[Asterisk-Users] SIP Retransmission Patch

Mark Spencer markster at digium.com
Sat Mar 29 09:13:48 MST 2003


Turn off reinvites and that will likely fix this.

Notice how the first invite is totally ignored, and then for some reason
the second gives us the 481.

Mark

On Sat, 29 Mar 2003, Luke Howard wrote:

>
> 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
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>




More information about the asterisk-users mailing list