asterisk users - Oct 2006 - Call bridged, but no sound

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@inode-outbound|30|r") in new stack
  -- Called 32079781@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@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@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@192.168.1.200;user=phone>;tag=as2bff66b8
  To: "Chef" <sip:1@192.168.1.200>;tag=3mzvp0gi42
  Contact: <sip:032079781@192.168.1.200>
  Call-ID: 3c39ef1530d4-llu3czvejlxu@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

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@inode-outbound|30|r") in new stack
>   -- Called 32079781@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@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@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@192.168.1.200;user=phone>;tag=as2bff66b8
>   To: "Chef" <sip:1@192.168.1.200>;tag=3mzvp0gi42
>   Contact: <sip:032079781@192.168.1.200>
>   Call-ID: 3c39ef1530d4-llu3czvejlxu@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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>

On Thu, Oct 12, 2006 at 02:26:16PM +0200, Norbert Zawodsky
wrote:
> 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?
tcpdump is your friend, and a lot more useful than a flashing light :-)

# tcpdump -i eth0 -n -s0 udp

Look at the source and destination IPs and port numbers of the packets.

You might see:
1. packets from phone to Asterisk
2. packets from Asterisk to phone
3. packets from Asterisk to outside IP address
4. no packets from outside world to Asterisk

Or you might see:
1. packets from phone to outside IP address
2. no packets from outside IP address to phone

The message "Attempting native bridge" makes me think of this second
possibility. If so, you could try setting nat=yes and/or canreinvite=no on
the channel to the SIP provider, so that Asterisk proxies the RTP data.

Regards,

Brian.