add ARI SIP transfer handling documentation #126
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jcolp
requested changes
Feb 10, 2025
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| Transfer is a telephony feature. There are two types of transfer, attended and blind. These names may differ depending on the system you are looking on, or what background you have. | ||
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| The ARI transfer handling is supposed to be used in an ARI only environment to control a traditional channel or bridge with ARI is not possible since it needs to be in a Stasis context. Please refer to "how to use ARI" for more information. For this scenario all channels and bridges need to be placed into an ARI context. |
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I think the addition of "is not possible since it needs to be in a Stasis context." is confusing. What are you trying to convey there?
| Attended transfer happens, if a call on hold and an active call are connected on request of the user. The user can talk to both parties before connecting them. This is also called a consultative transfer. | ||
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| ### Blind transfer | ||
| Blind transfer happens, if the user transfers the active call to a destination, not knowing the state or reachability of the destination. The user does not talk to the destination before the transfer.Blind transfer is also called a cold transfer. |
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| #### Attended transfer | ||
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| Asterisk will try to get as much information as possible about the transfer. The `REFER` does only contain the corresponding SIP-Call ID. With this SIP-Call ID, asterisk try to resolve the channel-ID used by ARI for the channel. |
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| Where the channel marked with an "X" is set on hold, channel A should receive MOH but this is not necessary. | ||
| Depending on the used phone a SIP REFER is send to the channel B or D. Either way, Asterisk tries to resolve all the channel information involved. | ||
| the `channelTransferEvent` contains both `refere-to` and `refered-by` information. |
| ### Actions after the event is received | ||
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| #### Attended transfer | ||
| The ARI application MUST take some action in order to handle the REFER. The Asterisk will not do anything by itself. The first thing to do when received an SIP REFER is acknowledge it. This is done by the Asterisk, as well as the first `NOTIFY` telling the phone an `100 Trying` SIP frag. After that, the telephon waits for a `Notify` containing a SIP frag with `200 OK`. This has to be sent by the ARI application manually. |
| * channel_unavailable | ||
| * channel_declined | ||
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| these will result in the corresponding SIP frag status codes being transmit via SIP notify to the phone. `channel_progress` will do nothing but tell the phone to hold on. This is done by asterisk itself but can be repeated. The other responses will terminate the REFER process in the one or the other way. Using `answered` the phone will most likely hangup the active channel and will wait for the hangup of the other one. |
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