[Freeswitch-users] Automation of calls?

Bhavesh Soni bhavesh.soni at gslab.com
Mon Jul 14 10:11:24 MSD 2014


Correction Markus.
Endpoints in fact dont use PSTN network for making calls. They have 10 
digit number for ID, and using 10digitno at domain, sip url is formed and 
registered.
So now two endpoints are on pure SIP enabled n/w with 10digitno at domain 
URLs. Now, is it possible to set auto-answer for ENDPOINT B.

Regards
Bhavesh
GS LAB
(O) +91 20 4671 1256
(M) +91 86000 64681

On 14-07-2014 11:24, Bhavesh Soni wrote:
> Hey Markus,
> thanks for inputs.
> I got your points.
>
> Let me give you some details. I have two boards running freeswitch; 
> Both have PSTN number at the back-end. (Presumably these are also SOft 
> SIP phone as per my understanding in FS).
> So, I want to automate a call between these two guys. Where I want to 
> write some scripts, that suppose originate from ENDPOINT A and call to 
> ENDPOINT B.
> And, then EP B auto-answers the call. (I can take up media path 
> manual; One can after picking up calls, speak manually holding Handsets.)
>
> Till now, somehow, using fs_cli originate I am able to DIAL B. But, B 
> still has to pick manually.
>
> Now, from your explanation what I think is, you are right in the sense 
> of making receiving automatic. I mean, I will have to intervene 
> somewhere on receiving endpoint and configure auto-answer somewhere; 
> AM I RIGHT? This is what you are trying to explain in the context of 
> receive path.
>
> Again thanks for inputs. They are helping.
> Regards
> Bhavesh
> GS LAB
> (O) +91 20 4671 1256
> (M) +91 86000 64681
> On 11-07-2014 18:12, Markus Lindenberg wrote:
>> Hi Bhavesh,
>>
>> what exactly are you trying to achive?
>>
>> If you originate a call, freeswitch creates two outgoing calls (it 
>> calls the "caller", as soon as the caller picks up, a call to the 
>> callee is created and the calls are bridged). So technically from the 
>> phone's perspective that's just two normal incoming phonecalls.
>>
>> Not to answer a call programatically, you'd have to control the 
>> phone. With some SIP phone you could make them answer the call 
>> automatically by setting a SIP header (e.g. for Snom: 
>> http://wiki.snom.com/Category:HowTo:Intercom). Or you could control 
>> the phone itself using a CTI mechanism supported by the phone, like 
>> uaCSTA.
>>
>> For analog lines that's not even possible I think, because there's no 
>> way to tell the phone that it should answer the call. What you can do 
>> though is look at mod_fifo and log those people in as "consumers", so 
>> their phones are always off-hook and connected to mod_fifo. I guess 
>> you can then write some scrips that connect a caller to a waiting 
>> consumer without having to touch the phone.
>>
>>
>>
>>
>> On Fri, Jul 11, 2014 at 10:52 AM, Bhavesh Soni 
>> <bhavesh.soni at gslab.com <mailto:bhavesh.soni at gslab.com>> wrote:
>>
>>     Can I achieve this using DIALPLANs?
>>     If yes, any example/link giving directions on dialplans.
>>
>>     Regards
>>     Bhavesh
>>     GS LAB
>>     (O) +91 20 4671 1256 <tel:%2B91%2020%204671%201256>
>>     (M) +91 86000 64681 <tel:%2B91%2086000%2064681>
>>
>>     On 11-07-2014 12:06, Normando Hall wrote:
>>     > Hi Bravesh.
>>     >
>>     > I sugest to use events socket to implement the full ESL API
>>     from you own
>>     > lnaguage. It is more flexible than scripts applications.
>>     >
>>     > This is my advice.
>>     >
>>     > Normando
>>     >
>>     >
>>     > El 11/07/2014 03:22 a.m., Bhavesh Soni escribió:
>>     >> Guys,
>>     >>
>>     >> Are there APIs available to automate calls in freeswitch.
>>     >>
>>     >> DIALING:
>>     >> TIll now using mod_lua, I have tried "originate"; And I am able to
>>     >> establish a call(again tweaking some logs, i found endpoint
>>     URI and used
>>     >> this with originate); In logs I can see multiple UUIDs
>>     generated, and I
>>     >> dont know which one is the correct for the session initiated
>>     with which
>>     >> I can answer from other end.
>>     >>
>>     >> Answering:
>>     >> On other side, to answer incoming call, I am not able to find
>>     or do
>>     >> anything as of now. I have tried uuid_answer; but with no
>>     luck.(I tried
>>     >> uuid_answer with all different UUID I saw at originate), But
>>     no luck.
>>     >>
>>     >> Apart from these, are there other approaches. I am using
>>     freeswitch
>>     >> version 1.2.1(I know quite an old; but I dont have much liberty to
>>     >> update this.)
>>     >>
>>     >
>>     >
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