[cisco-voip] Call Routing Problems

Madziarczyk, Jonathan JMad at cityofevanston.org
Tue Jan 5 11:19:14 EST 2010


If you want to look at your PRIs and their usage you can use RTMT to
monitor their activity.  That should give you a better idea if they are
filling up or not.

 

As for where your getting your busy signal from.  Start at the first
point your call comes in from the PSTN.  Like Mike said, debug isdn q931
(and terminal monitor) to see if the call even enters your site.  From
there start working your way down the line.

 

Good luck!

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Huffman, Tim
Sent: Tuesday, January 05, 2010 7:26 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Call Routing Problems

 

So I've checked QOS and all other settings. I enabled more bandwidth in
the Locations settings in CM (which fixed my not enough bandwidth
problem, THANKS!).  After about 65 calls went through I started getting
busy signals.  We have 4 T1s at the remote site, that should give us 92
lines.  I've checked all route groups/route lists in CM and everything
looks to be setup correctly (All LD T1's enabled and ordered correctly).
I believe this could be a T1 issue, as we've never saturated the lines
like that in our remote office.  Is there a way to test all 23 ports per
T1 to confirm everything's working the way it should?  Any other ideas
why I'm getting fast busies after approx 65 calls?

 

Thanks,

 

Tim Huffman

 

From: Madziarczyk, Jonathan [mailto:JMad at cityofevanston.org] 
Sent: Monday, January 04, 2010 2:43 PM
To: Huffman, Tim; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Call Routing Problems

 

Hey Tim,

 

Like Nate was saying, be sure to check your network link between the
sites to make sure if there is QOS on the routers it will match whatever
you're upping your location bandwidth to in CUCM.  Otherwise call
quality will be fine up to 432 kbps, but on a fully saturated link that
26th call will hurt everyone's call quality.  Also if you have redundant
wan links, you'll want to check both links and not just the one.

 

Jon

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Huffman, Tim
Sent: Monday, January 04, 2010 1:23 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Call Routing Problems

 

Thanks for all of your responses.  Location was set to 432 kbps.  This
was most likely the issue as we had 25+ calls going over to the remote
site.

 

Thanks,

 

Tim Huffman

 

From: Matt Slaga (US) [mailto:Matt.Slaga at us.didata.com] 
Sent: Monday, January 04, 2010 1:57 PM
To: Huffman, Tim; cisco-voip at puck.nether.net
Subject: RE: Call Routing Problems

 

'Not enough bandwidth' only is sent to a device when the 'location'
bandwidth is overrun.  Check your location bandwidth settings (System ->
Locations).

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Huffman, Tim
Sent: Monday, January 04, 2010 1:47 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Call Routing Problems

 

All,

 

We have a Subscriber and Publisher at our main location.  We have
another subscriber at one of our remote office that is associated with
the publisher at the main location.

 

I made a route pattern on our Publisher that would allow for some of the
calls that are incoming at our main site's T1s, to go outbound our
remote sites' T1s so we can free up some channels on our T1s at the main
site.  Once I made this change, it worked with no issues until we hit a
certain call volume (not sure how many).  Once it went haywire phones at
our remote location started saying "not enough bandwidth."  It also
caused issues trying to call inter-office from our main location to our
remote location.  How would I tell what the limitation is of
simultaneous calls from our main location to our remote location?

 

I did already confirm that we did not saturate our WAN connection
between the two sites.

 

Thanks,

 

Tim Huffman

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee,
you are hereby notified that you have received this communication in
error and that any use or reproduction of this email or its contents is
strictly prohibited and may be unlawful. If you have received this
communication in error, please notify us immediately by replying to this
message and deleting it from your computer. Thank you. 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20100105/d17cd692/attachment.html>


More information about the cisco-voip mailing list