[Asterisk-Users] Phones were working fine - Now there is no audiowhen calling between extensions

[Gabriel Afana]

I just did a little RTP debug and this is what it shows:

  == Spawn extension (304, 301, 1) exited non-zero on 'SIP/304-d9f8'
    -- Accepting AUTHENTICATED call from 216.152.244.81:
       > requested format = ulaw,
       > requested prefs = (),
       > actual format = ulaw,
       > host prefs = (),
       > priority = mine
    -- Executing Dial("IAX2/to_80-1", "SIP/301") in new stack
    -- Called 301
    -- SIP/301-1fec is ringing
    -- SIP/301-1fec answered IAX2/to_80-1
Got RTP packet from 24.50.66.128:2228 (type 0, seq 16810, ts 344311448, len
160)
Got RTP packet from 24.50.66.128:2228 (type 0, seq 16811, ts 344311608, len
160)
Got RTP packet from 24.50.66.128:2228 (type 0, seq 16812, ts 344311768, len
160)
Got RTP packet from 24.50.66.128:2228 (type 0, seq 16813, ts 344311928, len
160)
Got RTP packet from 24.50.66.128:2228 (type 0, seq 16814, ts 344312088, len
160)
Got RTP packet from 24.50.66.128:2228 (type 0, seq 16815, ts 344312248, len
160)
Got RTP packet from 24.50.66.128:2228 (type 0, seq 16816, ts 344312408, len
160)
Got RTP packet from 24.50.66.128:2228 (type 0, seq 16817, ts 344312568, len
160)
Got RTP packet from 24.50.66.128:2228 (type 0, seq 16818, ts 344312728, len
160)
Got RTP packet from 24.50.66.128:2228 (type 0, seq 16819, ts 344312888, len
160)
Got RTP packet from 24.50.66.128:2228 (type 0, seq 16820, ts 344313048, len
160)
Got RTP packet from 24.50.66.128:2228 (type 0, seq 16821, ts 344313208, len
160)
Got RTP packet from 24.50.66.128:2228 (type 0, seq 16822, ts 344313368, len
160)
..................


that goes on for ever while the call is in progress.  This is a call between
phones that go between two * servers.  If I make a call between phones both
registered to the same asterisk server, this is my RTP stream:

    -- Executing Dial("SIP/304-c211", "SIP/301|30|r") in new stack
    -- Called 301
    -- SIP/301-b2c8 is ringing
    -- SIP/301-b2c8 answered SIP/304-c211
    -- Attempting native bridge of SIP/304-c211 and SIP/301-b2c8
Got RTP packet from 24.50.66.128:2234 (type 0, seq 39729, ts -1972065425,
len 160)
Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24819, ts 0, len 160)
Got RTP packet from 24.50.66.128:2234 (type 0, seq 39730, ts -1972065265,
len 160)
Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24820, ts 160, len 160)
Got RTP packet from 24.50.66.128:2234 (type 0, seq 39731, ts -1972065105,
len 160)
Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24821, ts 320, len 160)
Got RTP packet from 24.50.66.128:2234 (type 0, seq 39732, ts -1972064945,
len 160)
Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24822, ts 480, len 160)
Got RTP packet from 24.50.66.128:2232 (type 0, seq 46694, ts 1105329892, len
160)
Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42690, ts 0, len 160)
Got RTP packet from 24.50.66.128:2232 (type 0, seq 46695, ts 1105330052, len
160)
Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42691, ts 160, len 160)
Got RTP packet from 24.50.66.128:2232 (type 0, seq 46696, ts 1105330212, len
160)
Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42692, ts 320, len 160)
Got RTP packet from 24.50.66.128:2232 (type 0, seq 46697, ts 1105330372, len
160)
Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42693, ts 480, len 160)
Got RTP packet from 24.50.66.128:2232 (type 0, seq 46698, ts 1105330532, len
160)
Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42694, ts 640, len 160)
[THE END]

Once I anser the call, the RTP string starts and then stops right where I
put [THE END].

- Gabe








> Hey group,
>     I have a Polycom 501 and a 301 together in my office.  Each phone is
> registered to a different server.  When I call one of the phones from the
> other, the other phone rings no problem (the calls are passed between
> servers via IAX).  However, when I answer it, there is absolutely no audio
> in either direction.  This just started happening today.
>
>     It was working great before - I just plugged them in, got them
> registered and I was calling between phones no problem.  Now I dont know
> what is happening and I cannot figure it out.  It seems like a NAT issue,
> but I have qualify=yes, nat=yes and insecure=port,invite.
>
> - Gabe
>
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