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- Polycom dialplan digitmap
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09-14-2012 03:06 AM
Hi,
I'm actually getting a problem. We have this dialplan :
|*xx|#x.T|R+R000Rx.T|[1-9]xxxxxx|R0R00084Rx.T|RR000R[1-9]xxxx.T|x.T|
When we want to call an IDD , the number must be transformed from 000XXXXXXXXXXX to 0008400XXXXXXXXXXX .
But there is a problem when we put # or * at the end of the number, it's call directly 000XXXXXXXXXXX .
Where is the mistake on my dialplan ?
Regards
Solved! Go to Solution.
Accepted Solutions
09-17-2012 04:33 AM
Hi,
This dialplan does not modify the number when adding # on off-hook.
But adding R0R00084Rx.T# to the dialplan seems working : The number is modified when adding # at the end.
Thanks for your assistance.
09-14-2012 03:23 AM
Hello Knuckles,
have a play with the dialplan.removeEndOfDial Parameter.
Value usually is 1
If set to 1, strip trailing # digit from digits sent out.
Best regards
Steffen Baier
Notice: I am an HP Poly employee but all replies within the community are done as a volunteer outside of my day role. This community forum is not an official HP Poly support resource, thus responses from HP Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge.
If you need immediate and/or official assistance for former Poly\Plantronics\Polycom please open a service ticket through your support channels
For HP products please check HP Support.
Please also ensure you always check the General VoIP , Video Endpoint , UC Platform (Microsoft) , PSTN
09-14-2012 09:26 AM
Hello Knuckels,
I actually did test your Digitmap earlier:
dialplan.digitmap="|*xx|#x.T|R+R000Rx.T|[1-9]xxxxxx|R0R00084Rx.T|RR000R[1-9]xxxx.T|x.T|"
Using the above with an SPIP335 running UCS 3.3.4 and using an Asterisk 1.8 SIP server did always result in the number being modified.
I dialled 00012345678901 as an example to have the same lenght of digits and adding a # at the end did not affect to change the dialled number into 000840012345678901.
Using a * at the end did not modifie the number and dialled the original number instead.
I was not using any modifications apart from a normal SIP registration.
We always advise users to post their current software version so check's against release notes can be done to verify someone's findings.
Are you dialing off hook or on hook?
Best regards
Notice: I am an HP Poly employee but all replies within the community are done as a volunteer outside of my day role. This community forum is not an official HP Poly support resource, thus responses from HP Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge.
If you need immediate and/or official assistance for former Poly\Plantronics\Polycom please open a service ticket through your support channels
For HP products please check HP Support.
Please also ensure you always check the General VoIP , Video Endpoint , UC Platform (Microsoft) , PSTN
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