How could I forward the SIP incoming call to one of the available extensions?
Hi, Right now I only have one extension in the SIP configuration as the destination. Could I put a list of extensions then FS forwards the call to the first available one? Thanks! Jian
Right now I only have one extension in the SIP configuration as the destination. Could I put a list of extensions then FS forwards the call to the first available one?
I'm not sure if i understand you exactly. By "extension" do you mean a SIP-endpoint like a phone or a gateway? Take a look at this simple example, is this good enough for your situation?: http://wiki.freeswitch.org/wiki/Extension_Status_Example If not, please let us now in more detail what you want to achieve(Is this a Call Center solution? Take a look at http://wiki.freeswitch.org/wiki/Mod_fifo) Best regards, Christian
Yes, my "extension" means the physical ATA connected to phones. I have 3 at home all registered to the FS server. I want all of them ring, or one by one for incoming calls from an external SIP connection. Then when two of them are being used to call outside I could still answer the incoming call. Thanks! Jian On Sat, Mar 5, 2011 at 2:19 AM, Christian Benke <benkokakao@gmail.com>wrote:
Right now I only have one extension in the SIP configuration as the destination. Could I put a list of extensions then FS forwards the call to the first available one?
I'm not sure if i understand you exactly. By "extension" do you mean a SIP-endpoint like a phone or a gateway? Take a look at this simple example, is this good enough for your situation?: http://wiki.freeswitch.org/wiki/Extension_Status_Example
If not, please let us now in more detail what you want to achieve(Is this a Call Center solution? Take a look at http://wiki.freeswitch.org/wiki/Mod_fifo)
Best regards, Christian
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On 5 March 2011 16:14, Jian Ren <renjian@gmail.com> wrote:
Yes, my "extension" means the physical ATA connected to phones. I have 3 at home all registered to the FS server. I want all of them ring, or one by one for incoming calls from an external SIP connection. Then when two of them are being used to call outside I could still answer the incoming call.
Take a look at http://wiki.freeswitch.org/wiki/Dialplan_XML and the link i've posted earlier(http://wiki.freeswitch.org/wiki/Extension_Status_Example) - this is easy to accomplish. If you are still stuck after reading the wiki, let us know. Best regards Christian
participants (2)
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Christian Benke
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Jian Ren