Asterisk Instant Messaging Errors - asterisk

Question

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.

Answer 1

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