SIP trunk problem

vloose

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Dec 2, 2008
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#1
I have a T1 for in incoming call center. I am using a SIP trunk (Callcentric) for outbound calls for long distance cost savings.

If I lose connectivity to my DNS servers (DNS is external to my phone system network so If I lose internet connection I lose DNS) my internal phones become virtually unusable. Usually takes about 30sec for a call to connect from one local phone to the other.

If I disable my SIP trunk the phones start behaving properly when DNS isn't available. I was able to find some info confirming this behavior with Asterisk, but I haven't seen any solutions I like yet such as a installing a caching server on my phone box.

Any suggestions out there I might have missed?

Vaughn
 

Megabyte

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#2
Hi, do you have some DNS parameters configured in the phones ? those fhones are ipphones or soft? what brand?, are your phones and elastix server in the same subnet? are you getting ip address from DHCP server, Elastix or your router?
 

vloose

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#3
I'm using SIP pones, Polycom 430's to be exact. Some phones are on a LAN on the same subnet as one interface on the server, external phones are on different subnets connecting to the external interface of the server. Not concerned with external phones however because when I lose DNS connectivity it is usually because of internet connection failure and obviously external phones won't work.

I can ping the phones on the LAN from the server when I lose internet connection and I can call between extensions, You just have to wait about 30 sec for the call to hit the extension you are calling. Incoming calls on the T1 to an extension experience the same issue.

Again, I can fix the problem by disabling my SIP connection to Callcentric. I don't believe, however, this is a Callcentric specific problem.

Here is a little test I did. I removed all DNS entries in /etc/resolv.conf. With my SIP trunk to Callcentric enabled, calls between extensions take 30sec or more to ring. Same scenario with no DNS configured on the box, and Callcentric trunk disabled, calls work normally between extensions and incoming and outgoing on the T1.

I'm up for suggestions, but I'm not leaning towards a phone DNS issue like I think you might be initially as well did I initially. I don't want this to sound like criticism for you attempted help. I appreciate your questions and your attempt to help me.

Hopefully your questions and my response help clear up the issue a little better.
 

dicko

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#4
This is an ongoing problem with asterisk, if there is a sip trunk in the system it will check it even if it is not part of the call, use IP address if possible instead of dns resolution but perhaps better to setup dnsmasq on your Elastix box I believe most problems will be resolved (by this local caching dns server) (this forum is full of posts on this topic as to specifics).
 

Megabyte

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#5
Hi you can do the following

Create 2 boxes. Link both boxes using IAX. Make one of the boxs handles all your SIP
trunks (Box A). Make the other box (Box B) handles all your PSTN trunk and make sure
you comment out externhost from sip_general_custom.conf or sip_nat.conf in Box B.
Attach all your extensions to to box B
Direct all incoming SIP calls from Box A to Box B.
Direct all outgoing SIP calls from Box B to Box A.
Naturally you must create all the appropriate dial plans and routes to handles your
requirements in each of the boxes.
If for any reason internet is down, it will only effect Box A while Box B, where all your
extensions and PSTN trunks are will be spared. Of course if your router dies then all will
be dead.

You can use a basic computer configuration to do that.
 

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