[asterisk-bugs] [JIRA] (ASTERISK-22851) Asterisk/SIP stops responding
Jeremy Kister (JIRA)
noreply at issues.asterisk.org
Tue Nov 12 23:18:03 CST 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-22851?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Jeremy Kister updated ASTERISK-22851:
-------------------------------------
Description:
I have regularly (once a week, once per few hundred calls?) been having
problems with Asterisk's SIP stack not responding to packets from any of
my registered devices. In the past, I could not tolerate the outage, so
i would restart asterisk to make things happy.
My Asterisk server is currently in this broken state and I can leave it
this way for a short while. Devices are registered to it and I can 'sip
qualify peer xxx':
{code}
pbx1*CLI> sip qualify peer 111
[Nov 13 00:16:33] NOTICE[27681]: chan_sip.c:29489 sip_poke_peer: Still have a QUALIFY dialog active, deleting
Really destroying SIP dialog '0513566165179f7a3e2ccbd64424e614 at 10.9.1.3:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 10.9.1.111:5060:
OPTIONS sip:111 at 10.9.1.111:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.9.1.3:5060;branch=z9hG4bK276d50c3;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.9.1.3>;tag=as7e156001
To: <sip:111 at 10.9.1.111:5060;transport=udp>
Contact: <sip:asterisk at 10.9.1.3:5060>
Call-ID: 07d95a2a12b438373c441a227877202f at 10.9.1.3:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0-rc1
Date: Wed, 13 Nov 2013 05:16:33 GMT
Session-Expires: 7200
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
{code}
on the network, this shows:
'sip show peer xxx' all show Status OK.
but whenever one of the devices tries to make a new call, Asterisk just
doesnt respond. 'sip set debug on' shows no packets.
from the asterisk server (10.1.0.3), i can see one of my phones
(10.1.0.111) trying to make a call:
{code}
# tcpdump -i eth0 -s 0 -t -n host 10.1.0.111
ARP, Request who-has 10.1.0.3 tell 10.1.0.111, length 46
ARP, Reply 10.1.0.3 is-at 00:0c:29:07:39:8e, length 28
IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
IP 10.1.0.111.123 > 10.1.0.3.123: NTPv3, Client, length 48
IP 10.1.0.3.123 > 10.1.0.111.123: NTPv3, Server, length 48
IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
IP 10.1.0.111.123 > 10.1.0.3.123: NTPv3, Client, length 48
IP 10.1.0.3.123 > 10.1.0.111.123: NTPv3, Server, length 48
ARP, Request who-has 10.1.0.111 tell 10.1.0.3, length 28
ARP, Reply 10.1.0.111 is-at 00:13:c4:01:da:4a, length 46
IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
{code}
was:
I have regularly (once a week, once per few hundred calls?) been having
problems with Asterisk's SIP stack not responding to packets from any of
my registered devices. In the past, I could not tolerate the outage, so
i would restart asterisk to make things happy.
My Asterisk server is currently in this broken state and I can leave it
this way for a short while. Devices are registered to it and I can 'sip
qualify peer xxx'. 'sip show peer xxx' all show Status OK.
but whenever one of the devices tries to make a new call, Asterisk just
doesnt respond. 'sip set debug on' shows no packets.
from the asterisk server (10.1.0.3), i can see one of my phones
(10.1.0.111) trying to make a call:
{code}
# tcpdump -i eth0 -s 0 -t -n host 10.1.0.111
ARP, Request who-has 10.1.0.3 tell 10.1.0.111, length 46
ARP, Reply 10.1.0.3 is-at 00:0c:29:07:39:8e, length 28
IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
IP 10.1.0.111.123 > 10.1.0.3.123: NTPv3, Client, length 48
IP 10.1.0.3.123 > 10.1.0.111.123: NTPv3, Server, length 48
IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
IP 10.1.0.111.123 > 10.1.0.3.123: NTPv3, Client, length 48
IP 10.1.0.3.123 > 10.1.0.111.123: NTPv3, Server, length 48
ARP, Request who-has 10.1.0.111 tell 10.1.0.3, length 28
ARP, Reply 10.1.0.111 is-at 00:13:c4:01:da:4a, length 46
IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
{code}
> Asterisk/SIP stops responding
> -----------------------------
>
> Key: ASTERISK-22851
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-22851
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/General
> Affects Versions: 11.6.0, 11.7.0
> Environment: Debian6
> Reporter: Jeremy Kister
> Severity: Blocker
>
> I have regularly (once a week, once per few hundred calls?) been having
> problems with Asterisk's SIP stack not responding to packets from any of
> my registered devices. In the past, I could not tolerate the outage, so
> i would restart asterisk to make things happy.
> My Asterisk server is currently in this broken state and I can leave it
> this way for a short while. Devices are registered to it and I can 'sip
> qualify peer xxx':
> {code}
> pbx1*CLI> sip qualify peer 111
> [Nov 13 00:16:33] NOTICE[27681]: chan_sip.c:29489 sip_poke_peer: Still have a QUALIFY dialog active, deleting
> Really destroying SIP dialog '0513566165179f7a3e2ccbd64424e614 at 10.9.1.3:5060' Method: OPTIONS
> Reliably Transmitting (NAT) to 10.9.1.111:5060:
> OPTIONS sip:111 at 10.9.1.111:5060;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 10.9.1.3:5060;branch=z9hG4bK276d50c3;rport
> Max-Forwards: 70
> From: "asterisk" <sip:asterisk at 10.9.1.3>;tag=as7e156001
> To: <sip:111 at 10.9.1.111:5060;transport=udp>
> Contact: <sip:asterisk at 10.9.1.3:5060>
> Call-ID: 07d95a2a12b438373c441a227877202f at 10.9.1.3:5060
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX 11.7.0-rc1
> Date: Wed, 13 Nov 2013 05:16:33 GMT
> Session-Expires: 7200
> Min-SE: 90
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> Content-Length: 0
> {code}
> on the network, this shows:
> 'sip show peer xxx' all show Status OK.
> but whenever one of the devices tries to make a new call, Asterisk just
> doesnt respond. 'sip set debug on' shows no packets.
> from the asterisk server (10.1.0.3), i can see one of my phones
> (10.1.0.111) trying to make a call:
> {code}
> # tcpdump -i eth0 -s 0 -t -n host 10.1.0.111
> ARP, Request who-has 10.1.0.3 tell 10.1.0.111, length 46
> ARP, Reply 10.1.0.3 is-at 00:0c:29:07:39:8e, length 28
> IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
> IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
> IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
> IP 10.1.0.111.123 > 10.1.0.3.123: NTPv3, Client, length 48
> IP 10.1.0.3.123 > 10.1.0.111.123: NTPv3, Server, length 48
> IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
> IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
> IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
> IP 10.1.0.111.123 > 10.1.0.3.123: NTPv3, Client, length 48
> IP 10.1.0.3.123 > 10.1.0.111.123: NTPv3, Server, length 48
> ARP, Request who-has 10.1.0.111 tell 10.1.0.3, length 28
> ARP, Reply 10.1.0.111 is-at 00:13:c4:01:da:4a, length 46
> IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
> IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
> IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
> {code}
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