SRST Config assistance

[feisalb]

Hello All

My customer has centralised cucm cluster with satellite sites.

All extension numbers are the full E.164 numbers e.g 441304863889.

ISDN passes the last 6 digits at some sites and 4 at others.

Incoming calls are translated from 6 or 4 digits to the full E.164 extension numbers.

Users are currently logged in with Extension Mobility.

Now what happens if I implement SRST.

The gateways are running H.323, see attached config.

The commands needed for SRST:

call-manager-fallback

max-conferences 8 gain -6

transfer-system full-blind

ip source-address 10.X.X.X port 2000

max-ephones X

max-dn X

dialplan-pattern 1 20335..... extension-length 5          <-----Not Sure what this does but was copied from a config

Do I need the dialplan-pattern command?

Will the phone in SRST mode have the full E.164 number dn as before?

If this is the case then I will need a translation pattern for inbound called numbers (4 or 6 digits) to full E.164 (extension number of phone)?

if I do at this translation how will this effect the config in non srst mode where the cucm does the translation.

How about outgoing calls, do I just configure " forward-digit" 6 or 4  (depending on site) on the outgoing pots dial-peer?

How do I do SRST for VG224 in sccp mode?

Do I add a voip dial-peer with destination 9.T pointing to the voice gateways ip address?

How will it work for ATA187 in sip mode?

I may have missed some things please fill in the gaps.

Many thanks

Feisal

[Osama Mahmoud]

First of all you can do manipulation a that are only related to srst under call-manager-fallback using translation-profile outgoing and translation-profile incoming , for vg224 I assume that you have a dial peer pointing to the cucm , what you need is to add another one pointing at the srst source ip with the same destination-pattern and with lower preference ( higher in value ) ,for ATA it will fall into srst normally as other phones .

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[feisalb]

VG224 and fxs ports on the 2921 gateways are sccp controlled so do not have dial-peers pointing to cucm.

Do you have any relevant configs that I can use?

Thanks

[Dennis Mink]

Feisal,

looking at your config, I see a 9.T pattern for your outbund calls to PSTN and two 23.. patterns pointing to two CUCMs(?) with different preference.  

i dont see any references in your config to extensions on phones that are E.164 compliant. Unless you translate  E164 extensions  (23..  > E.164) to the CUCM?    Also, inbound calls,  where are you translating them into 4 digits to match the 23.. pattern?

R.,

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[feisalb]

Yep, on cucm I have a translation pattern to change the incoming number to 353 19752XXX for this site. So I assume that the phone in srst will have the same dn and I will need to put in a translation on the gateway that works only when it is in srst mode somehow.

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[Osama Mahmoud]

The directory numbers on the phone will be inherited to srst , if you do manipulation on ccm not on the VGW to expand the numbers then you can add the same dial peer that points to the ccm but pointing to the source ip that u configures under call manager fallback . For the vg224 I think the solution of Adding a VoIP dial peer towards the primary VGW will work even if the ports are already controlled by sccp

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[Osama Mahmoud]

I forgot to add that in the dial peer pointing to the source ip you should add translations to expand the number

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[feisalb]

I read somewhere that I need to add station-Id to the vg224 ports for srst. It was in the sccp supplementary guide. Under the sccp group I just add the srst gw with a lower preference than the cucm.

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[feisalb]

If I have 6 digits from the isdn do I just add the dial-pattern command to expand it to the extension numbers on the phone?

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[Osama Mahmoud]

Yes , but you should do that on a dial-peer pointing to call manager fallback ip , so that yo don't affect normal mode

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