Asterisk on Raspberry Pi -- stops accepting inbound calls after a while until restart

Forums › VoIP › Asterisk on Raspberry Pi -- stops accepting inbound calls after a while until restart

breakaway

105 posts

Master Geek

#139048 25-Jan-2014 22:53

I'm having a bit of a play with my Raspberry Pi model B and the incredible PBX (link). It's working pretty well. Normally I run my PBX in a VMWare VM which has been working fine but thought I'd have a play with this.

It appears I've come across a strange issue. Asterisk stops accepting inbound calls after a period of inactivity (like 15-20 minutes). asterisk -r on verbose shows absolutely NO activity when attempting a call from my cell. I get a message saying "The party you are trying to reach is unavailable" on my cell.

When this fault is 'active', if I make an outbound call from an extension registered to my PBX, it works fine. Then, for a short period I can accept some calls. Also if I restart the system or simply restart the asterisk service using "restart amportal" over SSH, it seems to accept incoming calls for a short period.

It's almost as if asterisk isn't even listening for calls at all, and restarting/making an outbound call wakes it up for a short period.

Any ideas on what this might be?

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#974408 25-Jan-2014 23:39

Firewall / router 99% will be the issue

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kornflake

409 posts

Ultimate Geek

#974460 26-Jan-2014 07:57

maybe your registration timers are out, change them to 1800,or 3600 depending on what the system will allow.

kornflake

409 posts

Ultimate Geek

#974462 26-Jan-2014 07:59

annoying cell phone! I mean 180 or 360.

sbiddle

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#974466 26-Jan-2014 08:12

If making a call fixes it the problem is almost certainly your registrations timing out which is either your registration times or your firewall.

breakaway

105 posts

Master Geek

#974631 26-Jan-2014 14:45

I don't think it's the firewall since it works fine using PiaF which I previously ran as VMWare VM. Also my register settings are already at 180.

Here's my PEER registration settings (I copied them out of my PIAF box): port=5060

port=5060
dtmfmode=rfc2833
disallow=all
allow=ulaw&alaw&g722
registertimeout=20
regseconds=180
type=peer
fromuser=<REDACTED>; this is your VFX Number without the leading 0 i.e. 9950XXXX
host=pan.wxnz.net
insecure=invite,port
canreinvite=no
nat=yes
secret=<REDACTED>
username=<REDACTED>

Here's what the PBX shows when the fault is 'active':

incrediblepbx*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time
pan.wxnz.net:5060 N <REDACTED> 285 Registered Sun, 26 Jan 2014 14:44:52
1 SIP registrations.

The "reg time" it shows is always <5 minutes from the current time, but it's still not going. Will try reducing it further and report back.

breakaway

105 posts

Master Geek

#974650 26-Jan-2014 15:40

Ok that didn't work. But it appears to be a SIP trunk registration issue. When I run "sip reload" it starts working again.

sbiddle

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#974655 26-Jan-2014 15:54

I'm assuming your box is firewalled off and port 5060 isn't exposed to the internet?

breakaway

105 posts

Master Geek

#974657 26-Jan-2014 16:03

port 5060 is NATed from my external IP to my PBX's internal IP, with incoming connections only allowed from pan.wxnz.net.

coffeebaron

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#974674 26-Jan-2014 16:43

Change that firewall rule to the actual IP address 58.28.20.150

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sbiddle

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#974676 26-Jan-2014 16:44

DOes Asterisk still show registered when the issue happens?

And are you running pure Asterisk or a distro?

breakaway

105 posts

Master Geek

#974705 26-Jan-2014 17:15

coffeebaron -- am explicitly using the IP

sbiddle -- I'm running Incredible PBX 3.11.8 (followed this guide here -- http://nerdvittles.com/?p=8178)

I've also made sure Asterisk framework & all add-ons are up to date using the module manager.

sbiddle

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#974707 26-Jan-2014 17:19

Is DNS configured correctly on the Pi?

breakaway

105 posts

Master Geek

#974737 26-Jan-2014 18:46

Yes it is -- it's configured via DHCP and I've checked that resolv.conf contains my DNS server.

ubergeeknz

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Vocus

#974786 26-Jan-2014 19:54

Does your router have any SIP capability? I would suggest using a port other than 5060 on both asterisk and on the outside - maybe 25060 or something, and see what happens.

kornflake

409 posts

Ultimate Geek

#974821 26-Jan-2014 21:03

try changing the host to the IP address

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