asterisk users - Jan 2010 - rtp.c:883 ast_rtcp_read: RTCP Read too short

Hi:
I have a linksys voip gateway connecting to an asterisk server ,when i dial a
call from the linksys gateway to asterisk , i see repeated messages of a RTP
errors ,and at same time i hear fake ring on the linksys?, This is wht i see on
asterisk console?:
?
-- Executing [9613070741 at direct:1] Set("SIP/03070741-088bd470",
"CALLERID(number)=96170707070") in new stack
??? -- Executing [9613070741 at direct:2]
Dial("SIP/03070741-088bd470", "SIP/usa/9613070741") in new
stack
??? -- Called usa/9613070741
[Jan 28 18:17:36] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short
[Jan 28 18:17:42] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short
??? -- Call on SIP/usa-08906450 left from hold
??? -- SIP/usa-08906450 is making progress passing it to SIP/03070741-088bd470
??? -- SIP/usa-08906450 is ringing
??? -- Call on SIP/usa-08906450 left from hold
??? -- SIP/usa-08906450 is making progress passing it to SIP/03070741-088bd470
[Jan 28 18:17:50] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short
[Jan 28 18:17:53] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short
[Jan 28 18:17:57] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short

?


      
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The ring isn't fake :] The Linksys GW isn't dissing, is just responding
to an INVITE. The problem is that you have problem passing voice. In other
words: check RTP ports settings on server & client or the firewall rules.

Alex

Da: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at
lists.digium.com] Per conto di wassim darwich
Inviato: gioved? 28 gennaio 2010 17:38
A: asterisk-users at lists.digium.com
Oggetto: [asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short

Hi:
I have a linksys voip gateway connecting to an asterisk server ,when i dial a
call from the linksys gateway to asterisk , i see repeated messages of a RTP
errors ,and at same time i hear fake ring on the linksys , This is wht i see on
asterisk console :

-- Executing [9613070741 at direct:1] Set("SIP/03070741-088bd470",
"CALLERID(number)=96170707070") in new stack
    -- Executing [9613070741 at direct:2]
Dial("SIP/03070741-088bd470", "SIP/usa/9613070741") in new
stack
    -- Called usa/9613070741
[Jan 28 18:17:36] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short
[Jan 28 18:17:42] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short
    -- Call on SIP/usa-08906450 left from hold
    -- SIP/usa-08906450 is making progress passing it to SIP/03070741-088bd470
    -- SIP/usa-08906450 is ringing
    -- Call on SIP/usa-08906450 left from hold
    -- SIP/usa-08906450 is making progress passing it to SIP/03070741-088bd470
[Jan 28 18:17:50] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short
[Jan 28 18:17:53] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short
[Jan 28 18:17:57] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short



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Hi:
Firewall is disabled ,so no need to worry about firewall,but i dont know?where
to check rtp settings and what do?i need to search for ,can you guide me please.


      
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