October 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

oz dump Saying error
by lakshmanan ganapathy 05 Oct '09

05 Oct '09

HI all,

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register problem
by Frank Carmickle 04 Oct '09

04 Oct '09

Can someone point out what is wrong here. Thanks. Siptrace at http://carmickle.com/fs.txt <include> <gateway name="voiptalk.org"> <param name="username" value="81100068"/> <param name="password" value="my_pass"/> <param name="from-user" value="81100068"/> <param name="from-domain" value="voiptalk.org"/> <param name="proxy" value="voiptalk.org"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> <param name="extension" value="1000"/> <param name="contact-params" value="domain_name=$${domain}"/> <param name="context" value="public"/> </gateway> </include> --FC

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internal & external ip addresses of freeswitch
by Timur Irmatov 04 Oct '09

04 Oct '09

Hi. We have a local network 192.168.1.0/24, where all the users are. Out FreeSWITCH server is connected to this network, and has ip address 192.168.1.242. Through different network card it is connected to external gateway, and has address 172.16.12.11 in this network. I set up a test client with softphone. When incoming call is deliviered to this client, call is set up normally, but client can't hang it up. It sends BYE to external address - 172.16.12.11 - which is not reachable from the client. It seems this address is coming from Contact: field in INVITE that FreeSWITCH sends: U 192.168.1.242:5060 -> 192.168.1.34:37169 INVITE sip:100@192.168.1.34:37169 SIP/2.0. Via: SIP/2.0/UDP 172.16.12.11;rport;branch=z9hG4bKrvp6jm3myyaaF. Max-Forwards: 70. From: "FreeSWITCH" <sip:0000000000@172.16.12.11>;tag=v817pS9c6v6Fe. To: <sip:100@192.168.1.34:37169>. Call-ID: 797bd088-29cd-122d-9b93-0060979d54c5. CSeq: 121117089 INVITE. Contact: <sip:mod_sofia@172.16.12.11:5060>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14898. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 267. Remote-Party-ID: "FreeSWITCH" <sip:0000000000@172.16.12.11>;party=calling;screen=yes;privacy=off. What should I tweak in freeswitch to change this behaviour? -- Timur Irmatov, xmpp:irmatov@jabber.ru

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Problem with subscription expire
by Helmut Kuper 04 Oct '09

04 Oct '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, it seems exired subsciptions are never cleared in FS. A look into sofia_presence.c confirms explains this /* negative in exptime means keep bumping up sub time to avoid a snafu where every device has it's own rules about subscriptions that somehow barely resemble the RFC not that I blame them because the RFC MAY be amibiguous and SHOULD be deleted. So to avoid the problem we keep resetting the expiration date of the subscription so it never expires. Eybeam completely ignores this option and most other subscription-state: directives from rfc3265 and still expires. Polycom is happy to keep upping the subscription expiry back to the original time on each new notify. The rest ... who knows...? */ For some reasons subscriptions created by Snom phones are filling up the sip_subscriptions table over time. This leads to some kind of DOS by FS against the subscribing phone ... The subscribtions are differentiate by call-id. This can be explained by RFC 3842 chapter 3.6 where expired subscriptions must be renewed with a NEW call-id. Because there is no hint about unsubscribing the old subscription I guess the clean up process has to be done by FS. Any way to get FS to do this job? Since there is no creation date or expire value which represents the expire as a timestamp I have no way to clean up the table manually via sql and cronjob - except cleaning the whole table ... A further (but background) question is, why do the subscriptions expire in snom phones at all ... regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKxG3q4tZeNddg3dwRArNEAJ9fjHLox1tt038ze0liUG0ki+wrfgCgsz09 pO+XUioXrBKJ/ozUOy1ZqeA= =nZaf -----END PGP SIGNATURE-----

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Outbound INVITE rejected with 480 Temp Unavail, Reason "MANDATORY_IE_MISSING"
by Jerry Richards 04 Oct '09

04 Oct '09

Hello All, I have an internal extension that needs to send an INVITE without SDP body (Content Length 0). Freeswitch is replying with 480 Temporarily Unavailable with reason "MANDATORY_IE_MISSING". Would anyone know what I need to do to enable this? Best Regards, Jerry

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Freeswitch as a softphone - presence?
by Raffaele P. Guidi 04 Oct '09

04 Oct '09

Hi, I was wondering how FreeSWITCH could notify presence through sofia gateways - the basic idea is to use it as a softphone, of course. Thanks, Raffaele

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wav files compression
by Keith Wood 04 Oct '09

04 Oct '09

I am working on an implementation for managing thousands of IVR within an organization. Right now, I am storing all audio files in wav format, but it quickly become unmanagable because the size of these wav files ( 8 bits mono ) quickly consuming a lot of the disk space. Is there anyway I can store those audio files and still have high quality audio for IVR? I know mp3 is smaller but freeswitch does not support it. any ideas? keith

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CDR page reworked
by Diego Viola 03 Oct '09

03 Oct '09

Hi FreeSWITCH community. I just wanted to say that I have reworked this page a bit as it was a bit poor, feel free to add anything else on it. http://wiki.freeswitch.org/wiki/CDR Regards, Diego

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voiptalk.org register 904
by Frank Carmickle 03 Oct '09

03 Oct '09

I hope this is more helpful. http://carmickle.com/fs-2009-10-02.txt <include> <gateway name="voiptalk.org"> <param name="username" value="81100068"/> <param name="password" value="my_pass"/> <param name="from-user" value="81100068"/> <param name="from-domain" value="voiptalk.org"/> <param name="proxy" value="voiptalk.org"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> <param name="extension" value="1000"/> <param name="contact-params" value="domain_name=$${domain}"/> <param name="context" value="public"/> </gateway> </include> Let me know what you think. Thanks. --FC

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Need Help in Getting DTMF
by Thangappan.M 03 Oct '09

03 Oct '09

Dear all, I am in the process of implementing IVR server in Perl using event outbound socket. Let take the following scenario. There are three menus in the IVR. First menu will authenticate you, second menu get the option value from you,. third menu will give the you the result. You already know all the numbers that you could give. So when the call answered you are giving the value in ONE SHOT. Is it possible to get all the DTMF values in one shot in freeswitch? It should have facility to recollect DTMF values and clear the DTMF values. -- Regards, Thangappan.M

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