[Asterisk-Users] RTP codec 13 received - Cisco incompatibility?
Cerrajetto
cerrajetto at pyme.net
Thu Jul 31 01:08:57 MST 2003
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 at 192.168.200.200 (Extension 300 in
Asterisk)
- Asterisk transfers to 666554433 at 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 at 192.168.200.200")
in new stack
-- Called 666554433 at 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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