I really, think you need to re-think what you are trying to do. It simply doesn't make any sense the scenario you've described. <div><br></div><div>I imagine your mystery device is something like a door control. If it's analog, and you are using some sort of ATA, it doesn't make sense to send it a DTMF before you make a call to it.. That would be like leaving a message on your home answering machine without ever calling it. If your analog mystery device requires a telephone line, you'll need to place a call to it first.. Once you have the dialog established, it should be pretty easy to send it a tone from whatever established the dialog.</div>
<div>-Brett</div><div><br class="webkit-block-placeholder"></div><div><br><div class="gmail_quote">On Sun, Nov 23, 2008 at 9:10 AM, Giuseppe Roberti <span dir="ltr"><<a href="mailto:jnod@jnod.org">jnod@jnod.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Hi Bogdan.<br>
<br>
Sorry, i cannot reply to your request.<br>
Simply i don't know if the INFO will be a in-dialog request or not.<br>
I am learning sip and opensips day by day.<br>
<br>
Thank you, i will try t_uac_dlg()<br>
<br>
Regards.<br>
<div><div></div><div class="Wj3C7c"><br>
Bogdan-Andrei Iancu wrote:<br>
> Hi Brett,<br>
><br>
> I think you refer to the TM module (not UAC) for generating SIP requests<br>
> - the t_uac_dlg MI command:<br>
> <a href="http://www.opensips.org/html/docs/modules/1.4.x/tm.html#id2493975" target="_blank">http://www.opensips.org/html/docs/modules/1.4.x/tm.html#id2493975</a><br>
><br>
> As Giuseppe mentioned, the DTMFs he needs are transported via SIP INFO<br>
> method, so OpenSIPS could generate such a method. The problem is (please<br>
> correct me if I'm wrong) that the INFO will a in-dialog request (INFO<br>
> request will belong to the INVITE dialog). And OpenSIPS cannot generate<br>
> within the dialog request (because of the old Cseq story).<br>
><br>
> Regards,<br>
> Bogdan<br>
><br>
> Brett Nemeroff wrote:<br>
>> I've mentioned several times.. try using the UAC module to generate<br>
>> the message. OpenSIPs is a proxy. It isn't made to GENERATE any<br>
>> message, but to relay and reply to messages. You can use the fifo<br>
>> method of the UAC module to generate an outgoing message. Look up the<br>
>> click to dial (ctd) examples. It's not for SIP INFO, but it has an<br>
>> INVITE method.<br>
>><br>
>> On the other hand:<br>
>> sipp (or some sort of real UAC)----> opensips ---><br>
>> your-fancy-device-we-don't-understand<br>
>> makes plenty of sense with the given technology<br>
>><br>
>> That's my $0.02<br>
>><br>
>> On Mon, Nov 17, 2008 at 10:41 PM, Giuseppe Roberti <<a href="mailto:jnod99@gmail.com">jnod99@gmail.com</a>> wrote:<br>
>><br>
>>> I need only to send an INFO request that describe a DTMF tone, an INFO<br>
>>> application/dtmf-relay request. I dont need sipp nor any other media<br>
>>> software.<br>
>>> Take a look to <a href="http://www.voip-info.org/wiki/view/SIP+Info+DTMF" target="_blank">http://www.voip-info.org/wiki/view/SIP+Info+DTMF</a><br>
>>> Its only a think that is relative to uac instead of proxy.<br>
>>> Maybe its better to talk about on other places.<br>
>>><br>
>>> Regards.<br>
>>><br>
>>> Brett Nemeroff wrote:<br>
>>><br>
>>>> Perhaps you can use sipp to generate a call and send RTP<br>
>>>><br>
