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.) -- Regards Bhavesh GS LAB (O) +91 20 4671 1256 (M) +91 86000 64681
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.)
Ok But then using ESL; Do I not need same APIs as fs_clis? I may be wrong, but I tried ORIGINATE from ESL as well. So thought, similarly I should have some API to receive a call in ESL. If not, Is there some other way with ESL , I can make my own functions to create and answer calls? Regards Bhavesh GS LAB (O) +91 20 4671 1256 (M) +91 86000 64681 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.)
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Can I achieve this using DIALPLANs? If yes, any example/link giving directions on dialplans. Regards Bhavesh GS LAB (O) +91 20 4671 1256 (M) +91 86000 64681 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.)
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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 (M) +91 86000 64681
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.)
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@gslab.com <mailto:bhavesh.soni@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.) >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org <mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Correction Markus. Endpoints in fact dont use PSTN network for making calls. They have 10 digit number for ID, and using 10digitno@domain, sip url is formed and registered. So now two endpoints are on pure SIP enabled n/w with 10digitno@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@gslab.com <mailto:bhavesh.soni@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.) >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org <mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (3)
-
Bhavesh Soni
-
Markus Lindenberg
-
Normando Hall