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[asterisk-users] Recurring SIP problem with asterisk 11.6 & 11.7

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asterisk-03 at jeremyk...
Guest

Posted: Tue Nov 12, 2013 7:37 pm Post subject: [asterisk-users] Recurring SIP problem with asterisk 11.6 &a

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

asterisk-03 at jeremyk...
Guest

Posted: Tue Nov 12, 2013 8:11 pm Post subject: [asterisk-users] Recurring SIP problem with asterisk 11.6 &a

On 11/12/2013 7:37 PM, Jeremy Kister wrote:

Quote:

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.

--
_____________________________________________________________________
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duncan at e-simple.co.nz
Guest

Posted: Tue Nov 12, 2013 8:46 pm Post subject: [asterisk-users] Recurring SIP problem with asterisk 11.6 &a

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@jeremykister.com> wrote:

Quote:

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

--
_____________________________________________________________________
-- 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

asterisk-03 at jeremyk...
Guest

Posted: Tue Nov 12, 2013 8:50 pm Post subject: [asterisk-users] Recurring SIP problem with asterisk 11.6 &a

On 11/12/2013 8:46 PM, Duncan Turnbull wrote:

Quote:

Any chance DNS is dying about the same time the problem occurs

good idea, but I don't use DNS anywhere in Asterisk. well, except for
sip.conf:externhost. it's all IP addresses.

--
_____________________________________________________________________
-- 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:
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asterisk_list at earth...
Guest

Posted: Wed Nov 13, 2013 4:12 am Post subject: [asterisk-users] Recurring SIP problem with asterisk 11.6 &a

On Wednesday 13 November 2013, Jeremy Kister wrote:

Quote:

On 11/12/2013 8:46 PM, Duncan Turnbull wrote:

Quote:

Any chance DNS is dying about the same time the problem occurs

good idea, but I don't use DNS anywhere in Asterisk. well, except for
sip.conf:externhost. it's all IP addresses.

That doesn't matter. Without a working DNS server, Asterisk will fall over --
even if all your configuration files are using IP addresses and you have all
your own machines in /etc/hosts .

Probably the easiest thing to do is just install bind9 on your Asterisk box
(and make sure to put
nameserver 127.0.0.1
in its resolv.conf). BIND is pretty lightweight as servers go, so should not
impact badly on performance, especially if it's the only machine using it.

--
AJS

Answers come *after* questions.

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello

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webaccounts at jgoettg...
Guest

Posted: Wed Nov 13, 2013 4:44 am Post subject: [asterisk-users] Recurring SIP problem with asterisk 11.6 &a

Throwing in my my 2cents, I prefer dnsmasq, which is even lighter and Asterisk doesn't mind. As far as maintenance (firmware updates of phones, etc) goes, dnsmasq also offers TFTP and DHCP functionality. If nothing like that is running on a customer site, this is quite handy. Not that you can't do this with other packages, but dnsmasq is very easy to configure. -- _____________________________________________________________________ -- 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

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