From: Kurtis H. <khe...@cs...> - 2012-01-16 07:26:34
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Can you give us the SIP traffic? On Sun, Jan 15, 2012 at 11:15 PM, Han S <han...@gm...> wrote: > I've got OpenBTS 2.8 and Asterisk setup and semi-working using a B100 but am > having a couple of issues placing calls. > > The phones will happily register with OpenBTS and I can dial a sip extension > and call through ok however i'm unable to dial from my softphone to my > mobile. Or from one mobile to another. In both cases I get the following > message in asterisk; > > app_dial.c:2041 dial_exec_full: Unable to create channel of type 'SIP' > (cause 20 - Unknown) > == Everyone is busy/congested at this time (1:0/0/1) > > The only log messages I receive from OpenBTS (with log level set to INFO) > are the following; > transceiver: INFO 3042212720 Transceiver.cpp:732:driveTransmitFIFO: reduced > latency: 7:5 > transceiver: INFO 3042179952 Transceiver.cpp:757:writeClockInterface: > ClockInterface: sending IND CLOCK 1670853 > > I'm using the extensions.conf and sip.conf that is included in the > openbts2.8 svn and I've added both phones to the sqlite3.db using the > existing entry as a template. Is there something obvious I am missing? > > > > > ------------------------------------------------------------------------------ > RSA(R) Conference 2012 > Mar 27 - Feb 2 > Save $400 by Jan. 27 > Register now! > http://p.sf.net/sfu/rsa-sfdev2dev2 > _______________________________________________ > Openbts-discuss mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/openbts-discuss > |