[asterisk-users] oneway audio with asterisk behind cisco pix 506

[Ravichandran Rajagopal]

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:listacc at ocosa.com] 
Sent: Friday, February 08, 2008 11:11 PM
To: ravi at vaishnavy.com; 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/4028805362 at incoming,30)
>
> [incoming]
>
> exten => 4025901000,1,Goto(1000,1)
>
> exten => 1000,1,Goto(customer_ivr,s,1)
>
> Thanks
>
> sunMoonstar.
>
> ------------------------------------------------------------------------
>
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