Philipp Kempgen wrote: > Did you enable pedantic=yes in sip.conf? > thank you very much for your help, it fix the problem.
Is there any other issue that i have to take in mind for placing calls ? is there any option for set up pedantic for selected peers ? i use broadvoice too and it requires pedantic=no on the configuration: http://www.broadvoice.com/support_install_asterisk.html , how can coexist this two peers on just one asterisk ? [general] context=from-sip ; Default context for incoming calls allowguest=no ; Allow or reject guest calls (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls pedantic=yes ; Enable checking of tags in headers, language=es ; Default language setting for all users/peers dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity limitonpeers = yes ; Apply call limits on peers only. This will improve #include custom.d/sip_general.conf [authentication] #include custom.d/sip.conf -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
