[Freeswitch-users] Force SIP UA to pick up call during ringing?

Peter P GMX Prometheus001 at gmx.net
Tue Jun 16 13:11:05 PDT 2009


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:1000 at 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:1000 at 217.24.11.189:2752>;transport=tls;line=er6kxnib
>>    Max-Forwards: 68
>>    From: "Peter FS" <sip:723323 at 217.xx.xx.xxx>;tag=9eQ8rjQy533HF
>>    To:
>> <sip:1000 at 192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true 
>>     
>>    Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e
>>    CSeq: 116467629 INVITE
>>    Contact: <sip:mod_sofia at 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:723323 at 217.xx.xx.xxx>;tag=9eQ8rjQy533HF
>>    To:
>> <sip:1000 at 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:1000 at 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 <Prometheus001 at 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 <Prometheus001 at 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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>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
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>>>>>>
>>>>>>             
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>>>>>
>>>>>
>>>>>           
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>
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