hi,
Does asterisk supports SIP Messaging Protocol using MESSAGE method.
Can we send Instant message using Asterisk.
If so , could you please tell me which version of Asterisk supports SIP Message.
When i tried in forums, I found there is a patch for this and tried with that.
But i am not able to send message.
Is anyone successful sending messages by applying patch
Thanks
krishnaveni.
I looked at the link you gave, and there is nothing on that web page regarding the MESSAGE method.
Exactly how do we get Asterisk to support the MESSAGE method?
Thanks,
Paul
I have sent MESSAGE requests through Asterisk and they were successfully received by the other party.
Here is a trace I have got:
<— SIP read from 10.8.155.27:12060 —>
MESSAGE sip:102@10.8.155.27 SIP/2.0
Via: SIP/2.0/UDP 10.8.155.27:12060;rport;branch=z9hG4bK5449
From: sip:web@10.8.155.27;tag=27565
To: sip:102@10.8.155.27;tag=as7ec5c4b8
Call-ID: 23466@10.8.155.27
CSeq: 25 MESSAGE
Max-Forwards: 70
User-Agent: Linphone-1.7.0/eXosip
Content-Type: text/plain
Content-Length: 16
session_id=100
<------------->
— (10 headers 1 lines) —
Receiving message!
Message received: ‘session_id=100
’
<— Transmitting (no NAT) to 10.8.155.27:12060 —>
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 10.8.155.27:12060;branch=z9hG4bK5449;received=10.8.155.27;rport=12060
From: sip:web@10.8.155.27;tag=27565
To: sip:102@10.8.155.27;tag=as7ec5c4b8
Call-ID: 23466@10.8.155.27
CSeq: 25 MESSAGE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:102@10.8.155.27
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘23466@10.8.155.27’ in 32000 ms (Method: MESSAGE)
Sending text session_id=100
on SIP/user2-087dabb0
Really sending text session_id=100
on SIP/user2-087dabb0
set_destination: Parsing sip:user2@10.8.155.27:10060;transport=UDP for address/port to send to
set_destination: set destination to 10.8.155.27, port 10060
Reliably Transmitting (no NAT) to 10.8.155.27:10060:
MESSAGE sip:user2@10.8.155.27:10060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.8.155.27:5060;branch=z9hG4bK542bfe41;rport
From: “web” sip:web@10.8.155.27;tag=as3fdd85c2
To: sip:user2@10.8.155.27:10060;transport=UDP;tag=0cca000c67458b6bc623
Call-ID: 40139002476d2c99683e34d1220e011d@10.8.155.27
CSeq: 103 MESSAGE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Type: text/plain
Content-Length: 15
session_id=100
<— SIP read from 10.8.155.27:10060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.155.27:5060;rport=5060;received=10.8.155.27;branch=z9hG4bK542bfe41
Call-ID: 40139002476d2c99683e34d1220e011d@10.8.155.27
From: “web” sip:web@10.8.155.27;tag=as3fdd85c2
To: sip:user2@10.8.155.27;tag=0cca000c67458b6bc623
CSeq: 103 MESSAGE
Content-Length: 0
You can try this and see.
The trouble I got is that the call was automatically destroyed by Asterisk itself which is what I didn’t want.