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

Martin Joseph ast at stillnewt.org
Tue Mar 21 01:00:38 MST 2006


On Mar 20, 2006, at 4:39 PM, Gabriel Afana wrote:

> 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].
>

Did you try setting reinvite to no?  Seems the native bridge is what's 
failing.  Rethink your routing with regards native bridging (ie 
everybody is able to get through there nats and be identified?

I don't really know,  I am only trying to be helpful.  Hope it's worth 
something.

Marty




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