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pradeep kumar

2003-May-25 08:56 UTC

[Asterisk-Users] iconnecthere problem 481 "Call Leg/Transaction Does Not Exist"

Hi All,

I am trying to use iconnecthere to make outbound calls. I am behind a 
linksys router. I keep getting this error
481 "Call Leg/Transaction Does Not Exist". Does anyone have any prior 
experience with this problem. Any leads will be much appreciated. Attached 
are the conf files and logs

#SIP.CONF

; SIP Configuration for Asterisk
[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
context = from-sip

register => mylogin:mypassword@sipauth.deltathree.com/2905

[iconnect]
type=friend
username=mylogin
password=mypassword
host=natrelay.deltathree.com

#EXTENSIONS.conf
[general]
static=yes
writeprotect=no

[globals]

[intern]
include => iconnecthere

[iconnecthere]
;Outbound calls to PSTN numbers via iconnecthere.com's SIP service
exten => _61NXXNXXXXXX,1,Dial,SIP/${EXTEN-1}@iconnect
exten => _61NXXNXXXXXX,2,Congestion

#SIP DEBUG
*CLI> dial 616507148980@iconnecthere
*CLI> WARNING[1194325696]: File pbx.c, Line 821 
(pbx_substitute_variables_temp): The use of 'EXTEN-foo' has been
derprecated
in favor of 'EXTEN:foo'
    -- Executing Dial("OSS/dsp", "SIP/16507148980@iconnect")
in new stack
We're at 192.168.1.100 port 10860
Answering with capability 2
Answering with capability 4
Answering with capability 8
Answering with non-codec capability 1
10 headers, 11 lines
Reliably Transmitting:
INVITE sip:16507148980@natrelay.deltathree.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK41236072
From: "asterisk" <sip:asterisk@192.168.1.100>;tag=as476eacfd
To: <sip:16507148980@natrelay.deltathree.com>
Contact: <sip:asterisk@192.168.1.100>
Call-ID: 2a3fe9bc1d8dd93a400263c775c63f5b@192.168.1.100
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 3296 3296 IN IP4 192.168.1.100
s=session
c=IN IP4 192.168.1.100
t=0 0
m=audio 10860 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
(no NAT) to 213.137.73.140:5060
    -- Called 16507148980@iconnect
Retransmitting #1 (no NAT):
INVITE sip:16507148980@natrelay.deltathree.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK41236072
From: "asterisk" <sip:asterisk@192.168.1.100>;tag=as476eacfd
To: <sip:16507148980@natrelay.deltathree.com>
Contact: <sip:asterisk@192.168.1.100>
Call-ID: 2a3fe9bc1d8dd93a400263c775c63f5b@192.168.1.100
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 3296 3296 IN IP4 192.168.1.100
s=session
c=IN IP4 192.168.1.100
t=0 0
m=audio 10860 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

to 213.137.73.140:5060
Retransmitting #2 (no NAT):
INVITE sip:16507148980@natrelay.deltathree.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK41236072
From: "asterisk" <sip:asterisk@192.168.1.100>;tag=as476eacfd
To: <sip:16507148980@natrelay.deltathree.com>
Contact: <sip:asterisk@192.168.1.100>
Call-ID: 2a3fe9bc1d8dd93a400263c775c63f5b@192.168.1.100
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 3296 3296 IN IP4 192.168.1.100
s=session
c=IN IP4 192.168.1.100
t=0 0
m=audio 10860 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

to 213.137.73.140:5060
Retransmitting #3 (no NAT):
INVITE sip:16507148980@natrelay.deltathree.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK41236072
From: "asterisk" <sip:asterisk@192.168.1.100>;tag=as476eacfd
To: <sip:16507148980@natrelay.deltathree.com>
Contact: <sip:asterisk@192.168.1.100>
Call-ID: 2a3fe9bc1d8dd93a400263c775c63f5b@192.168.1.100
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 3296 3296 IN IP4 192.168.1.100
s=session
c=IN IP4 192.168.1.100
t=0 0
m=audio 10860 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

to 213.137.73.140:5060
Retransmitting #4 (no NAT):
INVITE sip:16507148980@natrelay.deltathree.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK41236072
From: "asterisk" <sip:asterisk@192.168.1.100>;tag=as476eacfd
To: <sip:16507148980@natrelay.deltathree.com>
Contact: <sip:asterisk@192.168.1.100>
Call-ID: 2a3fe9bc1d8dd93a400263c775c63f5b@192.168.1.100
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 3296 3296 IN IP4 192.168.1.100
s=session
c=IN IP4 192.168.1.100
t=0 0
m=audio 10860 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

to 213.137.73.140:5060
Retransmitting #5 (no NAT):
INVITE sip:16507148980@natrelay.deltathree.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK41236072
From: "asterisk" <sip:asterisk@192.168.1.100>;tag=as476eacfd
To: <sip:16507148980@natrelay.deltathree.com>
Contact: <sip:asterisk@192.168.1.100>
Call-ID: 2a3fe9bc1d8dd93a400263c775c63f5b@192.168.1.100
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 3296 3296 IN IP4 192.168.1.100
s=session
c=IN IP4 192.168.1.100
t=0 0
m=audio 10860 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

to 213.137.73.140:5060
WARNING[1125329600]: File chan_sip.c, Line 407 (retrans_pkt): Maximum 
retries exceeded on call 2a3fe9bc1d8dd93a400263c775c63f5b@192.168.1.100 for 
seqno 102 (Request)
  == No one is available to answer at this time
Reliably Transmitting:
CANCEL sip:16507148980@natrelay.deltathree.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK41236072
From: "asterisk" <sip:asterisk@192.168.1.100>;tag=as476eacfd
To: <sip:16507148980@natrelay.deltathree.com>
Contact: <sip:asterisk@192.168.1.100>
Call-ID: 2a3fe9bc1d8dd93a400263c775c63f5b@192.168.1.100
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

(no NAT) to 213.137.73.140:5060
    -- Executing Congestion("OSS/dsp", "") in new stack
Sip read:
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK41236072
From: "asterisk" <sip:asterisk@192.168.1.100>;tag=as476eacfd
To: sip:16507148980@192.168.1.100
Call-ID: 2a3fe9bc1d8dd93a400263c775c63f5b@192.168.1.100
CSeq: 102 CANCEL
Content-Length: 0


7 headers, 0 lines
    -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back
from
213.137.73.140


Thanks and sorry about the lengthy mail.

Pradeep

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Brian Capouch

2003-May-26 00:31 UTC

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[Asterisk-Users] iconnecthere problem 481 "Call Leg/Transaction Does Not Exist"

pradeep kumar wrote:> Hi All,
> 
> I am trying to use iconnecthere to make outbound calls. I am behind a 
> linksys router. I keep getting this error
> 481 "Call Leg/Transaction Does Not Exist". Does anyone have any
prior
> experience with this problem. Any leads will be much appreciated. 
> Attached are the conf files and logs
What version of asterisk?

This was a persistent problem months ago, but was fixed in some major 
code stuff that was done in March.

Not to say it may not have resurfaced. .

B.

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asterisk users - May 2003 - iconnecthere problem 481 "Call Leg/Transaction Does Not Exist"

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[Asterisk-Users] iconnecthere problem 481 "Call Leg/Transaction Does Not Exist"

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