Difference between revisions of "SIP phone: Forward unattended call"
From Zenitel Wiki
(14 intermediate revisions by 2 users not shown) | |||
Line 1: | Line 1: | ||
+ | {{AI}} | ||
The Event Type [[Private Ringing Outgoing(Event Type)|33 - Private Ringing Outgoing]] can be used to forward unattended SIP calls. In order to work, it requires that the SIP device returns the SIP status "180 Ringing" when the phones starts to ring, and "200 OK" when connecting. | The Event Type [[Private Ringing Outgoing(Event Type)|33 - Private Ringing Outgoing]] can be used to forward unattended SIP calls. In order to work, it requires that the SIP device returns the SIP status "180 Ringing" when the phones starts to ring, and "200 OK" when connecting. | ||
Line 5: | Line 6: | ||
* [[Configuration guide for X-Lite|X-Lite]] | * [[Configuration guide for X-Lite|X-Lite]] | ||
* [[Configuration guide for Grandstream GXP2000|Grandstream GXP2000]] | * [[Configuration guide for Grandstream GXP2000|Grandstream GXP2000]] | ||
− | * [[ | + | * [[Cisco Call Manager 6 configuration guide|Cisco Call manager 6 with Cisco IP Phone 7960]] |
* Alcatel OmniPCX (OXO) via SIP trunking | * Alcatel OmniPCX (OXO) via SIP trunking | ||
* [[Configuration guide for Ascom IP-DECT|Ascom IP Dect System]] | * [[Configuration guide for Ascom IP-DECT|Ascom IP Dect System]] | ||
Line 12: | Line 13: | ||
=== Configuration examples === | === Configuration examples === | ||
− | + | ||
− | + | ====AMC10.55 and later==== | |
:*The SIP phone is 9555. | :*The SIP phone is 9555. | ||
− | :*When an intercom station calls 9555, the phone will start to ring, the ringing time is controlled by the "[[Exchange_%26_System_%28AlphaPro%29#Timers|Private Ringing Time" timer]] (30 sec default). If not answered, the call will time out, and the [[Private Ringing Outgoing(Event Type)|event 33]] is triggered. This event sets up a new call to directory number 9547 (which can be an intercom or another SIP phone). | + | :*When an intercom station calls 9555, the phone will start to ring, the ringing time is controlled by the "[[Exchange_%26_System_%28AlphaPro%29#Timers|Private Ringing Time" timer]] (30 sec default). If not answered, the call will time out, and the [[Private Ringing Outgoing(Event Type)|event 33]] is triggered. This event sets up a new call to directory number 9547 (which can be an intercom or another SIP phone). |
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
:*[[C KEY|$C]] is needed to "speed up" the disconnection, so that the [[DIAL DIGITS|$DD]] command is not lost during the disconnection tone | :*[[C KEY|$C]] is needed to "speed up" the disconnection, so that the [[DIAL DIGITS|$DD]] command is not lost during the disconnection tone | ||
In [[AlphaPro]], go to [[Exchange_%26_System_%28AlphaPro%29#Events|Exchange and System -> Events]], press Insert and create the following event: | In [[AlphaPro]], go to [[Exchange_%26_System_%28AlphaPro%29#Events|Exchange and System -> Events]], press Insert and create the following event: | ||
− | + | [[Image:SIP Phone Forwarding.png|left|thumb|500px|Event programming for SIP phone forwarding]] | |
− | + | <br style="clear:both;" /> | |
− | |||
− | | | ||
− | | | ||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | |||
− | + | {{Code2| | |
− | + | [[C KEY|$C]] L%1.dir | |
+ | [[DIAL DIGITS|$DD]] L%1.dir L9547 | ||
+ | }} | ||
− | [[Category:Applications]] | + | [[Category: ICX-AlphaCom - SIP Integration]] |
+ | [[Category: AlphaCom - SIP Integration]] | ||
+ | [[Category: AlphaCom Applications]] | ||
+ | [[Category: ICX-AlphaCom Applications]] | ||
+ | [[Category: Applications using Event Handler]] |
Latest revision as of 11:21, 24 February 2023
The Event Type 33 - Private Ringing Outgoing can be used to forward unattended SIP calls. In order to work, it requires that the SIP device returns the SIP status "180 Ringing" when the phones starts to ring, and "200 OK" when connecting.
The forwarding function has been verified with:
- IP Dect System 6000
- X-Lite
- Grandstream GXP2000
- Cisco Call manager 6 with Cisco IP Phone 7960
- Alcatel OmniPCX (OXO) via SIP trunking
- Ascom IP Dect System
Note - The event type 33 require AMC 10.30 or later.
Configuration examples
AMC10.55 and later
- The SIP phone is 9555.
- When an intercom station calls 9555, the phone will start to ring, the ringing time is controlled by the "Private Ringing Time" timer (30 sec default). If not answered, the call will time out, and the event 33 is triggered. This event sets up a new call to directory number 9547 (which can be an intercom or another SIP phone).
- $C is needed to "speed up" the disconnection, so that the $DD command is not lost during the disconnection tone
In AlphaPro, go to Exchange and System -> Events, press Insert and create the following event:
Action commands: