VoIP SIP partial number dialing

When using an old style analogue or ISDN telephone, the dialing of a number is not closed to the end. There is no signal the number is complete and finished. However, adapters and such enable old phones for VoIP using SIP.

As I understand, the SIP request headers contain the whole client address or number.

How then is a SIP session established without knowing if the dialed number is complet开发者_开发问答e?

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SIP (per se) doesn't say anything about when calls are made or dialing, that's entirely up to the device or program. Most ATAs act like traditional POTS phones connected to a switch, and dial off a completed dial-plan entry that matched (like 1-212-345-6789 or 911 or 411), or when a time since the last digit has elapsed (though most of those will end up forwarded to the "you've dialed an invalid number, please try again" message or beeps). True IP phones often function closer to a cellphone (or cordless phone) model, with a "call" or "dial" button.

In many devices the dial-plan is programmable, sometimes by the user, other times (more often) by the service provider (Vonage, etc), in a few by either party.

Depending on the dial-plan, it may do more or less validation of the number being dialed in the matching (like checking for valid area-code digits or not, etc).

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by guessing. If there comes no additional digit within a certain number of seconds, the call will be made. Often you can speed up this by terminating your number with a # or similiar.

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glglgl's guess is correct, a SIP device only initiates a call once it has acquired the full number it needs to use. SIP uses URIs in call requests which are very similar to email addresses and in the same manner that sending an email to a partial address is likely to fail initiating a call with a partial SIP URI is also likely to.

As to how SIP devices recognise when the user has completed the number it's normally done with a timeout, for example no more keys pressed within 10 seconds, or by the user pressing a "Send" key which as glglgl also alludes to will often be the # key on phones connected to an ATA. IP Phones typically have a "Send" or "Dial" button.

Some ATAs do allow you to adjust the timeout to detect when a user has completed dialling. I know the original Sipura ATAs (now owned by Cisco) allowed the delay to be configured in their internal dialplan.