Asterisk is driving me nuts with multiple incoming trunks

[vpnwiz]

Hi, new here and new to Asterisk and Linux in general but getting the hang of it fairly quickly. I have a test box (1Ghz P3, 256MB RAM) and 3 SIP trunks that I have been installing/configuring/reinstalling with a few Cisco 79xx's, a Polycom 600 and some old Pingtel xpressas. Most recently I have been using the A@H 2.3 distro.

My goal is to have each of the 3 SIP incoming trunks ring to a different extension so I can take a different action for each trunk by forwarding that extension to a ring group. I should be able to do this by adding the "/<extension no>" to the end of the registration line. This was working last night with two of the trunks, but when I added the third one all three stopped working and rang to the default incoming rule.

Today I spent 12 hours on it changing contexts, extensions, ring groups, incoming routing settings and NOTHING worked, * just seems to ignore the /extension registration. I can see in my sip FW the extension after the / is registering. I've deleted all entries in my .conf files, cleaned out the MySQL tables but still something persists.

Is there an easy way to set my * install back to it's default state or do I have to wipe everything out and go through the whole iso image again? Help help please.... I can provide any info as needed.

Thanks.
Patrick

[grymlock]

If I am not mistaken, Asterisk@Home 2.3 contains Asterisk 1.2.2.....According to the Asterisk website, there is a major bug in version 1.2.2 that prevents asterisk from working. It is critical that you go up to 1.2.3 to resolve this issue. There is a release of Asterisk@Home 2.4.....This release contains Asterisk 1.2.3.....It is not on the main download page yet but is located on the files page.....

[MillsapsPE]

When you say 3 SIP trunks, are these 3 different accounts with different providers, or 3 DIDs, or ???
I'm no expert, but I don't think you assign where your SIP trunk goes in the registration line of the Trunks menu. If you have 3 DIDs, you can direct your calls by using the Inbound Routing menu. Otherwise, I guess you have to set up a new context for each extension?? Hopefully someone with more experience can chime in here. I'm curious, why would you direct calls to an extension, then to a ring group. Why not set up ring groups in AAH?

If you deleted all of your .conf files, then AAH can not operate. AAH only rewrites the .conf files ending with the _additional.conf. It won't rewrite the other .conf files once installed. It would seem to me that you will either have to recreate the .conf files for reinstall AAH. If you are going to reinstall, you might as install the newest version, 2.4.

[vpnwiz]

Thanks for the reply. I was bit by the no-audio bug in 1.2.2... that took two days to figure out. I installed from the AAH 2.3 ISO image but upgraded via shell script to 1.2.3 last week and then 1.2.4 this afternoon.

Of my 3 SIP trunks, 2 are with the same provider and one is a different one. I had been using "s" for the User Context for the inbound trunk name - I could see the s user registering with my SIP fw, so I thought the only way to get some differentiation between the incoming trunks was to send the registration to an extension ( I used 0001, 0002, 0003) then configure the extension to perma-forward to a ring group. None of this worked until I set up inbound routing and keyed off of the 0001, 0002, etc and configured the inbound routing to send to the ring group.

The worst part is I had this set up and working on two trunks, backed it up, added the third trunk and nothing worked. When I set everything back the way it was it STILL didn't work... even after spending all day at it. Man that's enough to send someone over the edge

Next time I install AAH I will back up the default config files so I can "reset" my server as needed. I think the only other thing I would need to do is empty out the MySQL tables.

Thanks,
Patrick