Help with inbound calls please

[geoffers1]

Hi. I'm trying to setup the inbound calling (I have read the manual and followed the section on it very carefully), however inbound calls are showing an engaged tone.

On the Asterix CLI for inbound calls I get the following message:

"[NOTICE 3509] chan_sip c: Call from '08454133***' to extension 's' rejected because extension not found"

Any help would be appreciated please.

[rrb555]

how did u setup your DID? please post all necessary details that u have done. this will help us answer your question

[geoffers1]

god i hate this system! i've deleted my DID now to go back to where I was, now I can't make outgoing calls. I knew I should never have started playing with this thing! If I do a "sip show peers" my SIP provider is listed as unreachable. I've not changed the carrier setup in anyway. Grrrrr!

[rrb555]

i guess u need someone who manages your servers other than yourself.

[williamconley]

if you are saying you deleted your DID (which was not being recognized anyway, that's what the "s" extension meant) ... how precisely did you do this? If all you did is choose delete from the bottom of the "Modify DID" page ... then the "unreachable" is not related to what you are doing.

advice: THREE voip providers minimum. Then if it turns out your problem is the carrier ... you can test that theory by switching. and if the carrier IS the problem, you can chuck 'em.

Alternative: You could add an "s" extension to the "[trunkinbound]" section in extensions.conf. Copy the _X. extension. If this works, the system is not recognizing the method your provider is using to send the DID (or they are not sending one! LOL).

SIP debug is very helpful in this situation.

And please post the VERSION of GoAutoDial (there is more than one, and it can make a difference).

[geoffers1]

Hi. If we leave the DID and inbound calling out of the equation for now (I could see from the Asterix CLI incoming calls but was getting the extension invalid error).

I have undone (as far as I can remember) everything I did to attempt to create the inbound calling; I deleted the InGroup and deleted the DID.

My SIP provider is still up and running (I can register a softphone directly with them), nothing has changed on my router either.

If I do "sip show peers" then I can see my SIP provider is listed as unreachable. Ive not touched any of the entries in the Carrier section of Admin.

If I do "sip set debug" and then look at the Asterix Real Time Logs, I'm not able to make anything out (not that I know what I'm looking at really).

AutoDial version is 2.4-309a

Everything was working before I started playing with inbound calling, and has been working fine for 3 months now.

Any help would be appreciated, our call centre is now non-operational and I'm getting grief.

Many thanks.

[geoffers1]

If I do "sip show registry" the output shows my SIP provider has a state of Registered but the realtime logs show my SIP provider is unreachable.

[geoffers1]

Okay, I've changed the carrier entry "qualify=yes" to "qualify=no". The Asterix realtime logs no longer show any info as to my SIP channel. If I do "SIP show peers" I'm told my SIP channel is "unmonitored" although "SIP show regsitry" shows my SIP channel as registered. And now I can make outbound calls again. Why is this the case?

[williamconley]

qualify = "check this sip connection for a round trip packet time and disqualify if the packet takes too long".

If you qualify=400 for instance, if the round trip packet takes over 400MS to return (4 tenths of a second), the sip account is disqualified. Marked "Unreachable" and asterisk will refuse to use it. If you set qualify to "yes" it defaults to 1000 with most stock systems. So a round trip packet taking over 1 second = disabled carrier.

Excellent IF you have a backup carrier. But not so good if you do not.

Remember: Asterisk was built to be a full-blown telephone PBX with auto-failover and everything. If there is a "down switch" between you and the person you are calling (or a poor quality one ...), the system should self-heal and route you to a healthier/functional route for a good quality call. A 1 second packet round trip is outrageous. It should ordinarily be between 20MS and 200MS (give or take half or double, LOL).