Jeremy Kister
2013-Nov-13 00:37 UTC
[asterisk-users] Recurring SIP problem with asterisk 11.6 & 11.7
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: # 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 any ideas how we can find out what's upset ?
Jeremy Kister
2013-Nov-13 01:11 UTC
[asterisk-users] Recurring SIP problem with asterisk 11.6 & 11.7
On 11/12/2013 7:37 PM, Jeremy Kister wrote:> any ideas how we can find out what's upset ?more info: when I create a /var/spool/asterisk/outgoing/callfile (with multiple SIP/xxx&SIP/yyy), the extensions ring. but when i answer with the handset the call does not connect and the other extensions continue ringing. if i am in the asterisk CLI while the phones are ringing, i can use 'sip show channels' and see the extensions in Init: INVITE. but if i use "channel request hangup <tab>" the session hangs. I can strace these hung rasterisk, but nothing's useful: # strace -p 25331 Process 25331 attached - interrupt to quit read(3, ^C <unfinished ...> Process 25331 detached # strace -p 26727 Process 26727 attached - interrupt to quit read(3, ^C <unfinished ...> Process 26727 detached # strace -p 26768 Process 26768 attached - interrupt to quit read(3, ^C <unfinished ...> Process 26768 detached the ringing eventually times out, but still no errors on the console.
Duncan Turnbull
2013-Nov-13 01:46 UTC
[asterisk-users] Recurring SIP problem with asterisk 11.6 & 11.7
Any chance DNS is dying about the same time the problem occurs I get this occasionally every 6-12 months and usually because DNS got messed up and then something didn?t fall back into place when it recovered - networking looks okay on the machine but asterisk is stuck. I have been meaning to follow it up, but usually just set domain names to IPs to avoid it Cheers Duncan On 13/11/2013, at 1:37 pm, Jeremy Kister <asterisk-03 at jeremykister.com> wrote:> 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: > # 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 > > any ideas how we can find out what's upset ? > > -- > _____________________________________________________________________ > -- 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