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Cisco AS5300 + Asterisk 1.6: SDES ?

m.monsieur
Level 1
Level 1

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

4 Replies 4

Hi Mickael,

Could you post the complete trace if the call?

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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

<0032487997160>

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:

<65939191>

;tag=as12acaefb

To:

<0032487997160>

;tag=36CA05C-167B

Contact:

<65939191>

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

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:

<65939191>

;tag=as12acaefb

To:

<0032487997160>

;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

<0032487997160>

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:

<65939191>

;tag=as12acaefb

To:

<0032487997160>

;tag=36CA05C-167B

Contact:

<65939191>

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: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"

;tag=as4eb3efa7

To:

<10.4.0.10>

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

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"

;tag=as4eb3efa7

To:

<10.4.0.10>

;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

UDP:10.4.0.10:54336

--->

BYE

sip:65939191@10.4.0.1:5060

SIP/2.0

Via: SIP/2.0/UDP  10.4.0.10:5060

From:

<0032487997160>

;tag=36CA05C-167B

To:

<65939191>

;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:

<0032487997160>

;tag=36CA05C-167B

To:

<65939191>

;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)

Ok, i solved.

The problem is Asterisk :

https://issues.asterisk.org/jira/browse/ASTERISK-15787

Good to know

Please rate all useful posts
Favor calificar todos las respuestas útiles.
___________________________________________
LinkedIn Profile: do.linkedin.com/in/leosalcie
MDGDP, CCNA, CCNA Voice, CCNP Voice Certified

__________________________________________________
Please remember to rate useful posts clicking on the stars below.
LinkedIn Profile: do.linkedin.com/in/leosalcie