[asterisk-users] Asterisk 1.6 + Cisco AS5300
Eduardo A Muñoz
eagmunoz at gmail.com
Thu Mar 7 08:52:30 CST 2013
Can u debug on AS ?
On Thu, Mar 7, 2013 at 9:20 AM, Mickael Monsieur
<mickael.monsieur at gmail.com> wrote:
> 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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>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
--
Eduardo A. Muñoz
GPG Key fingerprint = 175E 6AEB AD23 8EFE 0FC3 F558 9AB1 7885 40A4 ABBB
CCNA - CCNP
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