[asterisk-users] Call bridged, but no sound

[Norbert Zawodsky]

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