Re: [Freeswitch-users] Force SIP UA to pick up call during ringing?
clippyLooks like you are trying to build a call center/clippy
have you seen mod_fifo?
It's designed to let people on headsets sit idle and you can send calls to
them at will.
On Tue, Jun 16, 2009 at 3:11 PM, Peter P GMX prometheus...@gmx.net wrote:
Thanks Michael,
I have disabled it now.
I finally got it to work, (sip_h_Call-Info=sip:$${domain};answer-after=0)
but the behaviour was not as desired, as I didn't manage the phone to
pick up the call on the headset. It will only have the speaker enabled.
So I will have to go a different way with parking the call and then
forward it.
Best regards
Peter
Michael Jerris schrieb:
uuid_setvar unique_id sip_invite_params intercom=true should be
unnecessary.
Mike
On Jun 16, 2009, at 1:49 PM, Peter P GMX wrote:
It mainly works now by uuid_transfer the following way via event
socket.
uuid_setvar unique_id sip_invite_params intercom=true
uuid_setvar unique_id sip_auto_answer true
uuid_transfer unique_id 1000 XML default
so the call is transferred from 1000 to 1000.
What happens:
1) If I disable intercom on the Snom phone, the phone rings, stops
ringing and rings again (ok)
1) If I enable intercom on the Snom phone, the phone rings, stops
ringing and hangs up (not ok)
So I do not get the Snom to pick up the call in intercom mode.
The last invite is:
INVITE sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib
SIP/2.0
Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF
Route: sip:1...@217.24.11.189:2752;transport=tls;line=er6kxnib
Max-Forwards: 68
From: Peter FS sip:723...@217.xx.xx.xxx;tag=9eQ8rjQy533HF
To:
sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true
Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e
CSeq: 116467629 INVITE
Contact: sip:mod_so...@217.xx.xx.xxx:5061;transport=tls
Call-Info: sip:217.xx.xx.xxx;answer-after=0
The intercom part is there and the Call-Info line with answer-after
also.
The phone answers with
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF
From: Peter FS sip:723...@217.xx.xx.xxx;tag=9eQ8rjQy533HF
To:
sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true
;tag=71rskygkr2
Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e
CSeq: 116467629 INVITE
Contact:
sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;reg-id=1
WWW-Authenticate: Digest realm=sip2.mycompany.de,
nonce=2ee26efe6ab27f88, algorithm=MD5
Content-Length: 0
and hangs up.
Anybody know how to solve this Snom intercom issue?
Best regards
Peter
Michael Jerris schrieb:
The transfer should work but it sounds like offhook agents is what
your really trying to accomplish so I would go with brian's
suggestion.
On Jun 16, 2009, at 7:38 AM, Peter P GMX prometheus...@gmx.net
wrote:
Hello Michael,
I want the phone be ringing, just for acoustical feedback reasons.
But what if I
* transfer it to the same user destination again (now with
intercom
enabled), will this work?
* transfer it to park and then transfer it to the same destination
again (now with intercom enabled)
Best regards
Peter
Michael Jerris schrieb:
The only way I can think to do this today would be to cancel the
call
and re send with the intercom headers for a phone that supports it.
It may be possible to send a reinvite with autoanswer headers but I
doubt that would work, all you could do is try making code to do it
it
a sipp or sipsak scenario and test it. A better aproach might be
to
answer the call normally and detect that to start your web workflow
or
not really ring the phone, just the web app and deliver the call
with
autoanswer when the button is hit in the web ui.
Mike
On Jun 16, 2009, at 4:24 AM, Peter P GMX prometheus...@gmx.net
wrote:
Hello Brian,
this is too easy :-).
This is for a small callcenter app and I only want the user to
pickup
the call once (to accept the call in X-Lite (or a Snom phone)
and to
start the workflow on the web application). I do not want him to
accept
the call on the phone and then on the Web app.
