Still, regardless of the specific requirements. You've mentioned that it's a analog phone.. And thus you won't have a connected line to it without "calling" it.<div><br class="webkit-block-placeholder">
</div><div>Maybe you could tell us more about it and we'd be able to give you some idea of how you might be able to do it?</div><div>-Brett</div><div><br><br><div class="gmail_quote">On Mon, Nov 24, 2008 at 12:56 PM, 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 brett.<br>
<br>
I really would be able to re-think what i have to do.<br>
But the device is not on my control and i need to satisfy my customer.<br>
However i don't need to speak with this device before the call is<br>
started but immediately after the caller response.<br>
And i can't do it from the caller.<br>
When i have read about SIP INFO DTMF i have decided to try if it can<br>
work for me, but its not so, because a sip proxy cant generate transaction.<br>
<br>
The only think i can try now is to inject RTP.<br>
I use mediaproxy for every call that come from those devices, so i see<br>
two possibility.<br>
The first, send RTP to media-relay caller address/port (and maybe i also<br>
need to send it from ip/port of the caller, using raw socket) so it than<br>
be relayed to the callee.<br>
The second is to send directly to callee address/port audio rtp from<br>
media-relay server, using the same address/port used by the relay for<br>
this call audio stream, but if so i think i have to do it before the<br>
port become used by conntrack, isnt it ?<br>
Please send me any comment/suggestion.<br>
<br>
Regards.<br>
<div><div></div><div class="Wj3C7c"><br>
Brett Nemeroff wrote:<br>
> I really, think you need to re-think what you are trying to do. It simply<br>
> doesn't make any sense the scenario you've described.<br>
> I imagine your mystery device is something like a door control. If it's<br>
> analog, and you are using some sort of ATA, it doesn't make sense to send it<br>
> a DTMF before you make a call to it.. That would be like leaving a message<br>
> on your home answering machine without ever calling it. If your analog<br>
> mystery device requires a telephone line, you'll need to place a call to it<br>
> first.. Once you have the dialog established, it should be pretty easy to<br>
> send it a tone from whatever established the dialog.<br>
> -Brett<br>
><br>
><br>
> On Sun, Nov 23, 2008 at 9:10 AM, Giuseppe Roberti <<a href="mailto:jnod@jnod.org">jnod@jnod.org</a>> wrote:<br>
><br>
>> 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>
>><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>><br>
>> wrote:<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>><br>
>> wrote:<br>
>>>>>>> Ill try.<br>
>>>>>>><br>
>>>>>>> I have a custom hardware phone that do something when receive a dtmf<br>
>> (it<br>
>>>>>>> cant read sip, its an analogical phone).<br>
>>>>>>> So i have to send dtmf when some event occur (like, received new<br>
>> 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<br>
>> it.<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<br>
>> the ctd example works??<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<br>
>> message<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<br>
>> 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<br>
>> 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>
>>> 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>
>> Giuseppe Roberti<br>
>> <<a href="mailto:jnod@jnod.org">jnod@jnod.org</a>><br>
>><br>
><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>