Hello again! So I tried again, experimented a bit more and got this: channel originate sip/[email protected] [Jun 5 12:14:29] WARNING[8537]: chan_sip.c:17882 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '[email protected]'. Giving up. Below you can find a condensed version of my sip.conf. *** /etc/asterisk/sip.conf *** [general] context=sip-in ; Default context for incoming calls allowguest=yes ; Allow or reject guest calls (default is yes) match_auth_username=yes ; if available, match user entry using the allowoverlap=yes ; Disable overlap dialing support. (Default is yes) allowtransfer=yes ; Disable all transfers (unless enabled in peers or users) udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) tcpenable=no ; Enable server for incoming TCP connections (default is no) tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) srvlookup=yes ; Enable DNS SRV lookups on outbound calls disallow=all ; First disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=ilbc ; see doc/rtp-packetization for framing options mohinterpret=default mohsuggest=default language=en ; Default language setting for all users/peers relaxdtmf=yes ; Relax dtmf handling useragent=J's Asterisk ; Allows you to change the user agent string sdpsession=J's Asterisk ; Allows you to change the SDP session name string, (s=) sdpowner=juliencoder ; Allows you to change the username field in the SDP owner string, (o=) videosupport=no ; Turn on support for SIP video. You need to turn this alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected, dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering shrinkcallerid=yes ; on by default 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 rtpkeepalive=50 ; Send keepalives in the RTP stream to keep NAT open hash_users=32 hash_peers=32 hash_dialogs=16 recordhistory=yes ; Record SIP history by default allowsubscribe=no ; Disable support for subscriptions. (Default is yes) callcounter = yes ; Enable call counters on devices. This can be set per registertimeout=20 ; retry registration calls every 20 seconds (default) registerattempts=10 ; Number of registration attempts before we give up localnet=192.168.220.1/255.255.0.0 localnet=192.168.220.105/255.255.0.0 localnet=192.168.220.106/255.255.0.0 externip=my_networks_external_ip_adress ; JPC IP goes here directmedia=yes ; Asterisk by default tries to redirect the rtcachefriends=yes ; Cache realtime friends by adding them to the internal list rtsavesysname=yes ; Save systemname in realtime database at registration rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule domain=iptel.org,sip-in allowexternaldomains=yes [authentication] secret=password_for_iptel.org remotesecret=password_for_+iptel.org_again transport=upd,tcp nat=yes language=en [iptel] type=friend host=iptel.org [email protected] [email protected] ; how your provider knows you fromdomain=iptel.org remotesecret=password_for_iptel.org ; The password we use to authenticate to them secret=password_for_iptel.org_again ; The password they use to contact us callbackextension=S ; Register with this server and require calls coming back to this extension transport=udp ; This sets the transport type to udp for outgoing, and will busylevel=2 port=5060 ; only templates and examples after this line *** END of /etc/asterisk/sip.conf *** so this is it? Could you give me some hints, tips to get me going? Kindly yours Julien -------- Music was my first love and it will be my last (John Miles)
======== FIND MY WEB-PROJECT AT: ======== http://ltsb.sourceforge.net the Linux TextBased Studio guide ======= AND MY PERSONAL PAGES AT: ======= http://www.juliencoder.de -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
