How to test/max out PSTN circuit capacity? (PRI, T1 CAS, etc)
So, I do a couple Cisco VOIP installs a year. Some CME, more CUCM lately. I'm wondering what would be a good way for (1) person to test the PSTN circuit capacity for digital trunk type circuits like a PRI or T1 CAS. My concern isn't determining if the circuit is adequate for the user base; I'm aware of the general sizing guidelines, and haven't had too much issue with over or under subscribed PSTN circuits. I'm more interested in testing the capacity for troubleshooting purposes. As we all know in the VOIP world, the Telco's love to dodge problems and issues and blame it on the VOIP system, leaving you to PROVE it's an issue on their end.
I'm not aware of any command on IOS gateways that would let me max out a PSTN trunk and consume all the channels at once. I think for it to really be effective, you would also need a handful of DID's you could attempt to pass a call through since being able to just consume a channel on a circuit doesn't really verify it's working, so maybe this isn't as feasible as I'm thinking?
Still, I'd like to see if anyone has any methods or devices or ways they use to max out a trunked PSTN connection for testing purposes. If there's a piece of software or hardware that will do it, I'd love to hear about it, regardless of cost. I might be able to get my employer to pay for it.
It would be even more awesome if there was something that could use up every channel on the circuit, then hang up, and report to me if the channels open up, or if they're seized, or whatever. But if there were only a way to just max out the circuit and report what channels will allow a call through, that would be fine too.
Any ideas or thoughts are appreciated; I know this may just be a pipe-dream of mine..
Oh, and I should clarify, I'm not looking for a way to check out SIP trunk capacity; they're very rarely used in my corner of the globe due to poor Internet connectivity.
Cisco used to have a tool that would simulate calls, it worked with CUCM 3.x and it had a nice GUI (oh that was a long long time ago I know). Later they dropped that tool and told us to go and use a different tool which was scriptable with no GUI. Last year at the CLEUR they were handing out yet another tool - I did not test it yet but I can browse through my emails and hopefully find the email address of that guy who could give you the last version if you want to.
Anyway, I would do this: create a bunch of SIP phones. Make them register using linphone or Ekiga (anything that does not suck). Use JTAPI to initiate these phones to make phone calls over the trunk, wait for the result, and I think I would probably take a look at the media stream as well (just ot make sure the call is really connected). This might sound like an interesting weekend project, does it. :-)
IntroductionCUCM Routing RulesDial String implementation PolicyCUCM Routing LogicSIP URI Call Routing Analysis+++ Case Study: 1 ++++++ Case Study: 2 +++Conclusion
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