Best regards
Peter
Brian West schrieb:
click on the AA button? :)
/b
On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote:
What is the best way to have this done? Move the call to park
and
then
retransfer again with intercom, or is there a better solution?
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Re: [Freeswitch-users] Force SIP UA to pick up call during ringing?
Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. Best regards Peter Brian West schrieb: click on the AA button? :) /b On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? ___ 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 ___ 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
Re: [Freeswitch-users] Force SIP UA to pick up call during ringing?
The only way I can think to do this today would be to cancel the call and re send with the intercom headers for a phone that supports it. It may be possible to send a reinvite with autoanswer headers but I doubt that would work, all you could do is try making code to do it it a sipp or sipsak scenario and test it. A better aproach might be to answer the call normally and detect that to start your web workflow or not really ring the phone, just the web app and deliver the call with autoanswer when the button is hit in the web ui. Mike On Jun 16, 2009, at 4:24 AM, Peter P GMX prometheus...@gmx.net wrote: Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. Best regards Peter Brian West schrieb: click on the AA button? :) /b On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? ___ 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 ___ 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 ___ 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
Re: [Freeswitch-users] Force SIP UA to pick up call during ringing?
Peter P GMX wrote: Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. is there any reason you don't make your web app listen to event socket or event sink to catch the answer event and start the workflow? then you just need to answer the call on the softphone and the webapp should automatically start the workflow. -Ray ___ 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
Re: [Freeswitch-users] Force SIP UA to pick up call during ringing?
Hello Ray, I do use event socket and it pushes me a link on the website whenever a call for this agent comes in. It's just a matter of visibility. The agent may still finish his old workflow and is still entering data. When a call comes in then and he picks up the phone, the data he just entered is gone away. So I would like the web app to drive answering the call. It gives a better visibility about what he is doing to the callcenter agent. Best regards Peter Raymond Chandler schrieb: Peter P GMX wrote: Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. is there any reason you don't make your web app listen to event socket or event sink to catch the answer event and start the workflow? then you just need to answer the call on the softphone and the webapp should automatically start the workflow. -Ray ___ 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 ___ 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
Re: [Freeswitch-users] Force SIP UA to pick up call during ringing?
Hello Michael, I want the phone be ringing, just for acoustical feedback reasons. But what if I * transfer it to the same user destination again (now with intercom enabled), will this work? * transfer it to park and then transfer it to the same destination again (now with intercom enabled) Best regards Peter Michael Jerris schrieb: The only way I can think to do this today would be to cancel the call and re send with the intercom headers for a phone that supports it. It may be possible to send a reinvite with autoanswer headers but I doubt that would work, all you could do is try making code to do it it a sipp or sipsak scenario and test it. A better aproach might be to answer the call normally and detect that to start your web workflow or not really ring the phone, just the web app and deliver the call with autoanswer when the button is hit in the web ui. Mike On Jun 16, 2009, at 4:24 AM, Peter P GMX prometheus...@gmx.net wrote: Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. Best regards Peter Brian West schrieb: click on the AA button? :) /b On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? ___ 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 ___ 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 ___ 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 ___ 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
Re: [Freeswitch-users] Force SIP UA to pick up call during ringing?
Why not just keep the agent off hook.. in park state... then just playback ringing before you bridge? /b On Jun 16, 2009, at 7:38 AM, Peter P GMX wrote: Hello Michael, I want the phone be ringing, just for acoustical feedback reasons. But what if I * transfer it to the same user destination again (now with intercom enabled), will this work? * transfer it to park and then transfer it to the same destination again (now with intercom enabled) Best regards Peter ___ 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
Re: [Freeswitch-users] Force SIP UA to pick up call during ringing?
