[Asterisk-Users] SoftPhone to SoftPhone with No Voice

[Brian Cuthie]

Do you have iptables turned on with rules that restrict packets to the 
RTP ports?  Try doing an iptables --flush then see if it works. If so, 
you'll need to open up the UDP ports that RTP is configured to use.

Normally Asterisk would open up the firewall by sending packets out 
those ports, but when both sides of the call live outside the firewall 
(iptables in this case) and they're both SIP you'll have this problem. 
See my previous posting for a more detailed explanation (or email me 
directly).

-brian

deepak at netandroid.com wrote:

>Hello
>
>I Installed Asterisk on RedHat 9. I am currently try to configure minimum with
>two softphone talking to each other over the LAN. I am using X-Lite softphones
>from xten.com site. I defined 3 phones in sip.conf and also specifies in
>extensions.conf file. I am able to ring other computer but there is no voice
>exchange ( i can't hear any think except ring). Here is the portion of sip.conf
>and extensions.conf. 
>
>Let me know if i missed something.
>
>Thanks
>
>Deepak
>
>sip.conf
>[general]
>port = 5060                     ; Port to bind to
>bindaddr = 0.0.0.0              ; Address to bind SIP channel to
>context = from-sip              ; Default context for incoming calls
>;srvlookup = yes                ; Enable DNS SRV lookups on outbound calls
>                                ; Asterisk only uses the first host in SRV
>records
>;pedantic = yes                 ; Enable slow, pedantic checking for Pingtel
>                                ; and multiline formatted headers for strict
>                                ; SIP compatibility
>;tos=lowdelay                   ; IP QoS parameter, either keyword or value
>                                ; like tos=184
>;maxexpirey=3600                ; Max length of incoming registration we allow
>realm=asterisk                  ; Our global authentication realm
>;defaultexpirey=120             ; Default length of incoming/outoing
>registration
>;notifymimetype=text/plain      ; Allow overriding of mime type in NOTIFY
>;videosupport=yes               ; Turn on support for SIP video
>
>;disallow=all                   ; Disallow all codecs
>allow=all                       ; Allow codecs in order of preference
>;allow=ulaw                     ; Allow codecs in order of preference
>;allow=ilbc
>
>.....
>[Phone1]
>type=friend
>host=dynamic
>defaultip=192.168.3.103
>dtmfmode=rfc2833
>context=from-sip
>callerid=" Win box " <1>
>[Phone2]
>type=friend
>host=dynamic
>defaultip=192.168.3.119
>dtmfmode=rfc2833
>context=from-sip
>callerid=" Deepak" <2>
>[Phone3]
>type=friend
>host=dynamic
>defaultip=192.168.3.106
>dtmfmode=rfc2833
>context=from-sip
>callerid=" Ravi " <3>
>
>[extensions.conf]
>[from-sip]
>exten=>1,1,Dial(SIP/Phone1,20,tr)
>exten=>2,1,Dial(SIP/Phone2,20,tr)
>exten=>3,1,Dial(SIP/Phone3,20,tr)
>
>
>----------------------------------------------------------------
>This message was sent using IMP, the Internet Messaging Program.
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>  
>