[sip-comm-dev] Conference Problem


#1

Hi. All,
What kind of protocol does the conference function in sip-communicator base
on? I tried to connect the client with an Asterisk conference server, we set
a conference number for calling,but when I dialed,it kept reminding me of
inputting the pin number, and I entered the pin number in the number pad,
but it didn't work.
Please tell me why this happened and how I could use it for conference
through a conference server. Our server is based on SIP.
Thanks.

Best Regards.
Demi Xu

···

--
徐春丹


#2

Hi,

I've also find some bugs in dtmf implementation which are now fixed and
must be available in next build 1040.
Make sure your sip account is configured with option "dtmfmode = info".
I've tested this with Asterisk and it
is OK.

Thanks for the report :slight_smile:
damencho

Emil Ivov wrote:

···

Hello Demi,

This seems like a DTMF problem. Right now we only support sending DTMF
through SIP INFO requests. I know someone is working on an
implementation of DTMF through RTP but in the mean time you could make
sure that your Asterisk would accept the tones in the INFO requests.

Cheers
Emil

chundan xu �ߧѧ�ڧ��:
  

Hi. All,
What kind of protocol does the conference function in sip-communicator
base on? I tried to connect the client with an Asterisk conference
server, we set a conference number for calling,but when I dialed,it kept
reminding me of inputting the pin number, and I entered the pin number
in the number pad, but it didn't work.
Please tell me why this happened and how I could use it for conference
through a conference server. Our server is based on SIP.
Thanks.

Best Regards.
Demi Xu

--
�촺��
    
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#3

Thanks for all of your help. I'll check it.
:slight_smile:

···

2008/3/17, Damian Minkov <damencho@damencho.com>:

Hi,

I've also find some bugs in dtmf implementation which are now fixed and
must be available in next build 1040.
Make sure your sip account is configured with option "dtmfmode = info".
I've tested this with Asterisk and it
is OK.

Thanks for the report :slight_smile:
damencho

Emil Ivov wrote:
> Hello Demi,
>
> This seems like a DTMF problem. Right now we only support sending DTMF
> through SIP INFO requests. I know someone is working on an
> implementation of DTMF through RTP but in the mean time you could make
> sure that your Asterisk would accept the tones in the INFO requests.
>
> Cheers
> Emil
>
> chundan xu написа:
>
>> Hi. All,
>> What kind of protocol does the conference function in sip-communicator
>> base on? I tried to connect the client with an Asterisk conference
>> server, we set a conference number for calling,but when I dialed,it
kept
>> reminding me of inputting the pin number, and I entered the pin number
>> in the number pad, but it didn't work.
>> Please tell me why this happened and how I could use it for conference
>> through a conference server. Our server is based on SIP.
>> Thanks.
>>
>> Best Regards.
>> Demi Xu
>>
>> --
>> 徐春丹
>>
>
>
> ---------------------------------------------------------------------
> To unsubscribe, e-mail: dev-unsubscribe@sip-communicator.dev.java.net
> For additional commands, e-mail: dev-help@sip-communicator.dev.java.net
>
>
>

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