The transfer should work but it sounds like offhook agents is what your really trying to accomplish so I would go with brian's suggestion. On Jun 16, 2009, at 7:38 AM, Peter P GMX prometheus...@gmx.net wrote: Hello Michael, I want the phone be ringing, just for acoustical feedback reasons. But what if I * transfer it to the same user destination again (now with intercom enabled), will this work? * transfer it to park and then transfer it to the same destination again (now with intercom enabled) Best regards Peter Michael Jerris schrieb: The only way I can think to do this today would be to cancel the call and re send with the intercom headers for a phone that supports it. It may be possible to send a reinvite with autoanswer headers but I doubt that would work, all you could do is try making code to do it it a sipp or sipsak scenario and test it. A better aproach might be to answer the call normally and detect that to start your web workflow or not really ring the phone, just the web app and deliver the call with autoanswer when the button is hit in the web ui. Mike On Jun 16, 2009, at 4:24 AM, Peter P GMX prometheus...@gmx.net wrote: Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. Best regards Peter Brian West schrieb: click on the AA button? :) /b On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? ___ 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 ___ 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 ___ 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 ___ 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 ___ 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
Re: [Freeswitch-users] Force SIP UA to pick up call during ringing?
I actually do that with our call center application. For all incoming calls, our IVR engine parks the call in a virtual extension and plays back prompts, advertisements, MOH, process digits, etc. When the queue management finds an available agent, it sends an event to the client application for that agent (with an optional screen-pop) where the agent can click Answer Call and then we transfer the call with the auto-answer header set on to the agent phone. You could take a similar approach, if you're worrying about only providing ring-back tone to the caller you can simply park the call and use the playback app to play a tone_stream until the agent clicks the web link, which will transfer the call from the parking extension to the agent with the auto-answer flag. I'm still willing to make some tests with REINVITE providing auto-answer headers, as suggested by Mike. That would provide a more generic way to answer calls programmatically when it's already ringing the endpoint. I just need to find some time to read the sofia code and figure out how to do that :) Regards, Raul On Tue, 2009-06-16 at 02:19 +0200, Peter P GMX wrote: I have managed to have a realtme status of a phone on a web page with event_socket and a push service to the web bowser. What I am now trying to do is roughly the following: * when a call comes in, a flashing banner appears on the web page with an underlying link (this works so far) * when the user klicks on this flashing banner, the external SIP UA which is already ringing, shall pick up the call. I know that it's possible to autoanswer a call with the intercom feature. Also the SIP client X-Lite which we use here is able to autoanswer a call. I however want to manually decide when the UA takes the call with the following workflow: * X-Lite rings on incoming call * user klicks on the flashing banner * X-Lite takes the call What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? Best regards Peter ___ 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 ___ 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
Re: [Freeswitch-users] Force SIP UA to pick up call during ringing?
It mainly works now by uuid_transfer the following way via event socket. uuid_setvar unique_id sip_invite_params intercom=true uuid_setvar unique_id sip_auto_answer true uuid_transfer unique_id 1000 XML default so the call is transferred from 1000 to 1000. What happens: 1) If I disable intercom on the Snom phone, the phone rings, stops ringing and rings again (ok) 1) If I enable intercom on the Snom phone, the phone rings, stops ringing and hangs up (not ok) So I do not get the Snom to pick up the call in intercom mode. The last invite is: INVITE sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib SIP/2.0 Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF Route: sip:1...@217.24.11.189:2752;transport=tls;line=er6kxnib Max-Forwards: 68 From: Peter FS sip:723...@217.xx.xx.xxx;tag=9eQ8rjQy533HF To: sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e CSeq: 116467629 INVITE Contact: sip:mod_so...@217.xx.xx.xxx:5061;transport=tls Call-Info: sip:217.xx.xx.xxx;answer-after=0 The intercom part is there and the Call-Info line with answer-after also. The phone answers with SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF From: Peter FS sip:723...