Knowledgebase Article
| Knowledgebase > Unified Communications > SmartNode |
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| Why I can't call out to my SIP-provider ? | |||
| Article Id #: 46 | |||
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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