[asterisk-users] Handle Options Method

Thomas Deillon Thomas.Deillon at smart-telecom.ch
Tue Nov 21 05:21:38 MST 2006


Hi,

 

I have an Alteon in test (a sip/rtp load balancer).

This Alteon sends to the asterisk box a "SIP OPTIONS" to know if
asterisk is alive. 

However, asterisk sends me a 404 message and not a response like, for
example, a Thomson (200 + SDP)

 

I wrote a very little script (you can find it at the end of the email)
to send an Options message to asterisk/phones to try.

It works like this: ./Options.py eth0 192.168.1.35(ip of the UA) 

 

Do you know why asterisk send me a 404 message and how can I ask him to
answer correctly?

 

Thanks a lot for your help,

 

Thomas

 

Ps: for more information about the OPTION method:
http://www.ietf.org/rfc/rfc3261.txt ( page 66 and 67 )

 

 

Response of a Thomson:

/-----------------------------------------------------------------------
-----------------------

|           SIP/2.0 200 OK
|           Via: SIP/2.0/UDP 212.147.65.204:5060
|           From: "252"<sip:252 at 192.168.1.35:5060>
|           To: <sip:252 at 192.168.1.35:5060>;tag=c0a80101-23004a
|           Call-ID: 400842155 at 212.147.65.204
|           CSeq: 4660 OPTIONS
|           Contact: <sip:252 at 192.168.1.35:5060;user=phone>
|           Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGIST
ER,INFO
|           Supported: timer, replaces
|           Accept: application/sdp
|           Content-Type: application/sdp
|           Content-Length: 269
|
|           v=0
|           o=252 2293834 2293834 IN IP4 192.168.1.35
|           s=-
|           c=IN IP4 192.168.1.35
|           t=0 0
|           m=audio 41000 RTP/AVP 8 0 18 4 97
|           a=rtpmap:8 PCMA/8000
|           a=rtpmap:0 PCMU/8000
|           a=rtpmap:18 G729/8000
|           a=rtpmap:4 G723/8000
|           a=rtpmap:97 telephone-event/8000
|           a=fmtp:97 0-1
\-----------------------------------------------------------------------
-----------------------


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