Re: [asterisk-users] SIP OPTIONS response from the peer is ignored - peer becomes UNREACHABLE

2010-03-25 Thread --[ UxBoD ]--
33ebd;rport
> From: "asterisk" ;tag=as5e2d8165
> To: ;tag=CD9ADC40-8CECD6C3
> CSeq: 102 OPTIONS
> Call-ID: 5bf37c2d2d25a9346cc272d71c3a8...@172.11.0.201
> Contact: 
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
> NOTIFY, PRACK, UPDATE, REFER
> User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.2.0.0047
> Content-Length: 0
> 
> 
> Hm, what could be causing such delay between Wireshark getting the
> data and Asterisk logging it?
> 
> Regards,
> Alex
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
> Howes
> Sent: Thursday, March 25, 2010 11:35 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] SIP OPTIONS response from the peer is
> ignored - peer becomes UNREACHABLE
> 
> On 25 Mar 2010, at 10:18, Asterisk wrote:
> > How is it possible that the peer becames UNREACHABLE eventhough
> Wireshark logged its proper response? 
> 
> Wireshark received it, doesn't mean Asterisk did. what does a sip
> debug in Asterisk show?
> 
> S

This is interesting as I am seeing the same issue with Snom 360s and M3s on 
Asterisk 1.6.1.14 and 1.6.2.6.  I have also received a report from a colleague 
who sees a similar issue with Polycoms on Microsoft OCS.
-- 
Thanks, Phil

-- 
_
-- 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


Re: [asterisk-users] SIP OPTIONS response from the peer is ignored - peer becomes UNREACHABLE

2010-03-25 Thread Asterisk
Hi Steve,

Yes, that's true. It seems that Asterisk gets it with great delay. For instance:

Asterisk says:
==

Mar 25 11:26:54 VERBOSE[1385] logger.c: Reliably Transmitting (no NAT) to 
172.11.11.2:5060:
OPTIONS sip:mytestph...@172.11.11.2 SIP/2.0
Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport
From: "asterisk" ;tag=as5e2d8165
To: 
Contact: 
Call-ID: 5bf37c2d2d25a9346cc272d71c3a8...@172.11.0.201
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 25 Mar 2010 10:26:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Mar 25 11:26:58 VERBOSE[1385] logger.c: Retransmitting #1 (no NAT) to 
172.11.11.2:5060:
OPTIONS sip:mytestph...@172.11.11.2 SIP/2.0
Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport
From: "asterisk" ;tag=as5e2d8165
To: 
Contact: 
Call-ID: 5bf37c2d2d25a9346cc272d71c3a8...@172.11.0.201
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 25 Mar 2010 10:26:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Mar 25 11:26:58 NOTICE[1385] chan_sip.c: Peer 'MyTestPhone' is now UNREACHABLE! 
 Last qualify: 52
Mar 25 11:26:58 VERBOSE[1385] logger.c: Destroying call 
'5bf37c2d2d25a9346cc272d71c3a8...@172.11.0.201'

=== WEIRD: Asterisk logs two responses reposnses then (last one at 11:27:02 
eventhough last OPTIONS request was sent by Asterisk at 11:26:58):

Mar 25 11:26:58 VERBOSE[1385] logger.c:
<-- SIP read from 172.11.11.2:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport
From: "asterisk" ;tag=as5e2d8165
To: ;tag=CD9ADC40-8CECD6C3
CSeq: 102 OPTIONS
Call-ID: 5bf37c2d2d25a9346cc272d71c3a8...@172.11.0.201
Contact: 
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, 
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.2.0.0047
Content-Length: 0


Mar 25 11:26:58 VERBOSE[1385] logger.c: --- (10 headers 0 lines) ---
Mar 25 11:27:02 VERBOSE[1385] logger.c: 
<-- SIP read from 172.11.11.2:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport
From: "asterisk" ;tag=as5e2d8165
To: ;tag=CD9ADC40-8CECD6C3
CSeq: 102 OPTIONS
Call-ID: 5bf37c2d2d25a9346cc272d71c3a8...@172.11.0.201
Contact: 
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, 
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.2.0.0047
Content-Length: 0


But Wireshark for the same conversation says:
=

>> Sent by Asterisk at 11:26:54.047483000
-
OPTIONS sip:mytestph...@172.11.11.2 SIP/2.0
Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport
From: "asterisk" ;tag=as5e2d8165
To: 
Contact: 
Call-ID: 5bf37c2d2d25a9346cc272d71c3a8...@172.11.0.201
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 25 Mar 2010 10:26:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<< Received from the phone at 11:26:54.097936000

SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport
From: "asterisk" ;tag=as5e2d8165
To: ;tag=CD9ADC40-8CECD6C3
CSeq: 102 OPTIONS
Call-ID: 5bf37c2d2d25a9346cc272d71c3a8...@172.11.0.201
Contact: 
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, 
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.2.0.0047
Content-Length: 0

>> Sent by Asterisk at 11:26:58.486339000
-
OPTIONS sip:mytestph...@172.11.11.2 SIP/2.0
Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport
From: "asterisk" ;tag=as5e2d8165
To: 
Contact: 
Call-ID: 5bf37c2d2d25a9346cc272d71c3a8...@172.11.0.201
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 25 Mar 2010 10:26:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<< Received from the phone at 11:26:58.524907000

SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport
From: "asterisk" ;tag=as5e2d8165
To: ;tag=CD9ADC40-8CECD6C3
CSeq: 102 OPTIONS
Call-ID: 5bf37c2d2d25a9346cc272d71c3a8...@172.11.0.201
Contact: 
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, 
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.2.0.0047
Content-Length: 0


Hm, what could be causing such delay between Wireshark getting the data and 
Asterisk logging it?

Regards,
Alex

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes
Sent: Thursday, March 25, 2010 11:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discu

Re: [asterisk-users] SIP OPTIONS response from the peer is ignored - peer becomes UNREACHABLE

2010-03-25 Thread Steve Howes
On 25 Mar 2010, at 10:18, Asterisk wrote:
> How is it possible that the peer becames UNREACHABLE eventhough Wireshark 
> logged its proper response? 

Wireshark received it, doesn't mean Asterisk did. what does a sip debug in 
Asterisk show?

S
-- 
_
-- 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