Maybe try type=friend as opposed to type=peer Still a newbie, but my understanding from what I read is that a peer is call out only.
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of hugolivude Sent: Monday, April 04, 2005 3:11 PM To: Matt; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] broadvoice Woops forgot to include my config files... ;***************************************************************** ;/etc/hosts # Do not remove the following line, or various programs # that require network functionality will fail. 127.0.0.1 localhost.localdomain localhost # proxy.dca.broadvoice.com 147.135.0.128 sip.broadvoice.com ; ;***************************************************************** ; ;/etc/asterisk/sip.conf ; [general] port=5060 ; Port to bind to (SIP is 5060) bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine) context=from-sip-external ; Send unknown SIP callers to this context pedantic=no register => [EMAIL PROTECTED]:<password>:[EMAIL PROTECTED];/3003 ; [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=8145551212 secret=<password> username=8145551212 insecure=very context=from-broadvoice authname=8145551212 dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no ; ;***************************************************************** ; /etc/asterisk/extensions.conf [general] static=yes ; These two lines prevent the command-line interface writeprotect=yes ; from overwriting the config file. Leave them here. ; ; [from-broadvoice] exten => s,1,Dial(ZAP/1,30) exten => s,2,Hangup [from_FXS] exten => _1NXXNXXXXXX, 1, dial(SIP/[EMAIL PROTECTED],30) exten => _1NXXNXXXXXX, 2, congestion() exten => _1NXXNXXXXXX, 102, busy() ; ;***************************************************************** ; ;/etc/asterisk/zapata.conf ; ; [channels] language=en context=from-FXO signalling=fxs_ks usecallerid=yes echocancel=yes echocancelwhenbridged=yes channel => 4 ; language=en context=from_FXS signalling=fxo_ks channel=>1 ; language=en context=from-ILS-FXS signalling=fxo_ksFailed to authenticate on INVITE to '"asterisk" <sip:[EMAIL PROTECTED]>;tag=as4a325b3a' channel=>2 ; ;***************************************************************** ; ;/Asterisk Console Output ; Asterisk Ready. *CLI> sip show registry Host Username Refresh State 147.135.0.128:5060 8145551212 120 Registered *CLI> -- Starting simple switch on 'Zap/1-1' -- Executing Dial("Zap/1-1", "SIP/[EMAIL PROTECTED]|30") in new stack -- Called [EMAIL PROTECTED] Mar 27 20:55:26 NOTICE[1116941248]: chan_sip.c:5047 handle_response: Failed to authenticate on INVITE to '"asterisk" <sip:[EMAIL PROTECTED]>;tag=as4a325b3a' Mar 27 20:55:26 WARNING[1209214528]: app_dial.c:347 wait_for_answer: Unable to forward voice == Spawn extension (from_FXS, 13035551212, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' On Apr 4, 2005 2:29 PM, Matt <[EMAIL PROTECTED]> wrote: > Hi, > I'm currently routing my asterisk server out over broadvoice.. it > seems I can do multiple outgoing and incoming calls.... does anyone > know if broadvoice actually allows this or not? > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
