[Asterisk-Users] SIP Retransmission Patch

[Luke Howard]

This seems to fix incoming calls but outgoing calls terminate 
immediately, at least for me, with a

481 "Call Leg/Transaction Does Not Exist"

from the SIP phone. Here's the SIP debug output (NB: IP addresses
have been changed). It *used* to work.

-- Luke

    -- Attempting native bridge of SIP/515-Office-9b81 and SIP/iconnecthere-adae
Sip read: 
ACK sip:918006822878 at 1.2.3.65 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.85:5060;branch=z9hG4bK-6rm2uxcnwrui
Max-Forwards: 70
From: "PADL Software Pty Ltd" <sip:515-Office at voip.padl.net>;tag=t1yl4zosmb
To: <sip:918006822878 at voip.padl.net;user=phone>;tag=138fec7f
Call-ID: 3c2b705a2c8b-v61ehcrwkusc at 1.2.3.85
CSeq: 1 ACK
Contact: <sip:515-Office at 1.2.3.85:5060;line=1>
Content-Length: 0


9 headers, 0 lines
Sip read: 
OPTIONS sip:918006822878 at 1.2.3.65 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.85:5060;branch=z9hG4bK-qbu1ryvkb09t
Max-Forwards: 70
From: "PADL Software Pty Ltd" <sip:515-Office at voip.padl.net>;tag=t1yl4zosmb
To: <sip:918006822878 at voip.padl.net;user=phone>;tag=138fec7f
Call-ID: 3c2b705a2c8b-v61ehcrwkusc at 1.2.3.85
CSeq: 2 OPTIONS
Contact: <sip:515-Office at 1.2.3.85:5060;line=1>
Accept: application/sdp
Content-Length: 0


10 headers, 0 lines
Looking for 918006822878 in local
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 1.2.3.85:5060;branch=z9hG4bK-qbu1ryvkb09t
From: "PADL Software Pty Ltd" <sip:515-Office at voip.padl.net>;tag=t1yl4zosmb
To: <sip:918006822878 at voip.padl.net;user=phone>;tag=138fec7f
Call-ID: 3c2b705a2c8b-v61ehcrwkusc at 1.2.3.85
CSeq: 2 OPTIONS
User-Agent: Asterisk PBX
Contact: <sip:918006822878 at 1.2.3.65>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Accept: application/sdp
Content-Length: 0


 to 1.2.3.85:5060
We're at 1.2.3.65 port 19540
Answering with preferred capability 4
Answering with preferred capability 2
Answering with non-codec capability 1
Reliably Transmitting:
INVITE sip:918006822878 at voip.padl.net;user=phone SIP/2.0
Via: SIP/2.0/UDP 1.2.3.65:5060;branch=59885c3a
From: "PADL Software Pty Ltd" <sip:515-Office at voip.padl.net>;tag=t1yl4zosmb
To: <sip:918006822878 at voip.padl.net;user=phone>;tag=138fec7f
Call-ID: 3c2b705a2c8b-v61ehcrwkusc at 1.2.3.85
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 1271 1271 IN IP4 213.137.65.237
s=session
c=IN IP4 213.137.65.237
t=0 0
m=audio 18142 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
 (no NAT) to 1.2.3.85:5060
We're at 1.2.3.65 port 64016
Answering with preferred capability 4
Reliably Transmitting:
INVITE sip:18006822878 at 213.137.73.178 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.65:5060;branch=385d81d8
From: "PADL Software Pty Ltd" <sip:asterisk at 1.2.3.65>;tag=3fc4e8e1
To: <sip:18006822878 at 213.137.73.178>;tag=6823c85e-40d69d11
Call-ID: 64ad99a24e3f695d5f3ef63e05edb8bc at 1.2.3.65
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 131

v=0
o=root 1271 1271 IN IP4 1.2.3.85
s=session
c=IN IP4 1.2.3.85
t=0 0
m=audio 10178 RTP/AVP 0
a=rtpmap:0 PCMU/8000
 (no NAT) to 213.137.73.178:5060
Sip read: 
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 1.2.3.65:5060;branch=59885c3a
From: "PADL Software Pty Ltd" <sip:515-Office at voip.padl.net>;tag=t1yl4zosmb
To: <sip:918006822878 at voip.padl.net;user=phone>;tag=138fec7f
CSeq: 102 INVITE
Call-ID: 3c2b705a2c8b-v61ehcrwkusc at 1.2.3.85
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE
Supported: timer, 100rel, replaces
Content-Length: 0


9 headers, 0 lines
    -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 1.2.3.85
--
Luke Howard | PADL Software Pty Ltd | www.padl.com