[Asterisk-Users] SIP Retransmission Patch

Andre Bierwirth asterisk at kmb.de
Fri Mar 28 20:55:39 MST 2003


My Patch fixes only the 15Sec Channeldestroy bug, for incomming/outgoing
calls but your error is an other.

Andre

----- Original Message -----
From: "Luke Howard" <lukeh at PADL.COM>
To: <asterisk-users at lists.digium.com>
Sent: Saturday, March 29, 2003 3:59 AM
Subject: Re: [Asterisk-Users] SIP Retransmission Patch


>
> 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