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

newbie question
by Ngo-Vi Hoai-Anh 24 Jul '09

24 Jul '09

Hi folk, I'm very new to FreeSwitch. I've read all the FAQs and traced the mailing list back to 12.2008 but still not found the answers for my questions. Please help! 1. Is it possible to make unauthenticated call to FS in the manner 1000@<IP of the FS box>? 2. Is there already a java implementation for FS like http://asterisk-java.org/ for Asterisk? Thank you . Hoai-Anh

2 2

Dial up from confernece issue
by Łukasz Zwierko 24 Jul '09

24 Jul '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, sorry if you're getting this again, I'm not sure if this mail got deliverd to the mail-list (I didn't get a copy...) Anyway, I want to use bgdial command to add a person to a already started conference (that is, call that person and when answered - add the channel to conference). The scenario is I have two sip clients registered in default context - 1000 and 1001. 1000 dials conference number (3001 in this case) and new conference is started. I want to dial out to second using bgdial, unfortunately mod_sofia drops the call with 'Cannot Blind Transfer 1 Legged calls' message. Should I use the bgdial command differently? Or perhaps I should do this totally differently? Logs attached below. Thanks for any help, Lukasz freeswitch@Zwierko-laptop> conference list API CALL [conference(list)] output: Conference 3001-192.168.0.1 (1 member) 3;sofia/internal/1000@192.168.0.1;25f125a5-3556-a449-85ec-e4001336f313;1000;1000;hear|speak|floor;0;0;300 freeswitch@Zwierko-laptop> conference 3001-192.168.0.1 bgdial sofia/default/1001(a)192.168.0.1 API CALL [conference(3001-192.168.0.1 bgdial sofia/default/1001(a)192.168.0.1) output: OK freeswitch@Zwierko-laptop> 2009-07-19 13:19:44.451100 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001(a)192.168.0.1 [b9fada7f-9c1d-4949-af8a-a8220ce f9c5b] 2009-07-19 13:19:44.459100 [NOTICE] switch_channel.c:602 New Channel sofia/internal/48228882211(a)192.168.0.1 [4f6b26dd-a0cb-2846-ad17-5f517e60e2e7] 2009-07-19 13:19:44.488100 [INFO] mod_dialplan_xml.c:252 Processing TelkaSwitch->1001 in context public 2009-07-19 13:19:44.495100 [ERR] sofia.c:4174 Cannot Blind Transfer 1 Legged calls 2009-07-19 13:19:44.498100 [NOTICE] sofia.c:3775 Hangup sofia/internal/1001(a)192.168.0.1 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2009-07-19 13:19:44.501100 [ERR] mod_conference.c:4351 Cannot create outgoing channel, cause: NO_USER_RESPONSE 2009-07-19 13:19:44.512100 [NOTICE] switch_core_session.c:1085 Session 9 (sofia/internal/1001(a)192.168.0.1) Ended 2009-07-19 13:19:44.512100 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001(a)192.168.0.1 [CS_DESTROY] 2009-07-19 13:19:44.628100 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/48228882211(a)192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-19 13:19:44.708100 [NOTICE] switch_core_session.c:1085 Session 10 (sofia/internal/48228882211(a)192.168.0.1) Ended 2009-07-19 13:19:44.708100 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/48228882211(a)192.168.0.1 [CS_DESTROY] -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEcBAEBAgAGBQJKY3/FAAoJED7LBosr0F2uWo8H/iJEHblsAENCjRh/dsZvj9br Mq6txy7iafLE970XxvToaa0+FGBFxN+S6yQ6ampNPd8t+jl6WwC79Btwr+NLgXEc NcWpVQp65QxKxA+MgQOyqWIskcMcxdf4Uht3wuLPZtre0BpjcAFhykweYjOy1jFp AYAM61ShogHlpXtl9Z6upDvWPoOzdY4m13EM7f0NmpbC32Sg+OOULEtsxvSkZ8ah DBKDyDdXFo9iIcReDqjsu/kzAgrBsAZvOiEbSPoQTjZgzX+UrbgqIc+rhmP60vyt 8u8ufDgzh7MC/VQObHKHLe8e/Zbpaf+3JiGxZBtFyoUFyP3DjHoR5TYu5IsENPU= =QlRU -----END PGP SIGNATURE-----

