remote coverage path - busy here 1

[dmarenco]

Hello gents!,

I've been struggling to find out why CM keeps canceling the call with a busy here while sending the call to remote coverage path, I configured time of day routing so that during the day calls ring the main number in facility one and during the nights have them ring facility two at a remote destination. Here is what I did:

Mon – Sun 07:00 – 19:00 coverage path 318
Mon – Sun 19:00 – 07:00 coverage path 321 -- remote destination

CM sees the remote coverage 321 point 1 just fine:

[highlight #FCE94F]9:15:32 SIP>SIP/2.0 181 Call Is Being Forwarded >>>> Right here it starts forwarding to the remote destination[/highlight]
09:15:32 Call-ID: 51021679-3724755327-546024@SBC1-TOR.mydomain.c
09:15:32 om
09:15:32 no answer station [highlight #FCE94F]9058135700 cid 0x56 >>> 9058135700 is our main number [/highlight]
09:15:32 [highlight #FCE94F] coverage-path 321 point 1[/highlight] cid 0x56



LIST TRACE

time data
09:15:32 G711MU ssff ps:20
rgn:40 [10.146.7.182]:40672
rgn:40 [10.146.7.148]:2116
09:15:32 xoip options: fax:T38 modemff tty:US uid:0x50aa5
xoip ip: [10.146.7.148]:2116
[highlight #FCE94F]09:15:32 SIP>CANCEL sip:9058135700@oamp.sgns.net;user=phone SIP/2.0 >>>>[/highlight] it cancels the signaling to the main number because now the call found another destination, I believe this one message is ok
09:15:32 Call-ID: 805ee11be70e81a05b5a1da88e00
[highlight #FCE94F]09:15:32 SIP>SIP/2.0 486 Busy Here >>> right here I get a busy tone, it saw the coverage point but never sends the invite to it??
[/highlight]

I've checked trunk to trunk transfers, restrict call off net also

Thanks in advance for any suggestions

 

[kyle555]

need more than that. got a traceSM?

 

[dmarenco]

Sorry, I reedited the post for better understanding, I uploaded two files, traceSM and CM for the same failed call.

 

[dmarenco]

CM trace

 

[kyle555]

You have some CM configuration problem. There's no reason when terminating on a SIP station with 9058135700 that when CM picks trunk 6 member 217 that the call loops a few times:

12:44:07     Calling Number & Name NO-CPNumber NO-CPName                        
12:44:07     Proceed trunk-group 6 member 217    cid 0xbfa

I'll bet your CM alternate route timer is default at 6 seconds in your SIP sig group. Your SBC or carrier probably has something else to that effect when you're not answering nicely in good time.

Can you call from SIP or H323 stations through that main number and get out?
What's the traceSM looking like for those 4 invites bouncing between before failing to cover?

 

[dmarenco]

Kyle,

thanks for the input, I made a direct test call from the local facility, it went through just fine, I then figured that our CM was missing a route pattern in the all location table for this redirected calls, I added one route pattern for testing and calls are not getting busy signal anymore but still fail, I then took a direct call trace and had it compared to a redirected call and I am now noticing that the failed call is getting a 973 area code prepended which makes it end up in a 404 not found, but now I am getting a hard time figuring where it gets it from since both working and non-working are using the same route pattern and trunk group.

Non working:

15:53:08 route-pattern 431 preference 1 location ALL cid 0x192b
15:53:08 seize trunk-group 401 member 243 cid 0x192b
15:53:08 Calling Number & Name NO-CPNumber NO-CPName
15:53:08 G711MU ssff ps:20
rgn:40 [10.146.7.182]:41830
rgn:40 [10.146.7.149]:2052
15:53:08 xoip options: fax:T38 modemff tty:US uid:0x50a86
xoip ip: [10.146.7.149]:2052
15:53:08 SIP>INVITE sip:8[highlight #FCE94F]973[/highlight]9058212800@oamp.sgns.net SIP/2.0 this 973 shouldn't be here
15:53:08 Call-ID: 0caeeab1e1e81c6a55a1da88e00
15:53:08 Calling Number & Name NO-CPNumber NO-CPName
15:53:08 no answer station 9058135700 cid 0x192b
15:53:08 coverage-path 321 point 1 cid 0x192b


working:

