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david3
join:2000-03-21

david3

Member

[CallCentric] Callcentric and Asterisk Config Question / Rejecte

I've searched this forum and found this brought up a few times before, but I'm still not sure I understand it completely.

I've been adding some failover stuff to my asterisk setup since the Callcentric outage, and now that I'm into it and testing things, I'm noticing some problems.

With allowguest=no in sip.conf, if I try calling my 1777 number from a softphone, I get an error like this in asterisk:

handle_request_invite: Sending fake auth rejection for device "Soft Phone" <sip:17778888888@callcentric.com>;tag=123456789

Then if I set allowguest=yes in sip.conf and try again, I get an error like this instead:

handle_request_invite: Call from '' (204.11.192.171:5060) to extension '17779999999' rejected because extension not found in context 'default'.

It's clear from the second example that the call is coming from a different IP address and Port than what I'm registered to.

My asterisk server is behind NAT with no port forwarding, so how does the call even come through from a different IP address and Port than what I'm registered to?

That's the first thing I'm wondering about.

I'm also pretty sure that's the cause of the error, too (incoming calls from servers that I'm not registered to).

It seems I can get it to work by adding a peer for each of the 20 Callcentric servers with entries in sip.conf like:

[callcentric1]
type=peer
host=alpha1.callcentric.com
context=callcentric-trunk

But that seems fragile since it will break if callcentric adds more servers or changes the host names.

What do other people do to get it to work?

Also, I should note, I'm only testing it by calling my 1777 number and iNum right now, since my Callcentric DID is still down. Maybe incoming calls to the real DID all come in from the server you're registered to?

Trimline
Premium Member
join:2004-10-24
Windermere, FL

Trimline

Premium Member

Re: [CallCentric] Callcentric and Asterisk Config Question / Rej

Make sure you have an inbound route for 17778888888 that points to a real extension. I think that's all you need.
advocate99
join:2011-03-08

advocate99 to david3

Member

to david3
You are correct that Callcentric sends inbound calls from a different IP/Port than the one you are registered to. Asterisk is perfectly capable of handling such inbound calls, as long as you don't block them. Allowguest=no blocks them. If you have FreePBX, turning off allow anonymous inbound SIP calls also blocks them, although CC's recommended set-up instructions turns on allow anonymous at the configuration file level.

If you're asking about how the calls penetrate your NAT, I'm assuming that it is because the SIP setup instructions come from the proper port/IP and instruct your system to contact the foreign IP/port, thus opening it up on your router.

So, to answer your question, we all handle CC by allowing guests.

Personally, I use this configuration:

»www.pbxinaflash.com/comm ··· n.14445/
david3
join:2000-03-21

1 edit

david3 to Trimline

Member

to Trimline
said by Trimline:

Make sure you have an inbound route for 17778888888 that points to a real extension. I think that's all you need.

I've got an inbound route for my Callcentric 1777 number pointing to my phones extensions, but it's under my callcentric context, and not the default context.

If I set allowguest=yes and add a Callcentric route under the default context, that seems to work. Thanks.
said by advocate99:

You are correct that Callcentric sends inbound calls from a different IP/Port than the one you are registered to. Asterisk is perfectly capable of handling such inbound calls, as long as you don't block them. Allowguest=no blocks them. If you have FreePBX, turning off allow anonymous inbound SIP calls also blocks them, although CC's recommended set-up instructions turns on allow anonymous at the configuration file level.

If you're asking about how the calls penetrate your NAT, I'm assuming that it is because the SIP setup instructions come from the proper port/IP and instruct your system to contact the foreign IP/port, thus opening it up on your router.

So, to answer your question, we all handle CC by allowing guests.

Personally, I use this configuration:

»www.pbxinaflash.com/comm ··· n.14445/

Thanks for the explanation. I read about your setup in the other thread you listed.

I've got a PSTN trunk going through an Obi110 so I can make and receive landline calls through asterisk.

I'm running asterisk 1.8.17 by itself without FreePBX. I've got srvlookup=yes and the dnsmgr is enabled. I simulated an internet outage by releasing the DHCP lease to my DSL modem. I noticed the asterisk CLI would freeze up for about a minute at a time, and then come back for awhile and freeze up again.

Despite that, even with the CLI frozen, I was able to make and receive PSTN calls through asterisk via the PSTN trunk.

Edit: Nevermind, I see what you mean now. I set the refreshinterval=60 for the dnsmgr so it would try to resolve while the internet was out, and then I had problems with the PSTN trunk.

I've got an old fashioned phone in the bedroom connected directly to the landline I can still use in an emergency, I guess.