>>>> On Mon, Nov 17, 2008 at 7:24 PM, Giuseppe Roberti <<a href="mailto:jnod@jnod.org">jnod@jnod.org</a>> wrote:<br>
>>>><br>
>>>>> Ill try.<br>
>>>>><br>
>>>>> I have a custom hardware phone that do something when receive a dtmf (it<br>
>>>>> cant read sip, its an analogical phone).<br>
>>>>> So i have to send dtmf when some event occur (like, received new invite,<br>
>>>>> 200 ok for invite, etc...) to a sip ata that encode it for the<br>
>>>>> analogical hardware.<br>
>>>>> I know that a proxy does not generate request but i personally need it.<br>
>>>>><br>
>>>>> Thanks again.<br>
>>>>><br>
>>>>> <a href="mailto:brett@nemeroff.com">brett@nemeroff.com</a> wrote:<br>
>>>>><br>
>>>>>> What about using the uac module and generating it manually, like how the ctd example works??<br>
>>>>>><br>
>>>>>> I'm not sure why you'd want to do this....<br>
>>>>>> Sent from my Verizon Wireless BlackBerry<br>
>>>>>><br>
>>>>>> -----Original Message-----<br>
>>>>>> From: Alex Balashov <<a href="mailto:abalashov@evaristesys.com">abalashov@evaristesys.com</a>><br>
>>>>>><br>
>>>>>> Date: Mon, 17 Nov 2008 19:25:31<br>
>>>>>> To: Giuseppe Roberti<<a href="mailto:jnod@jnod.org">jnod@jnod.org</a>><br>
>>>>>> Cc: <<a href="mailto:users@lists.opensips.org">users@lists.opensips.org</a>><br>
>>>>>> Subject: Re: [OpenSIPS-Users] Generate INFO application/dtmf-relay message<br>
>>>>>><br>
>>>>>><br>
>>>>>> The fact remains that a proxy cannot originate requests. They would<br>
>>>>>> have to come from another UAC.<br>
>>>>>><br>
>>>>>> Giuseppe Roberti wrote:<br>
>>>>>><br>
>>>>>><br>
>>>>>>> Hi.<br>
>>>>>>><br>
>>>>>>> Sorry. I'll try to explain better.<br>
>>>>>>> I want to send an INFO request from the UAS to one UAC.<br>
>>>>>>> This request is a SIP request, specified by rfc 2976, it's not RTP media.<br>
>>>>>>> It's like this: <a href="http://www.voip-info.org/wiki/view/SIP+Info+DTMF" target="_blank">http://www.voip-info.org/wiki/view/SIP+Info+DTMF</a><br>
>>>>>>><br>
>>>>>>> Thank you all.<br>
>>>>>>><br>
>>>>>>> Bogdan-Andrei Iancu wrote:<br>
>>>>>>><br>
>>>>>>>> Hi Giuseppe,<br>
>>>>>>>><br>
>>>>>>>> OpenSIPS is a SIP proxy and has no media related capabilities, so it is<br>
>>>>>>>> not able to generate DTMF tones.<br>
>>>>>>>><br>
>>>>>>>> Regards,<br>
>>>>>>>> Bogdan<br>
>>>>>>>><br>
>>>>>>>> Giuseppe Roberti wrote:<br>
>>>>>>>><br>
>>>>>>>>> Hi.<br>
>>>>>>>>><br>
>>>>>>>>> I would able to generate a DTMF from opensips.<br>
>>>>>>>>> Is it possible ? How can i do it ?<br>
>>>>>>>>><br>
>>>>>>>>> Regards.<br>
>>>>>>>>><br>
>>>>>>>>><br>
>>>>>>>>><br>
>>>>> --<br>
>>>>> Giuseppe Roberti<br>
>>>>> <<a href="mailto:jnod@jnod.org">jnod@jnod.org</a>><br>
>>>>><br>
>>>>><br>
>>> --<br>
>>> Giuseppe Roberti<br>
>>> <<a href="mailto:jnod@jnod.org">jnod@jnod.org</a>><br>
>>><br>
>>><br>
>> _______________________________________________<br>
>> Users mailing list<br>
>> <a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>
>> <a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
>><br>
>><br>
><br>
><br>
> _______________________________________________<br>
> Users mailing list<br>
> <a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>
> <a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
<br>
<br>
</div></div>--<br>
<div><div></div><div class="Wj3C7c">Giuseppe Roberti<br>
<<a href="mailto:jnod@jnod.org">jnod@jnod.org</a>><br>
</div></div></blockquote></div><br></div>