@217.xx.xx.xxx;tag=9eQ8rjQy533HF To: sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true;tag=71rskygkr2 Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e CSeq: 116467629 INVITE Contact: sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;reg-id=1 WWW-Authenticate: Digest realm=sip2.mycompany.de, nonce=2ee26efe6ab27f88, algorithm=MD5 Content-Length: 0 and hangs up. Anybody know how to solve this Snom intercom issue? Best regards Peter Michael Jerris schrieb: The transfer should work but it sounds like offhook agents is what your really trying to accomplish so I would go with brian's suggestion. On Jun 16, 2009, at 7:38 AM, Peter P GMX prometheus...@gmx.net wrote: Hello Michael, I want the phone be ringing, just for acoustical feedback reasons. But what if I * transfer it to the same user destination again (now with intercom enabled), will this work? * transfer it to park and then transfer it to the same destination again (now with intercom enabled) Best regards Peter Michael Jerris schrieb: The only way I can think to do this today would be to cancel the call and re send with the intercom headers for a phone that supports it. It may be possible to send a reinvite with autoanswer headers but I doubt that would work, all you could do is try making code to do it it a sipp or sipsak scenario and test it. A better aproach might be to answer the call normally and detect that to start your web workflow or not really ring the phone, just the web app and deliver the call with autoanswer when the button is hit in the web ui. Mike On Jun 16, 2009, at 4:24 AM, Peter P GMX prometheus...@gmx.net wrote: Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. Best regards Peter Brian West schrieb: click on the AA button? :) /b On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? ___ 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 ___ 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 ___ 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 ___ 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 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org
Re: [Freeswitch-users] Force SIP UA to pick up call during ringing?
uuid_setvar unique_id sip_invite_params intercom=true should be unnecessary. Mike On Jun 16, 2009, at 1:49 PM, Peter P GMX wrote: It mainly works now by uuid_transfer the following way via event socket. uuid_setvar unique_id sip_invite_params intercom=true uuid_setvar unique_id sip_auto_answer true uuid_transfer unique_id 1000 XML default so the call is transferred from 1000 to 1000. What happens: 1) If I disable intercom on the Snom phone, the phone rings, stops ringing and rings again (ok) 1) If I enable intercom on the Snom phone, the phone rings, stops ringing and hangs up (not ok) So I do not get the Snom to pick up the call in intercom mode. The last invite is: INVITE sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib SIP/2.0 Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF Route: sip:1...@217.24.11.189:2752;transport=tls;line=er6kxnib Max-Forwards: 68 From: Peter FS sip:723...@217.xx.xx.xxx;tag=9eQ8rjQy533HF To: sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e CSeq: 116467629 INVITE Contact: sip:mod_so...@217.xx.xx.xxx:5061;transport=tls Call-Info: sip:217.xx.xx.xxx;answer-after=0 The intercom part is there and the Call-Info line with answer-after also. The phone answers with SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF From: Peter FS sip:723...@217.xx.xx.xxx;tag=9eQ8rjQy533HF To: sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true ;tag=71rskygkr2 Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e CSeq: 116467629 INVITE Contact: sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;reg-id=1 WWW-Authenticate: Digest realm=sip2.mycompany.de, nonce=2ee26efe6ab27f88, algorithm=MD5 Content-Length: 0 and hangs up. Anybody know how to solve this Snom intercom issue? Best regards Peter Michael Jerris schrieb: The transfer should work but it sounds like offhook agents is what your really trying to accomplish so I would go with brian's suggestion. On Jun 16, 2009, at 7:38 AM, Peter P GMX prometheus...@gmx.net wrote: Hello Michael, I want the phone be ringing, just for acoustical feedback reasons. But what if I * transfer it to the same user destination again (now with intercom enabled), will this work? * transfer it to park and then transfer it to the same destination again (now with intercom enabled) Best regards Peter Michael Jerris schrieb: The only way I can think to do this today would be to cancel the call and re send with the intercom headers for a phone that supports it. It may be possible to send a reinvite with autoanswer headers but I doubt that would work, all you could do is try making code to do it it a sipp or sipsak scenario and test it. A better aproach might be to answer the call normally and detect that to start your web workflow or not really ring the phone, just the web app and deliver the call with autoanswer when the button is hit in the web ui. Mike On Jun 16, 2009, at 4:24 AM, Peter P GMX prometheus...@gmx.net wrote: Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. Best regards Peter Brian West schrieb: click on the AA button? :) /b On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? ___ 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 ___ 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 ___ 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 ___ 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
Re: [Freeswitch-users] Force SIP UA to pick up call during ringing?
