Hello.
I am trying to connect door phone intercom Hikvision DS-KV8102-IP with Asterisk 13.1.0~dfsg-1.1ubuntu4.1.
In configure menu of this intercom is possible set only the ip address of SIP server. From sniffs of comunication bettwen intercom and asterisk I prepared one peer with following settings:
[againdevice]
fullname = Door intercom
username = againdevice
cid_number = 11
insecure=invite
qualify=yes
type=friend
host=0.0.0.0
allowguest=yes
allow=all
directmedia=yes
directrtpsetup=yes
progressinband=no
ignoresdpversion=yes
prematuremedia=yes
But intercom does not send m= attribute in SDP content of INVITE packet and the call is rejected with “SIP/2.0 488 Not acceptable here”.
<--- SIP read from UDP:192.168.35.222:5060 --->
INVITE sip:againdevice@192.168.35.116:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.35.222:5060;rport;branch=z9hG4bK1248230690
From: <sip:againdevice@192.168.35.222>;tag=1833694817
To: <sip:againdevice@192.168.35.116:5060>
Call-ID: 88674963
CSeq: 20 INVITE
Contact: <sip:againdevice@192.168.35.222:5060>
Content-Type: application/sdp
Max-Forwards: 70
User-Agent: eXosip/3.6.0
Subject: This is a call for conversation
Content-Length: 139
v=0
o=649788304 0 0 IN IP4 192.168.35.222
s=Talk session
c=IN IP4 192.168.35.222
t=0 0
a=doorFloor:6
a=responseType:0
a=doorType:0
<------------->
--- (12 headers 8 lines) ---
Sending to 192.168.35.222:5060 (no NAT)
Sending to 192.168.35.222:5060 (no NAT)
Using INVITE request as basis request - 88674963
Found peer 'againdevice' for 'againdevice' from 192.168.35.222:5060
== Using SIP RTP CoS mark 5
[Aug 15 09:18:04] WARNING[17232][C-00000005]: chan_sip.c:9970 process_sdp: Insufficient information for SDP (m= not found)
<--- Reliably Transmitting (NAT) to 192.168.35.222:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.35.222:5060;branch=z9hG4bK1248230690;received=192.168.35.222;rport=5060
From: <sip:againdevice@192.168.35.222>;tag=1833694817
To: <sip:againdevice@192.168.35.116:5060>;tag=as7ca216c6
Call-ID: 88674963
CSeq: 20 INVITE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '88674963' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:192.168.35.222:5060 --->
ACK sip:againdevice@192.168.35.116:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.35.222:5060;rport;branch=z9hG4bK1248230690
Route: <sip:againdevice@192.168.35.116:5060>
From: <sip:againdevice@192.168.35.222>;tag=1833694817
To: <sip:againdevice@192.168.35.116:5060>;tag=as7ca216c6
Call-ID: 88674963
CSeq: 20 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '88674963' Method: ACK
Really destroying SIP dialog '0a9455283166dcb47a86041118210bfa@127.0.0.1:5060' Method: OPTIONS
Is there any way to make call with this intercom?
Many thanks!