Why I can't call out to my SIP-provider ?
Posted by Elena Brambilla, Last modified by Daniel Lizaola on 12 December 2017 04:43 PM
In order to send a call over the SIP protocol you need to have the whole number complete as SIP does not support the so called "overlap-dialing" (dial digit by digit).
ISDN telephones can send the number already "en-bloc" (if you dial first an lift the handset afterwards) or send it digit-by-digit. Analog telephones always send the number digit-by-digit. In the SmartNode configuration you will need to configure now a routing-table to collect every digit and, if there is no more digit after a timeout, route the call out to SIP.

Configuration example:

context cs
routing-table called-e164 RT_DIG_COL
route .T dest-interface IF_SIP

NOTE: IF_SIP is the name of your SIP-interface and can differ in your configuration. Please use here the name as configured in your configuration.

You need to route now the calls from the fxs/isdn interface to this routing-table in order to collect the digits like this:
route call dest-table RT_DIG_COL
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