Thanks Michael,
I have disabled it now.
I finally got it to work, (sip_h_Call-Info=sip:$${domain};answer-after=0)
but the behaviour was not as desired, as I didn't manage the phone to
pick up the call on the headset. It will only have the speaker enabled.
So I will have to go a different way with parking the call and then
forward it.
Best regards
Peter
Michael Jerris schrieb:
uuid_setvar unique_id sip_invite_params intercom=true should be
unnecessary.
Mike
On Jun 16, 2009, at 1:49 PM, Peter P GMX wrote:
It mainly works now by uuid_transfer the following way via event
socket.
uuid_setvar unique_id sip_invite_params intercom=true
uuid_setvar unique_id sip_auto_answer true
uuid_transfer unique_id 1000 XML default
so the call is transferred from 1000 to 1000.
What happens:
1) If I disable intercom on the Snom phone, the phone rings, stops
ringing and rings again (ok)
1) If I enable intercom on the Snom phone, the phone rings, stops
ringing and hangs up (not ok)
So I do not get the Snom to pick up the call in intercom mode.
The last invite is:
INVITE sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib
SIP/2.0
Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF
Route: sip:1...@217.24.11.189:2752;transport=tls;line=er6kxnib
Max-Forwards: 68
From: Peter FS sip:723...@217.xx.xx.xxx;tag=9eQ8rjQy533HF
To:
sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true
Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e
CSeq: 116467629 INVITE
Contact: sip:mod_so...@217.xx.xx.xxx:5061;transport=tls
Call-Info: sip:217.xx.xx.xxx;answer-after=0
The intercom part is there and the Call-Info line with answer-after
also.
The phone answers with
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF
From: Peter FS sip:723...@217.xx.xx.xxx;tag=9eQ8rjQy533HF
To:
sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true
;tag=71rskygkr2
Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e
CSeq: 116467629 INVITE
Contact:
sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;reg-id=1
WWW-Authenticate: Digest realm=sip2.mycompany.de,
nonce=2ee26efe6ab27f88, algorithm=MD5
Content-Length: 0
and hangs up.
Anybody know how to solve this Snom intercom issue?
Best regards
Peter
Michael Jerris schrieb:
The transfer should work but it sounds like offhook agents is what
your really trying to accomplish so I would go with brian's
suggestion.
On Jun 16, 2009, at 7:38 AM, Peter P GMX prometheus...@gmx.net
wrote:
Hello Michael,
I want the phone be ringing, just for acoustical feedback reasons.
But what if I
* transfer it to the same user destination again (now with
intercom
enabled), will this work?
* transfer it to park and then transfer it to the same destination
again (now with intercom enabled)
Best regards
Peter
Michael Jerris schrieb:
The only way I can think to do this today would be to cancel the
call
and re send with the intercom headers for a phone that supports it.
It may be possible to send a reinvite with autoanswer headers but I
doubt that would work, all you could do is try making code to do it
it
a sipp or sipsak scenario and test it. A better aproach might be
to
answer the call normally and detect that to start your web workflow
or
not really ring the phone, just the web app and deliver the call
with
autoanswer when the button is hit in the web ui.
Mike
On Jun 16, 2009, at 4:24 AM, Peter P GMX prometheus...@gmx.net
wrote:
Hello Brian,
this is too easy :-).
This is for a small callcenter app and I only want the user to
pickup
the call once (to accept the call in X-Lite (or a Snom phone)
and to
start the workflow on the web application). I do not want him to
accept
the call on the phone and then on the Web app.
Best regards
Peter
Brian West schrieb:
click on the AA button? :)
/b
On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote:
What is the best way to have this done? Move the call to park
and
then
retransfer again with intercom, or is there a better solution?
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