I am running 1.8 and using Call-info header to cause phones to auto answer a page but when I try this using AMI Originate cmd I see that only the first (Originator) Invite gets the call-info header I specified and the others don’t - it this expected behavior or a bug?
Here is a simple Telnet version of my code that runs fine in that the 3 phones ring and can join the Meetme but only the Caller’s Invite has a call-Info header so the others do not auto answer.
Any ideas?
Bill
Working Telnet command:
Action: Originate
Channel: SIP/99906
Application: Page
Data: SIP/99903&SIP/99904,d
Variable: SIPAddHeader=Call-Info: sip:2.1.1.1;answer-after=0
Callerid: 999
Timeout: 30000
ActionID: ABC45678901234567895
The generated INVITES ( IP’s changed to protect the innocent)
Originator - call-info is present:
INVITE sip:99906@3.3.3.3:29564 SIP/2.0
Via: SIP/2.0/UDP 4.4.4.4:5060;branch=z9hG4bK63c13830;rport
Max-Forwards: 70
From: “999” sip:999@localhost;tag=as561eeb4e
To: sip:99906@3.3.3.3:29564
Contact: sip:999@4.4.4.4:5060
Call-ID: 1496d42b72f5e22c7b1d776905cf4d0c@localhost
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.24.0
Date: Thu, 16 Jan 2014 16:17:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Call-Info: sip:2.1.1.1;answer-after=0
Content-Type: application/sdp
Content-Length: 297
v=0
o=root 991050678 991050678 IN IP4 4.4.4.4
s=Asterisk PBX 1.8.24.0
c=IN IP4 4.4.4.4
t=0 0
m=audio 10966 RTP/AVP 0 18 4 110
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:110 speex/8000
a=ptime:20
a=sendrecv
To 99903 - Call-info is missing:
INVITE sip:99903@3.3.3.3:2476 SIP/2.0
Via: SIP/2.0/UDP 4.4.4.4:5060;branch=z9hG4bK1b3403aa;rport
Max-Forwards: 70
From: “999” sip:999@localhost;tag=as2fb9662e
To: sip:99903@3.3.3.3:2476
Contact: sip:999@4.4.4.4:5060
Call-ID: 53b9c9a95933b3bf1026439d6b0f1e73@localhost
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.24.0
Date: Thu, 16 Jan 2014 16:17:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 299
v=0
o=root 1549162942 1549162942 IN IP4 4.4.4.4
s=Asterisk PBX 1.8.24.0
c=IN IP4 4.4.4.4
t=0 0
m=audio 16236 RTP/AVP 0 18 4 110
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:110 speex/8000
a=ptime:20
a=sendrecv
To 99904 Call-info is missing
INVITE sip:99904@3.3.3.3:52117 SIP/2.0
Via: SIP/2.0/UDP 4.4.4.4:5060;branch=z9hG4bK67aa783a;rport
Max-Forwards: 70
From: “999” sip:999@localhost;tag=as176764fe
To: sip:99904@3.3.3.3:52117
Contact: sip:999@4.4.4.4:5060
Call-ID: 373de3d922500df15b528a5148dd1b8e@localhost
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.24.0
Date: Thu, 16 Jan 2014 16:17:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 299
v=0
o=root 1790266705 1790266705 IN IP4 4.4.4.4
s=Asterisk PBX 1.8.24.0
c=IN IP4 4.4.4.4
t=0 0
m=audio 15228 RTP/AVP 0 18 4 110
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:110 speex/8000
a=ptime:20
a=sendrecv