[asterisk-users] Asterisk 1.6 + Cisco AS5300

Thu Mar 7 07:50:42 CST 2013

Le 7/03/13 11:21, Steven Howes a écrit :
> On 7 Mar 2013, at 10:12, Mickael Monsieur wrote:
>> Do you have an explanation?
> Put a SIP debug on and we may be able to find one..
>
> Steve
Hello Steve,
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 <sip:0032487997160 at 10.4.0.10:5060> 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 at 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: <sip:65939191 at 10.4.0.1>;tag=as12acaefb
To: <sip:0032487997160 at 10.4.0.10>;tag=36CA05C-167B
Contact: <sip:65939191 at 10.4.0.1>
Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03 at 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 UDP:10.4.0.10:5060 --->
SIP/2.0 420 Bad Extension
Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport
From: <sip:65939191 at 10.4.0.1>;tag=as12acaefb
To: <sip:0032487997160 at 10.4.0.10>;tag=36CA05C-167B
Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03 at 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 <sip:0032487997160 at 10.4.0.10:5060> 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 at 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: <sip:65939191 at 10.4.0.1>;tag=as12acaefb
To: <sip:0032487997160 at 10.4.0.10>;tag=36CA05C-167B
Contact: <sip:65939191 at 10.4.0.1>
Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03 at 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:10.4.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK4d8b5654;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.4.0.1>;tag=as4eb3efa7
To: <sip:10.4.0.10>
Contact: <sip:asterisk at 10.4.0.1>
Call-ID: 6a43ad4b27d870d048e8425077bcc075 at 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 UDP:10.4.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK4d8b5654;rport
From: "asterisk" <sip:asterisk at 10.4.0.1>;tag=as4eb3efa7
To: <sip:10.4.0.10>;tag=37A724C-211C
Date: Sat, 01 Jan 2000 16:12:32 GMT
Call-ID: 6a43ad4b27d870d048e8425077bcc075 at 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 at 10.4.0.1' 
Method: OPTIONS

<--- SIP read from UDP:10.4.0.10:54336 --->
BYE sip:65939191 at 10.4.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP  10.4.0.10:5060
From: <sip:0032487997160 at 10.4.0.10>;tag=36CA05C-167B
To: <sip:65939191 at 10.4.0.1>;tag=as12acaefb
Date: Sat, 01 Jan 2000 16:12:26 GMT
Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03 at 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: <sip:0032487997160 at 10.4.0.10>;tag=36CA05C-167B
To: <sip:65939191 at 10.4.0.1>;tag=as12acaefb
Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03 at 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)


>
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