[asterisk-users] No reply to SIP OPTIONS - sip peers becoming randomly UNREACHABLE

[Aggio Alberto]

Hi,
I have occasionally experienced the same problem too, and I suspect it was caused by some spikes in network traffic (e.g. for an intensive file transfer) that delayed too much SIP OPTION response, so that Asterisk marked these devices as UNREACHABLE; I was able to use the devices too: in fact, the only drawback is that other devices are not able to call the UNREACHABLE devices using Asterisk. The only solution I found was to disable 'qualify' field in SIP account, in order to put these devices in unmonitored state. Maybe it's not your problem, but you can monitor the network with a sniffer (e.g. ethereal), in conjunction with SIP debug in Asterisk (sip set debug) in order to check the correct arrival of OPTION response.
Noevertheless, I'm wondering if there is another cause to this issue that is not depending on network, but on Asterisk itself, so let me know.


HTH,
cheers

Alberto.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Asterisk
Sent: lunedì 4 gennaio 2010 22.13
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] No reply to SIP OPTIONS - sip peers becoming randomly UNREACHABLE

Hi guys,

Am having a strange SIP problem in my call centre. The call centre has about 70 SIP agents (some of the are using SIP hard phones, other SIP softphones), and occasionally most of the SIP peers (hardphones and softphones) become UNREACHABLE and then after few second again REACHABLE. Some hardphones and softphones work perfectly normal during that period (even normally responding to OPTIONS message), but most of them get UNREACHABLE.

I don't have NAT - phones and Asterisk are in the same subnet, so nothing complicated really (regarding network configuration).

I'm currently suspecting my network to be the problem, but I would just like to confirm with you guys, if you have any similar experiences, what could be causing this?

Please, see bellow one of the sample SIP traces.

Regards,
Alex

Jan  1 11:17:42 VERBOSE[6046] logger.c: Reliably Transmitting (no NAT) to 165.11.1.41:5060:
OPTIONS sip:TestPhone1 at 165.11.1.41 SIP/2.0
Via: SIP/2.0/UDP 165.11.1.50:5060;branch=a4bG4bK4b7hf375;rport
From: "asterisk" <sip:asterisk at 165.11.1.50>;tag=as02e1afaa
To: <sip:TestPhone1 at 165.11.1.41>
Contact: <sip:asterisk at 165.11.1.50>
Call-ID: 3fa169320586bad01cd93bd87adf1c30 at 165.11.1.50
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 01 Jan 2010 11:17:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

Jan  1 11:17:45 VERBOSE[6046] logger.c: Retransmitting #1 (no NAT) to 165.11.1.41:5060:
OPTIONS sip:TestPhone1 at 165.11.1.41 SIP/2.0
Via: SIP/2.0/UDP 165.11.1.50:5060;branch=a4bG4bK4b7hf375;rport
From: "asterisk" <sip:asterisk at 165.11.1.50>;tag=as02e1afaa
To: <sip:TestPhone1 at 165.11.1.41>
Contact: <sip:asterisk at 165.11.1.50>
Call-ID: 3fa169320586bad01cd93bd87adf1c30 at 165.11.1.50
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 01 Jan 2010 11:17:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

Jan  1 11:17:46 NOTICE[6046] chan_sip.c: Peer 'TestPhone1' is now UNREACHABLE!  Last qualify: 14

Jan  1 11:17:56 VERBOSE[6046] logger.c: Reliably Transmitting (no NAT) to 165.11.1.41:5060:
OPTIONS sip:TestPhone1 at 165.11.1.41 SIP/2.0
Via: SIP/2.0/UDP 165.11.1.50:5060;branch=z2h16b637dKh2fd;rport
From: "asterisk" <sip:asterisk at 165.11.1.50>;tag=as796f6356
To: <sip:TestPhone1 at 165.11.1.41>
Contact: <sip:asterisk at 165.11.1.50>
Call-ID: 3367c4dc6cbdd57d67b0c5b53d5491e2 at 165.11.1.50
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 01 Jan 2010 11:17:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

Jan  1 11:17:56 VERBOSE[6046] logger.c: 
<-- SIP read from 165.11.1.41:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 165.11.1.50:5060;branch=z2h16b637dKh2fd;rport
From: "asterisk" <sip:asterisk at 165.11.1.50>;tag=as796f6356
To: <sip:TestPhone1 at 165.11.1.41>;tag=5A4BF5F8-460290A9
CSeq: 102 OPTIONS
Call-ID: 3367c4dc6cbdd57d67b0c5b53d5491e2 at 165.11.1.50
Contact: <sip:TestPhone1 at 165.11.1.41>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.2.0.0047
Content-Length: 0
Jan  1 11:17:56 VERBOSE[6046] logger.c: --- (10 headers 0 lines) ---
Jan  1 11:17:56 NOTICE[6046] chan_sip.c: Peer 'TestPhone1' is now REACHABLE! (16ms / 10000ms)

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