filer incoming calls

[GoogleCodeExporter]

HI, I do receive a lot Anonymous calls (spam, fax broadcast) since using one 
account with Csipsimple. I do have FreeSwitch? in home to have those unwanted 
calls (anonymous,800-advertising,sales,etc) being processed to either 
VoiceMail? or droped. Like the following rules,

<condition field="caller_id_number" 
expression="^(0000000000|(\+?1)?(8(00|55|66|77|88)[2-9]\d{6}))$">
<action application="voicemail" data="default abc 2000"/> or drop the call.
Can we develop a basic filer like this dealing with Incoming calls so we can 
have it filtered and processed the way we want? thanks

Original issue reported on code.google.com by lipeng8...@gmail.com on 24 Sep 2011 at 2:18

[GoogleCodeExporter]

Yes it's an idea of enhancement. 
There is already a rule to auto-answer, why not a rule to auto decline. 

However, it will probably not be possible to tell to deflect to voicemail. It 
will reply with a "busy here" for example, and then it's up to the sip server 
to decide to deflect calls to voicemail when sip client on the other side is 
busy.

Original comment by r3gis...@gmail.com on 5 Mar 2012 at 4:12

• Changed state: Accepted
• Added labels: Priority-Low, Type-Enhancement
• Removed labels: Priority-Medium, Type-Defect

[GoogleCodeExporter]

Issue 1726 has been merged into this issue.

Original comment by r3gis...@gmail.com on 9 May 2012 at 12:21

[GoogleCodeExporter]

This issue was closed by revision r1465.

Original comment by r3gis...@gmail.com on 9 May 2012 at 12:24

• Changed state: Fixed

[GoogleCodeExporter]

Revision 1465 bring the enhancement. It's an extension of the auto answer 
feature.
To use
*Create a filtering/rewriting rule to autoanswer and in the additional 
"optional sip code" field enter the sip code (an integer) that you'd like the 
app to send.

For example, if you want the app to return busy status to remote, set the code 
to "486" 

List of available sip codes are available here :
http://en.wikipedia.org/wiki/List_of_SIP_response_codes#4xx.E2.80.94Client_Failu
re_Responses

You'll likely need to use the 486 code. If your sip server does support that it 
will redirect to voice mail. Or maybe they need another sip code (probably one 
4xx).

Original comment by r3gis...@gmail.com on 9 May 2012 at 12:29