15:29:37 route-pattern 431 preference 1 location 43 cid 0x176a
15:29:37 seize trunk-group 401 member 230 cid 0x176a
15:29:37 Calling Number & Name NO-CPNumber NO-CPName
15:29:37 SIP>INVITE sip:[highlight #FCE94F]89058212800[/highlight]@oamp.sgns.net SIP/2.0 if it goes like this it will work
15:29:37 Call-ID: 80b050621b1e81cfa05a1da88e00
15:29:37 Setup digits 89058212800
15:29:37 Calling Number & Name +9058135796 Millcreek Wir
15:29:37 Proceed trunk-group 401 member 230 cid 0x176a

 

[kyle555]

If the source is SIP and CM has no routing, it will use the proxy selection route pattern in the locations table.

973=NJ. Is your core CM there? Does some route pattern have a prefix mark and NPA 973?

Why even send the 8 to SM? I suppose you could have steering codes in there if you have to.

Or, if you're really lazy, just have the adaptation on your CM entity have a digit conversion adapter. min/max digits 14, beginning with 8973905, delete 4 digits, insert 8 and you'd get the same result as your working call.

 

[dmarenco]

yes our CM resides in NJ, I uploaded a snapshot of what the pattern looks like, the adaptation sounds like a good idea for the moment at least to get it functional

 

[dmarenco]

What location is used to take calls out for redirected calls?

 

[kyle555]

Where does that 8 normally come from? is it inserted by the route pattern or is it in the remote coverage point?

It's gotta be the ARS calltype, prefix mark, NPA stuff that's making it think adding a NJ NPA is a good idea.

And why the heck is the NPA 200??? That's not a legal or valid NPA. 973 is an overlay to 201, but there is no possible reason to have 200 in there.

I thought your dial plan might be a mess. That 200 in the NPA proves it! Who thought that was a good idea?

Just duct tape it in the session manager adaptation and pretend none of this ever happened!

 

[dmarenco]

Right, there is things that don't make much sense, but I followed your advise on the adaptation and it worked, ultimately I used a brand new route pattern with 905 as NPA to a different trunk and works too, thanks for your help, I'll go over the parameters and stuff that you mentioned for the sake of my own learning.

 

[kyle555]

Good luck! It's not easy to put it all together on your own with something built wrong!

Hadn't seen your question on where the location is picked for redirected calls.

If the redirecting party is registered, it'll use it's location. If it's an Xport station/something with no IP address, it'd be procr's I think.

Otherwise, as far as SIP goes, CM will look at the oldest via header in a SIP message to determine network region and location. Every proxy a message hops through get a via header added to the top. That way, the last via header is the first one and that of the originating IP (like a SIP phone). CM and SM compare that IP address against their IP maps/locations/etc to assess region.

That's how 1 SM on 1 CM serving 10 locations with SIP phones are able to use 10 locations of ARS.

*last thing - check the proxy sel rte for the NJ location and AAR for the NJ location. AAR and ARS only use the ALL table when a NR has no location defined or when it has a more exact match than the your location AAR or ARS table. So, if your SBC's IP happens to be in CMs network map to a region to a location in NJ, and that location has a route for AAR match 905, it would override the AAR location ALL match for 905. It's gotta be picking a route with a prefix mark and NPA somewhere.

And I'd advise you look into cleaning that prefix mark stuff up. Look into it - it's for systems to manage dialing 7 and 10 digit and when to insert the 1 or not - like breaking 905 out into every NXX and having the PBX know Scarborough to Oshawa needs a 1. Like with a prefix mark of 2 or 3, CM will check a toll table and see that Oshawa is not in the local 905 NXX list and insert a 1 when Mr Scarborough dials "905xxx" Oshawa, CM . If you're passing 10 digits + a steering code out to SM, I have no idea how prefix marks treat that because they're not intended for it.

Run a "list route" in GEDI or export to txt/csv. Youll find a 973 in there somewhere I'm sure.