I was trying to setup instant messaging in asterisk server. For the client i'm using Blink Softphone. I did add to my sip.conf
[general]
accept_outofcall_message=yes
outofcall_message_context=dialplan_name
auth_message_requests=yes
and to my extensions.conf
[dialplan_name]
exten => _XXX,1,MessageSend(sip:${EXTEN},"${CALLERID(name)}"${MESSAGE(from)})
So this is a simple extension for testing. But when i try and send the message from user1 to user2 i get in the asterisk log:
[Jan 30 21:17:47] WARNING[6420][C-00000005]: chan_sip.c:10515 process_sdp: Insufficient information in SDP (c=)...
What can be wrong here? I'am sure that the clients are on the nat, so i do have both nat=force_rport,comedia for all my users in sip.conf
My asterisk version is 13 (latest).
[UPDATED] I turned on the sip debug log, and tried to send the message (first i get some weird retransmission):
Retransmitting #9 (no NAT) to 14.228.14.150:5070: <- weird ip here
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 14.228.14.150:5070;branch=z9hA4cA-abcdef613aca8b1x3124abb0z3e1bc8;received=14.228.14.150;rport=5070
From: 2014<sip:2014@(server_ip_here)>;tag=a312facc
To: 0009735466221178<sip:0009735466221178@(server_ip_here)>;tag=bb62441233
Call-ID: abcdef613aca8b1x3124abb0z3e1bc8
CSeq: 1 INVITE
Server: Asterisk PBX 13.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2222abcf"
Content-Length: 0
Then
<--- SIP read from TLS:(my_ip_here):49312 --->
INVITE sip:123@(server_ip_here) SIP/2.0
Via: SIP/2.0/TLS (my_ip_here):49312;rport;branch=b4ae88b115be15a5n9244;alias
Max-Forwards: 70
From: "user1" <sip:user1@server_ip_here>;tag=39e388fd5f616b7
To: <sip:123@(server_ip_here)>
Contact: <sip:12313560@(server_ip_here):49311;transport=tls>
Call-ID: ac48128192za12a2f432e24c18aabc7q
CSeq: 28982 INVITE
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Supported: 100rel, replaces, norefersub, gruu
User-Agent: Blink 0.9.1.2 (Windows)
Content-Type: application/sdp
Content-Length: 308
v=0
o=- 3211422221 3211422221 IN IP4 (my_ip)
s=Blink 0.9.1.2 (Windows)
t=0 0
m=message 2855 TCP/TLS/MSRP *
c=IN IP4 (my_ip)
a=path:msrps://(my_ip):2855/37a1cc82ab315e21b222;tcp
a=accept-types:message/cpim text/* application/im-iscomposing+xml
a=accept-wrapped-types:*
a=setup:active
<------------->
--- (13 headers 10 lines) ---
Sending to (my_ip):49312 (NAT)
Sending to (my_ip):49312 (NAT)
Using INVITE request as basis request - ad72cd1681aff769721af12c21aaea7c
Found peer 'user1' for 'user1' from (my_ip):49312
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
[Jan 31 21:02:17] WARNING[27265][C-00000118]: chan_sip.c:10515 process_sdp: Insufficient information in SDP (c=)...
arheops was right! Using INVITE request as basis request <- here it is. Instead of MESSAGE for some reason INVITE request is used.
Looks like your softphone use INVITE instead of MESSAGE for messaging.
You can get more info by enable sip debug in asterisk console
asterisk -r
sip set debug on