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Beginner

SPA500 dial plan - internal numbers via external gateway

Hi,

I've been trying to set up a dial plan to make remote working easier with the SPA500 series (SPA504G in this case).  The phone is connected directly to a SIP provider, no IP PBX in between, so this needs to work on the phone itself.

What I want, is to be able to dial an internal number at the office, which all begin "19" and have the phone dial the corporate voice gateway, log in and then connect to the internal number. Thus:

USER      dials 192212345

PHONE    dials "08000111111" then pauses 4 seconds, sends "#", pauses 2 seconds, sends "192212345".

The nearest I can were these plans, but none of them seems to connect to the voice gateway at all:

(*xx|<19x.:08000111111 ,#19x.>|1xx|999|1xxxxx|18xx|0|00|xxxxxxxx.)

(*xx|<19xxxxxxx:08000684827 ,#19xxxxxxx>|1xx|999|1xxxxx|18xx|0|00|xxxxxxxx.)     -- an attempt to eliminate the wildcard "x."; most internals are "19" + 7 digits.

(*xx|<19:08000111111 ,#19>|1xx|999|1xxxxx|18xx|0|00|xxxxxxxx.)

Has anyone succesfully done anything like this that they could share?  Cisco team have any wisdom about other ways to structure this?

Thanks!

Jeff

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Beginner

SPA500 dial plan - internal numbers via external gateway

Hi Jeff,

Im not sure if you can insert a pause when substituting a dialstring.

You have to test it (with something like wireshark to see what kind of INVITEs the phone is sending to the provider).

On the other hand, if your SIP provider can't handle  'internal numbers' maybe it's time to switch to another provider who can do this cause it's pretty basic functionality.

Regards

Erik Dekkers

Jener Network Solutions

Beginner

SPA500 dial plan - internal numbers via external gateway

Thanks Erik, that pretty much confirms my research, no sign of any way to tell it "after call connects, dial x".  If I have time to waste at some point, I might point wireshark at it.

Not sure what you mean re the SIP provider - this is SIP at my end and a big corporate phone system at the other via the PSTN.  There's no option to connect by SIP to the other end, so it has to be dialled via the PSTN gateway and good old fashioned DTMF.

Jeff