[asterisk-users] Call bridged, but no sound

[Henry.L.Coleman]

I have had this problem before and it always turns out to be the fire wall.
You SIP registration and signaling (port 5060) is going thru okay but the
audio signals use a range of different ports which (if blocked) will cause
the problems you experience. Try putting * in DMZ to test this theory



Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada


> Hello everybody,
>
> I have a problem and already browsed the mailing list archives but
> didn't find any help. So I ask here....
>
> My new * Box ist up & runnig. Got access to the SIP server of my
> Internet provider (Userid, password, phone number, ...). And yesterday I
> tried my first calls to the outside world. (Internal calls work).
>
> Now when I call from the SNOM-360 connected to Asterisk to my cellphone
> (or to any other number), the call is set up, but both sides cannot hear
> each other. The asterisk console says:
>
>   -- Executing Dial("SIP/1-08182b48",
> "SIP/32079781 at inode-outbound|30|r") in new stack
>   -- Called 32079781 at inode-outbound
>   -- SIP/inode-outbound-081906e8 is ringing
>   -- SIP/inode-outbound-081906e8 answered SIP/1-08182b48
>   -- Attempting native bridge of SIP/1-08182b48 and
> SIP/inode-outbound-081906e8
>
> With SIP DEBUG, somewhere in the tons of output i finde the following
> lines:
>
>   Found RTP audio format 8
>   Found RTP audio format 101
>   Peer audio RTP is at port 192.168.1.201:56190
>   Found description format pcma
>   Found description format telephone-event
>   Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8
> (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
>   Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
> (telephone-event), combined - 0x1 (telephone-event)
>   set_destination: Parsing <sip:1 at 192.168.1.201:5060;line=vz5y8h67> for
> address/port to send to
>   set_destination: set destination to 192.168.1.201, port 5060
>   Transmitting (NAT) to 192.168.1.201:5060:
>   ACK sip:1 at 192.168.1.201:5060;line=vz5y8h67 SIP/2.0
>   Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK559b80fb;rport
>   From: <sip:032079781 at 192.168.1.200;user=phone>;tag=as2bff66b8
>   To: "Chef" <sip:1 at 192.168.1.200>;tag=3mzvp0gi42
>   Contact: <sip:032079781 at 192.168.1.200>
>   Call-ID: 3c39ef1530d4-llu3czvejlxu at snom360
>   CSeq: 103 ACK
>   User-Agent: Asterisk PBX
>   Max-Forwards: 70
>   Content-Length: 0
>
>>From this I understand that both sides agreed on a common codec (alaw).
>
> As soon as the connection is up and the receiver is lifted on both
> sides, the leds of the DSL Modem between Asterisk and my ISP, and the
> leds of the switch between Asterisk and the SNOM phone start rapidly
> flashing. So I assume there are lots of data packets on the wire. But no
> sound in both receivers.... Could it still be a firewall problem?
>
> Any hints or ideas?
>
> Norbert
>
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