â03-07-2013 02:20 AM - edited â03-16-2019 04:07 PM
Hello,
I have a Cisco AS5300 connected to Asterisk (1.6.2.9)
Between 15-16 minutes, the call is disconnected without reason.
Here is what is displayed in the Asterisk debug:
Received an SDES from 10.4.0.10:17399
-- Got SIP response 420 "Bad Extension" back from 10.4.0.10
-- Stopped music on hold on SIP/as5300-1-0000004d
== Spawn extension (dialin, 065939191, 2) exited non-zero on 'SIP/as5300-1-0000004d'
Do you have an explanation?
Best regards,
Mickael
â03-07-2013 04:54 AM
Hi Mickael,
Could you post the complete trace if the call?
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MDGDP, CCNA, CCNA Voice, CCNP Voice Certified
â03-07-2013 05:52 AM
After checking, I confirm that the call is cut precisely to 900 seconds (15 min).
10.4.0.1 = Asterisk
10.4.0.10 = Cisco AS 5300
Info : debug start at 14min30sec
set_destination: Parsing
for address/port to send to
set_destination: set destination to 10.4.0.10, port 5060
Audio is at 10.4.0.1 port 11842
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (NAT) to 10.4.0.10:54789:
INVITE
sip:0032487997160@10.4.0.10:5060
SIP/2.0
Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport
Max-Forwards: 70
From:
;tag=as12acaefb
To:
;tag=36CA05C-167B
Contact:
Call-ID:
FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10
CSeq: 102 INVITE
User-Agent: isdnbox1.1
Require: timer
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (Session-Timers)
Content-Type: application/sdp
Content-Length: 207
v=0
o=root 1538728127 1538728127 IN IP4 10.4.0.1
s=Asterisk PBX 1.6.2.9-2+squeeze8
c=IN IP4 10.4.0.1
t=0 0
m=audio 11842 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
---
<--- SIP read from
--->
SIP/2.0 420 Bad Extension
Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport
From:
;tag=as12acaefb
To:
;tag=36CA05C-167B
Call-ID:
FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10
CSeq: 102 INVITE
Unsupported: timer
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
-- Got SIP response 420 "Bad Extension" back from 10.4.0.10
set_destination: Parsing
for address/port to send to
set_destination: set destination to 10.4.0.10, port 5060
Transmitting (NAT) to 10.4.0.10:5060:
ACK
sip:0032487997160@10.4.0.10:5060
SIP/2.0
Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport
Max-Forwards: 70
From:
;tag=as12acaefb
To:
;tag=36CA05C-167B
Contact:
Call-ID:
FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10
CSeq: 102 ACK
User-Agent: isdnbox1.1
Content-Length: 0
---
-- Stopped music on hold on SIP/as5300-1-00000050
== Spawn extension (dialin, 065939191, 2) exited non-zero on 'SIP/as5300-1-00000050'
Reliably Transmitting (NAT) to 10.4.0.10:5060:
OPTIONS
SIP/2.0
Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK4d8b5654;rport
Max-Forwards: 70
From: "asterisk"
;tag=as4eb3efa7
To:
Contact:
Call-ID:
6a43ad4b27d870d048e8425077bcc075@10.4.0.1
CSeq: 102 OPTIONS
User-Agent: isdnbox1.1
Date: Thu, 07 Mar 2013 11:17:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from
--->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK4d8b5654;rport
From: "asterisk"
;tag=as4eb3efa7
To:
;tag=37A724C-211C
Date: Sat, 01 Jan 2000 16:12:32 GMT
Call-ID:
6a43ad4b27d870d048e8425077bcc075@10.4.0.1
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
CSeq: 102 OPTIONS
Supported: 100rel
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Accept: application/sdp
Allow-Events: telephone-event
Content-Length: 154
v=0
o=CiscoSystemsSIP-GW-UserAgent 793 5073 IN IP4 10.4.0.10
s=SIP Call
c=IN IP4 10.4.0.10
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15 3
c=IN IP4 10.4.0.10
<------------->
--- (14 headers 7 lines) ---
Really destroying SIP dialog '
6a43ad4b27d870d048e8425077bcc075@10.4.0.1
' Method: OPTIONS
<--- SIP read from
--->
BYE
SIP/2.0
Via: SIP/2.0/UDP 10.4.0.10:5060
From:
;tag=36CA05C-167B
To:
;tag=as12acaefb
Date: Sat, 01 Jan 2000 16:12:26 GMT
Call-ID:
FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 6
Timestamp: 946743153
CSeq: 102 BYE
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- Transmitting (NAT) to 10.4.0.10:54336 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 10.4.0.10:5060;received=10.4.0.10
From:
;tag=36CA05C-167B
To:
;tag=as12acaefb
Call-ID:
FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10
CSeq: 102 BYE
Server: isdnbox1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
15 min (call ended)
â03-07-2013 07:01 AM
â03-07-2013 08:59 AM
Good to know
Please rate all useful posts
Favor calificar todos las respuestas Ăștiles.
___________________________________________
LinkedIn Profile: do.linkedin.com/in/leosalcie
MDGDP, CCNA, CCNA Voice, CCNP Voice Certified
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