VoIP Mailing List Archives

[atis at iq-labs.net]

Hi,

I experience my log flooded with warning messages like this:

[Apr 14 01:30:24] WARNING[19514] chan_sip.c: Remote host can't match
request NOTIFY to call
'5239fdfb15d20e5153a2f7365ce2ea4d at 69.80.215.12'. Giving up

I traced this down to point when we added to sip.conf status notifications:

allowsubscribe=yes
rtcachefriends=yes

So, those notifications allow for queue to display (In Use) etc, and
creates no warnings for other devices except Audiocodes gateway.

I wonder is there any way how to disable this message in Asterisk, or
make Audiocodes act correctly?

Below is the sip debug for this (xx.xx.xx.xx is Audiocodes,
yy.yy.yy.yy is Asterisk).

Regards,
Atis

-------------------------------------------------------------------------------------
[Apr 14 01:30:24] VERBOSE[19514] logger.c: Scheduling destruction of
SIP dialog '5239fdfb15d20e5153a2f7365ce2ea4d at yy.yy.yy.yy' in 32000 ms
(Method: NOTIFY)
[Apr 14 01:30:24] VERBOSE[19514] logger.c: Reliably Transmitting (NAT)
to xx.xx.xx.xx:5060:
NOTIFY sip:90170 at xx.xx.xx.xx SIP/2.0
Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK788fbefd;rport
From: "Unknown" <sip:Unknown at yy.yy.yy.yy>;tag=as436bf308
To: <sip:90170 at xx.xx.xx.xx>
Contact: <sip:Unknown at yy.yy.yy.yy>
Call-ID: 5239fdfb15d20e5153a2f7365ce2ea4d at yy.yy.yy.yy
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 92

Messages-Waiting: no
Message-Account: sip:asterisk at yy.yy.yy.yy
Voice-Message: 0/0 (0/0)

---
[Apr 14 01:30:24] VERBOSE[19514] logger.c:
<--- SIP read from xx.xx.xx.xx:5060 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK788fbefd;rport
From: "Unknown" <sip:Unknown at yy.yy.yy.yy>;tag=as436bf308
To: <sip:90170 at xx.xx.xx.xx>;tag=1c73477527
Call-ID: 5239fdfb15d20e5153a2f7365ce2ea4d at yy.yy.yy.yy
CSeq: 102 NOTIFY
Contact: <sip:xx.xx.xx.xx>
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Content-Length: 0


<------------->
[Apr 14 01:30:24] VERBOSE[19514] logger.c: --- (10 headers 0 lines) ---
[Apr 14 01:30:24] WARNING[19514] chan_sip.c: Remote host can't match
request NOTIFY to call '5239fdfb15d20e5153a2f7365ce2ea4d at yy.yy.yy.yy'.
Giving up.


--
Atis Lezdins,
VoIP Project Manager / Developer,
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

[greymanvoip at gmail.com]

For blind transfers Asterisk will send the call back to the dial plan
and into the TRANSFER (I think, could be a different name) context if
it exists. Within that context you can access the channel that was
answered on the original call using ${DIALEDPEERNUMBER}.

Note that this mechanism cannot be use for attended transfers as they
are not sent back to the dial plan for processing.

Regards,

Greyman.

[atis at iq-labs.net]

On Tue, Apr 22, 2008 at 3:15 PM, Grey Man <greymanvoip at gmail.com> wrote:

[oej at edvina.net]

22 apr 2008 kl. 13.56 skrev Atis Lezdins: