To match the local number with the external number during a SRST failover, you can use the command dialplan-pattern <<tag_pattern>> extension-length <<length>>. this command also ensures that a full E.164 address is sent as the ANI for outgoing calls. The dialplan-pattern command causes the SRST router to create an additional virtual dial peer for local extension corresponding to the pattern specified in the command.So, this command can be used to map the DNIS sent by the PSTN to a local extension.
Pay attention that if you play with translation rules on your voice-port, it may be a good choice also to continue with digit manipulation to conform your SRST dialplan as the translation rule are processed before.
As with Call Transfer , Call Forwarding is enabled by default between IP Phones which are registered with the SRST process.Again if you need to enable H450v3 call-forward for non-local numbers , you have to enhance your config with the following comand:
Example of this will be :
call-forward pattern .T
SRST routers support also the call transfer between IP Phones by default when they are registered onto it. But to really have a successful transfer call , you need also to precise the pattern transfer and the way on how it will transfer your call. So don’t forget to implement these commands:
What are representing the parameters ?
- <<pattern>> is the pattern that you authorize to dial under SRST mode ( example …. or .T)
- <<methods>> is the call transfer method operation.
Let’s see what are the operation mode possible:
- Blind : Cisco proprietary method where calls are directly transferred without any consultation.
- Full-blind : H450v2 standard for call transfer where calls are directly transferred without any consultation.
- Full-consult : H450v2 standard for call transfer where you need to perform a consultation before to initiate the real transfer. This method requires two line to make the transfer and if the second line is not available then the full-blind is used instead.
- Local-consult : H450v2 standard for call tranfert but this time, it is only valid for local transfers.So again if you don’t have the second line or you are not transferring to a local party , the blind method is used instead.
Full-blind and full-consult are the Cisco preferred method.
Don’t forget also to configure your line as dual-line to allow not only the call transfer but also the call waiting , conferencing and so on
If you need to integrate your SRST router with analog/T1 CAS link to Unity, then you nedd to adapt the configuration as follows:
pattern direct * CGN
pattern ext-to-ext busy #FDN#2
pattern ext-to-ext no-answer #FDN#2
pattern trunk-to-ext busy #FDN#2
pattern trunk-to-ext no-answer #FDN#2
So you must use the configuration under the vm-integration command to enable voicemail integration with DTMF and analog mail systems. Here are a short explanation of all settings:
- Pattern direct represents the fact that you press the message button
- Pattern ext-to-ext is the forwarding to voicemail from an extension to another extension
- Pattern trunk-to-ext represents the forwarding to voicemail from an external trunk to an extension
When configuring the pattern direct, pattern ext-to-ext and pattern trunk-to-ext, you must also specify combinations of alphanumeric strings (fewer than 4 digits in length) as well the calling number (CGN), called number (CDN) or forwarding number (FDN) to be sent to the voicemail system.
Then imagine the following scenario : Phone 1 ( DN 1001) is calling Phone 2 (DN 1002) which is set up to forward all busy and no-answer calls to the VM
We have then 1001 is the CGN, 1002 is the FDN and the voicemail system is the CDN. In other words, the calling number (CGN) is the number of the call originator, the frowarding number is the number of the extenson which is forwarding the call to the voicemail.So let’s take back our previous example and see what the configuration looks like:
pattern ext-to-ext busy #FDN#2
will represent to dial the number 123456789 (VM system) and send to the VM the following string in DTMF #1002#2 which must route the call correctly to the right user mailbox.
Under SRST, you can still benefit of the voicemail system integration at the central site but this time instead to have an internal call to Unity, you need to make an outside calls to Unity when SRST is enbale.
So under the call-manager-fallback command, enable the voicemail <<VM_number_to_dial>> and your end-users will still have the opportunity to press the message button in SRST
If we want to have SRST on MGCP gateways, ensure that you have the two following commands:
This will ensure that the MGCP gateway can provide Call Processing with SRST. Don’t forget also to have an H323 config to take over in SRST as MGCP hasn’t any Call Control.