[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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