Otis, I am new to Cisco PIX 506 and I am learning this. If you can help me with how to do this change on Cisco PIX it would be greatly appreciated.
Thx Ravi -----Original Message----- From: ListAcct [mailto:[EMAIL PROTECTED] Sent: Friday, February 08, 2008 11:11 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506 Ravi, Open up the RTP (UDP) ports on your pix. (EX. conduit permit udp host x.x.x.x eq 10049 any). Also set your asterisk rtp config span to something you can configure (10000 to 10200) unless you write a script to just copy and paste about 10000 to 20000 ports in your config on the pix. Cisco's are strange but secure. It took me about two hours to figure out after taking off the fixup and no more logging/debugging from the cisco. I actually fixed while a call was coming in. LOL! Let me know!!! --Otis Ravichandran Rajagopal wrote: > > Hi, > > I have the Cisco PIX 506 firewall right in front of the asterisk and I > am getting a one-way audio. I need your help/guidance to resolve this > problem. I have the "fixups" disabled for SIP in the Cisco PIX 506. > Any help rendered by you in this subject is greatly appreciated. I > have been breaking my head trying to resolve this problem for more > than one month. I have included the sip.conf and the extensions.conf > below. > > [SIP.conf] > > ; SIP Configuration example for Asterisk > > [general] > > context=incoming > > allowoverlap=no > > bindport=5060 > > bindaddr=0.0.0.0 > > localnet=192.168.5.0/255.255.255.0 > > externip=a.b.ccc.dd > > srvlookup=yes > > allow=ulaw > > allow=alaw > > [incoming] > > type=peer > > nat=no > > canreinvite=no > > host=xx.y.z.aaa > > qualify=yes > > dtmfmode=rfc2833 > > context=default > > [extensions.conf] > > [general] > > static=yes > > writeprotect=yes > > clearglobalvars=no > > [default] > > include => customer > > exten => h,1,Hangup > > exten => i,1,Congestion > > exten => i,2,Hangup > > [agnosco] > > include => local-extensions > > include => customer_ivr > > include => incoming > > [customer_ivr] > > include => local-extensions > > exten => s,1,Answer > > exten => s,n,Background(agnosco_intro) > > exten => s,n,WaitExten > > ;Dial said extensions > > exten => 5,1,Dial(SIP/[EMAIL PROTECTED],30) > > [incoming] > > exten => 4025901000,1,Goto(1000,1) > > exten => 1000,1,Goto(customer_ivr,s,1) > > Thanks > > sunMoonstar. > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
