asterisk users - Jul 2003 - RTP codec 13 received - Cisco incompatibility?

Hello,

In our SIP network, Asterisk is the central PBX, and it routes calls to the 
PSTN thru a Cisco Router - IOS 12.2(11)T9.

If a client softphone calls directly via Cisco to the PSTN, the call works 
successfully.

If the client softphone calls via Asterisk to other SIP internal extension, 
it work fine too.

The problem is when a client calls an Asterisk extension, and Asterisk 
transfers the call (via SIP) to the Cisco:

 - Pingtel (192.168.1.10) calls 300@192.168.200.200 (Extension 300 in 
Asterisk)
 - Asterisk transfers to 666554433@192.168.200.99 (Cisco GW)
 - Cisco tries to call to PSTN (666554433)

In that context, Asterisk generates this message while ringing:

NOTICE[540685]: File rtp.c, Line 291 (ast_rtp_read): Unknown RTP codec 13 
received

The PSTN recipient's phone rings. The client does not receive the typical 
intermittent tone/signal that means "the recipient's phone is
ringing". When
the recipient answers, the call is inmediantly finished. Maybe a 
short "Hello" can be listened.

Asterisk shows a response back from Cisco:

Bad Request - 'Invalid IP Address'

In sip.conf, Asterisk is forced to use g711ulaw. I've tried other codecs
with
no success.

What is the real problem?.
Is it a RTP problem with "codec 13", o a SIP problem?.
Is there a Cisco-Asterisk incompatibility?.

This is the sequence generated by Asterisk:

    -- Registered SIP 'pingtel01' at 192.168.1.10 port 5061 expires 500
    -- Executing Dial("SIP/pingtel01-af0d",
"SIP/666554433@192.168.200.200")
in new stack
    -- Called 666554433@192.168.200.200
NOTICE[573453]: File rtp.c, Line 291 (ast_rtp_read): Unknown RTP codec 13 
received
NOTICE[573453]: File rtp.c, Line 291 (ast_rtp_read): Unknown RTP codec 13 
received
    -- SIP/192.168.200.200-a3d2 answered SIP/pingtel01-af0d
    -- Attempting native bridge of SIP/pingtel01-af0d and SIP/192.168.200.200-
a3d2
    -- Got SIP response 400 "Bad Request - 'Invalid IP
Address'" back from
192.168.200.99
  == Spawn extension (default, 003, 1) exited non-zero on
'SIP/peter-af0d'

Thank you very much,
Mark Cerrajetto.

Probably needs some more information.  I would consider placing a detailed
bug report in the bug tracker including the output of "sip debug" with
a
call going through.

Mark

On Thu, 31 Jul 2003, Cerrajetto wrote:
> Hello,
>
> In our SIP network, Asterisk is the central PBX, and it routes calls to the
> PSTN thru a Cisco Router - IOS 12.2(11)T9.
>
> If a client softphone calls directly via Cisco to the PSTN, the call works
> successfully.
>
> If the client softphone calls via Asterisk to other SIP internal extension,
> it work fine too.
>
> The problem is when a client calls an Asterisk extension, and Asterisk
> transfers the call (via SIP) to the Cisco:
>
>  - Pingtel (192.168.1.10) calls 300@192.168.200.200 (Extension 300 in
> Asterisk)
>  - Asterisk transfers to 666554433@192.168.200.99 (Cisco GW)
>  - Cisco tries to call to PSTN (666554433)
>
> In that context, Asterisk generates this message while ringing:
>
> NOTICE[540685]: File rtp.c, Line 291 (ast_rtp_read): Unknown RTP codec 13
> received
>
> The PSTN recipient's phone rings. The client does not receive the
typical
> intermittent tone/signal that means "the recipient's phone is
ringing". When
> the recipient answers, the call is inmediantly finished. Maybe a
> short "Hello" can be listened.
>
> Asterisk shows a response back from Cisco:
>
> Bad Request - 'Invalid IP Address'
>
> In sip.conf, Asterisk is forced to use g711ulaw. I've tried other
codecs with
> no success.
>
> What is the real problem?.
> Is it a RTP problem with "codec 13", o a SIP problem?.
> Is there a Cisco-Asterisk incompatibility?.
>
> This is the sequence generated by Asterisk:
>
>     -- Registered SIP 'pingtel01' at 192.168.1.10 port 5061 expires
500
>     -- Executing Dial("SIP/pingtel01-af0d",
"SIP/666554433@192.168.200.200")
> in new stack
>     -- Called 666554433@192.168.200.200
> NOTICE[573453]: File rtp.c, Line 291 (ast_rtp_read): Unknown RTP codec 13
> received
> NOTICE[573453]: File rtp.c, Line 291 (ast_rtp_read): Unknown RTP codec 13
> received
>     -- SIP/192.168.200.200-a3d2 answered SIP/pingtel01-af0d
>     -- Attempting native bridge of SIP/pingtel01-af0d and
SIP/192.168.200.200-
> a3d2
>     -- Got SIP response 400 "Bad Request - 'Invalid IP
Address'" back from
> 192.168.200.99
>   == Spawn extension (default, 003, 1) exited non-zero on
'SIP/peter-af0d'
>
> Thank you very much,
> Mark Cerrajetto.
>
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