RE: [Asterisk-Users] Unknown RTP codec 72 received

2004-10-24 Thread Robert Jackson 


> -Original Message-
> From: Danny Froberg [mailto:[EMAIL PROTECTED] 
> Sent: Sunday, October 24, 2004 4:37 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Unknown RTP codec 72 received
> 
> 
> 19 Question + this one and no answer;
> 
> Does anyone have a clue what causes "Unknown RTP codec 72 
> received" notice 
> and how to fix it?
> 
> Regards
> Danny
> 

I receive this message when I call from X-Lite.  I notice that
it is usually when I am sending DTMF digits.  I could be using 
the wrong dtmfmode (using info), but I am not sure.

This message is rather annoying so I would definitely like to 
see if anyone else has gotten it figured out.

Robert Jackson
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[Asterisk-Users] Unknown RTP codec 72 received

2004-10-24 Thread Danny Froberg 
19 Question + this one and no answer;
Does anyone have a clue what causes "Unknown RTP codec 72 received" notice 
and how to fix it?

Regards
Danny
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RE: [Asterisk-Users] Unknown RTP codec 72 received

2004-09-13 Thread Elman Efendiyev 
Thanks for the hint Eric, but yes, before sending a message to the list
I checked google and wiki and NO - I didn't find an answer/solution/any
info on this subject.
There was couple of the same questions on the list but none of them
answered

--
Sincerely,
Elman Efendiyev
[EMAIL PROTECTED] 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling
Sent: Monday, September 13, 2004 5:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Unknown RTP codec 72 received


On Mon, 2004-09-13 at 06:13, Elman Efendiyev wrote:

> I get "Unknown RTP codec 72 received" message in console when call in 
> progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN

None of the 19 hits I saw on Google about this were helpful?

To search the Asterisk mailing list archive go to www.google.com
and put site:lists.digium.com in addition to your other query
terms.

-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a
related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

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RE: [Asterisk-Users] Unknown RTP codec 72 received

2004-09-13 Thread Low, Adam 
According to IANA's list of RTP payload types 
(http://www.iana.org/assignments/rtp-parameters) RTP payload type 72 fulls within the 
following range:

  72--76 reserved for RTCP conflict avoidance [RFC3550]

I can't find much else in RFC3550 that defines it further but this should start you on 
the right path I hope.

-Original Message-
From: Eric Wieling [mailto:[EMAIL PROTECTED]
Sent: 13 September 2004 16:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Unknown RTP codec 72 received


On Mon, 2004-09-13 at 06:13, Elman Efendiyev wrote:

> I get "Unknown RTP codec 72 received" message in console when call in
> progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN



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Re: [Asterisk-Users] Unknown RTP codec 72 received

2004-09-13 Thread Eric Wieling 
On Mon, 2004-09-13 at 06:13, Elman Efendiyev wrote:

> I get "Unknown RTP codec 72 received" message in console when call in
> progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN

None of the 19 hits I saw on Google about this were helpful?

To search the Asterisk mailing list archive go to www.google.com
and put site:lists.digium.com in addition to your other query
terms.

-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

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[Asterisk-Users] Unknown RTP codec 72 received

2004-09-13 Thread Elman Efendiyev 
Hi all,

I get "Unknown RTP codec 72 received" message in console when call in
progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN
over voicepulse connect (IAX) and to FWD echo test (SIP). But this
message only with one SIP client, others (X-Lite too) not giving this
message. All X-Lite settings are identical. Asterisk is last cvs version

This what I see in console (FWD):

vgw3*CLI>
-- Executing Dial("SIP/332-552e", "SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/fwd-357f is ringing
-- SIP/fwd-357f answered SIP/332-552e
-- Attempting native bridge of SIP/332-552e and SIP/fwd-357f Sep 13
11:02:52 NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72
received Sep 13 11:02:57 NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown
RTP codec 72 received Sep 13 11:03:01 NOTICE[245776]: rtp.c:489
ast_rtp_read: Unknown RTP codec 72 received Sep 13 11:03:02
NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received
Sep 13 11:03:07 NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown RTP
codec 72 received
  == Spawn extension (full-access, 1700613, 1) exited non-zero on
'SIP/332-552e'

And voicepulse:
vgw3*CLI>
-- Executing Dial("SIP/332-3f30",
"IAX2/[EMAIL PROTECTED]/011") in new stack
-- Called [EMAIL PROTECTED]/011
-- Call accepted by 66.234.228.160 (format GSM)
-- Format for call is GSM
-- IAX2/voicepulse/3 stopped sounds
-- IAX2/voicepulse/3 stopped sounds
-- IAX2/voicepulse/3 answered SIP/332-3f30
Sep 13 11:06:37 NOTICE[262160]: rtp.c:489 ast_rtp_read: Unknown RTP
codec 72 received Sep 13 11:06:42 NOTICE[262160]: rtp.c:489
ast_rtp_read: Unknown RTP codec 72 received same
skipped--
Sep 13 11:10:24 NOTICE[262160]: rtp.c:489 ast_rtp_read: Unknown RTP
codec 72 received Sep 13 11:10:29 NOTICE[262160]: rtp.c:489
ast_rtp_read: Unknown RTP codec 72 received
-- Hungup 'IAX2/voicepulse/3'
  == Spawn extension (full-access, 492103998463, 1) exited non-zero on
'SIP/332-3f30'

My sip.conf (234 user didn't give Unknown RTP codec 72 received message,
332 gives this message, only difference is internet path to users and
firewall type):

[general]
port = 5060
tos=lowdelay
videosupport=no
disallow = all
allow = gsm
allow = ulaw
canreinvite = no

[fwd]
context = in
type = peer
disallow = gsm
allow = ulaw
userneme = XX
secret = xxx
host = fwd.pulver.com

[234]
context = full-access
type = friend
disallow = ulaw
insecure = no
username = 234
secret = xxx
host = dynamic
nat = yes
dtmfmode = rfc2833
callerid = <234>

[332]
context = full-access
type = friend
disallow = ulaw
insecure = no
username = 332
secret = xxx
host = dynamic
nat = yes
dtmfmode = rfc2833
callerid = <332>

Could somebody tell me whay this "Unknown RTP codec 72 received" means
and how to fix it? Thanks.

--
Sincerely,
Elman Efendiyev
[EMAIL PROTECTED] 

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[Asterisk-Users] Unknown RTP codec 72 received NOTICE while using Xlite with *

2003-11-20 Thread C M 
hi. i am using xlite and asterisk mailny for internal
operations(PBX). i have used ulaw. i can successfully
make calls but the sound quality is not so good. and
when the talking starts i get a notive saying that.. 

NOICE[17424]: ile rtp.c, Line 417 (ast_rtp_read):
Unknown RTP codec 72 received 

and i get tonnes of this message until i hang up. 

what can be the problem? how can th sound quality be
improved? 


cm

=
Designs

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