2 6

A stun server lookup
by velusamy velu 24 Jul '09

24 Jul '09

Dear All, When I start the freeSWITCH, I am receiving the following errors, 2009-07-24 16:56:23 [ERR] sofia_glue.c:566 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478[Remote Address Error!] 2009-07-24 16:56:23 [ERR] sofia.c:1972 config_sofia() Failed to get external ip. I commented the stun configurations in vars.xml.conf file eventhough I am receiving the same error. Pleas any one give solution to solve this error.... Regards, Velusamy.

2 1

Barge on on prompts
by Phillip Jones 23 Jul '09

23 Jul '09

Hi there, Very simple scenario: Session.DtmfReceivedFunction = (d, t) => { Log.WriteLine(LogLevel.Info, "Received {0} for {1}.", d, t); CollectedDigits = d.ToString().Trim(); return ""; }; Session.flushDigits(); Session.StreamFile(VoicemailPromptsDirectory + "abigfile.wav", 0); Question is, it there a way to kill the streaming when the a digit is pressed? I would use the Session.PlayAndGetDigits() but that does not help when want to string things together like: Session.StreamFile(VoicemailPromptsDirectory + "vm-to_delete_the_message.wav", 0); Session.StreamFile(VoicemailPromptsDirectory + "vm-press.wav", 0); Session.Say("7", "en", "number", "pronounced"); Any help would be appreciated.

3 4

Inbound call routing help
by Lars Zeb 23 Jul '09

23 Jul '09

I want to setup a dialplan for a single DID. I would like it to go to a specific extension, and if not picked up in 15 seconds, go to voicemail. I have set this scenario up and it works. But I would also like this person to be able to call this DID from outside FS via a phone and be able to retrieve their voicemail. I've seen the example of how to pick up an extension's voicemail while inside FS by checking to see if the destination_number is the same as the caller_id_number, and if so, listen to voicemail, otherwise leave the message with voicemail. But I don't have a clue how to accomplish this from outside, other than dedicating another DID to solely retrieving voicemail from outside. Any ideas? Thanks, Lars

5 9

Problem with Caller controls
by Rudolf Denert 23 Jul '09

23 Jul '09

Hello, I´m getting always the message: [ERR] mod_conference.c:5463 conference_new() Unable to install caller controls group 'test' I made a new caller-controls in the conference.conf.xml. It has the name "test". I also implemented the line: param name="caller-controls" value="test" in my conference-profil. Did I forget something? Thanks for your help! Greetz

1 0

Re: [Freeswitch-users] Confusing handling of incoming calls
by Pete Mueller 22 Jul '09

22 Jul '09

1 0

Good Information On How To Submit Bug Reports
by Michael Collins 22 Jul '09

22 Jul '09

FYI, Brian West called to my attention that one of our community members, John Wehle, has been very good at submitting useful bug reports, in many cases with patches. His style of reporting is worthy of imitation, so I've added a few links to the JIRA section of the Reporting Bugs wiki page: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Examples_Of_Well-Written_JIR… Please feel free to check it out. If you have any questions on what a bug report should look like then definitely read some of John's submissions and emulate his style. By submitting useful bug reports you will save the FreeSWITCH developers countless hours and headaches, not to mention the warm fuzzies you'll feel inside. :) -Michael

1 0

Re: [Freeswitch-users] Confusing handling of incoming calls
by Pete Mueller 22 Jul '09

22 Jul '09

3 3

Confusing handling of incoming calls
by Pete Mueller 22 Jul '09

22 Jul '09

2 1