Call Routing Table Not Working

[cgashlin01]

I have gone to System>Trunk Related Information>Call Routing Tables and set my 2nd(2) routing table up to do 1 to 1 match on extensions for a SIP system connected to the Mitel.

However, when i dial from my sip phone to my desk extension the receptionist picks up.

We are running PRI for the MITEL and the secondary system I am setting up is running SIP is that matters.

Perhaps the routing group is not set correctly.

I have my extension 1141. When I dial 1141 in my SIP phone i see in Asterisk it taking n 1141 and dialing the Mitel system. However it is treating it like a CO line it seems.

Once this is completed. I will do a complete write up on how to integrate the two system for anyone looking to do the same in the future.

[Tech Electronics]

cgashlin01,

You may want to just change the Ring-In Type of the first entry to Collected Digits; leave the Pattern and Ring-In Destination fields blank though.

Thanks,

TE

[cgashlin01]

I have done that.

It still is being overridden it seems.

Is there anything I can check to see why when I dial an extension and it comes in it is going the wrong route?

Thanks

[dwayneg]

Okay, here's a whole different way to approach this.
If, in fact, you can currently call one of your 5XX extensions by selecting the trunk group connecting to Asterix and dialing an extension you can actually make the Mitel phones call the Asterix phones by just dialing the extension.
As Tech said, you must renumber the CONFERENCE feature to something that doesn't start with 5.
Next create a PHANTOM EXTENSION 5.  Manually forward that phantom (using 359 etc) to the Asterix trunk group 92005.  When it asks for a destination number enter 5.
Now here's what happens when you dial an extension starting with 5: Call goes to phantom.  Phantom, invoking its forward rule, grabs Asterix trunk, then dials 5 into the trunk (first digit of Asterix extension) and then caller continues dialing other digits.  System is going to buffer the extra digits, so you don't need to worry that you're dialing too fast, caller just dials the entire extension.  This will not work, however, from analog phones, maybe not SIP phones

[Tech Electronics]

cgashlin01,

Are you using a new Call Routing table or an existing one for the PRI? Let's try doing this another way, instead of going to a Call Routing Table let's just use the Collected Digits right at the SIP Peer Trunk Group.

System > Devices and Feature Codes > SIP Peers > SIP Peer Trunk Groups > {SIP Trunk Group} > Trunk Group Configuration: Set Day and Night Ring-In Type to Collected Digits.

Thanks,

TE

[cgashlin01]

Tried it, Seems that its just ignoring stuff now. There has to be something over riding everything. At least what it sounds like.

[cgashlin01]

For dwayn, This is not the same as the other post I have. The other post worked with Tech answer. I also figured another way of doing it that seems to work well also.

This issue is that when a phone on the asterisk dials an extension that is on the Mitel it does not actually go to the extension instead it goes to the receptionist.

I have checked all my routing tables I am just trying to see if there is something over riding or taking over instead of routing tables.

[Tech Electronics]

cgashlin01,

The problem at this point is that the Asterisk is sending us something we are not expecting to see when we are collecting digits and since we don't know what that pattern is we send the call to the Primary Attendant; which is most likely your Receptionist.

So what we need to see is what the Asterisk is sending us in terms of digits and then modify them as needed. Unfortunately, we are not able to absorb digits directly from the Trunk Group like we can with a PRI so we will need to modify our call routing table a little to account for it.

If you have Wireshark setup already and are monitoring the SIP Traffic this will be easy; otherwise we will need to turn on the Full SIP Debug in the 5000 and check through our logs.

How do I turn on SIP Debug messages?
1. Enable OLM [Online Monitor]: This is up top under View

2. Then go to System > Maintenance > Message Print: Set the Expanded Message Print Flag to Yes.

3. Enable the SIP debug messages
    a. Go to System > Devices and Feature Codes > SIP Peers > SIP Trunk Groups > 9200X > Configuration: Set the SIP Message Output Format to "FULL"
    b. Go to System > Devices and Feature Codes > SIP Peers > General Configuration: Set SIP Message Output Format to "FULL" and set SIP Log Level to "DEBUG"

Nest after making a call we need to download the Error Logs and see what is going on.

Thanks,

TE

[cgashlin01]

You are all going to laugh.....

I figured it out....

Yes setting the Value to Collected Digits in the Value for the Trunk Group Configuration was 1 problem. The other was my dialing rule in my asterisk system. It was not passing correctly.

Does anyone here believe it would be beneficial to type up a massive how to with pictures and upload tot he forums?

I thank you Tech you really helped me on everything!

[Tech Electronics]

cgashlin01,

No one is laughing, at least not that I hear, about someone missing something. The first person that does is the one who most likely hasn't been working on the system.

As for the document that is the whole point of these forums in the first place. This way no one has to recreate the wheel like you did; unfortunately.

If you need help with the document I would be willing to lend a hand with that as well. Unfortunately I do not work on Asterisks systems nor do I have access to one so you will have to do all of those screen shots and writing.

Thanks,

TE

[cgashlin01]

Well what I have learned from this experience is that the Mitel system can handle things but it does seem to not be as stable as I would have though.

For instance. My asterisk system ALWAYS talks to the Mitel system without issue. If I run the reporting and logging I notice that sometimes the Mitel does not connect up to my asterisk box and instead just tells me destination don't exist lol.

I know this type of integration would be very useful for people not wanting to spend 500 bucks a phone for their branch offices.

I will most likely start the documentation tonight or tomorrow.

P.S. I used to be a project manager for a large company so I will write documentation that is easily followed for anyone. (Love them table of contents)

[Tech Electronics]

cgashlin01,

I am not sure what you mean by not being as stable, but I guess that since you don't know the system as well it would appear that way.

As far as saving money though I agree whole-heartedly on that topic as the equipment can be expensive, but then again you are paying for a known variable of it working versus trying to make something work when you aren't fully aware of how they would integrate.

I look forward to seeing your documentation.

Thanks,

TE

[cgashlin01]

I know asterisk really well :-) Been doing it forever.