<div class="gmail_quote">On Sat, Mar 7, 2009 at 2:20 PM, Johann Steinwendtner <span dir="ltr"><<a href="mailto:steinwendtner@gmx.net">steinwendtner@gmx.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div><div></div><div class="h5">John Todd wrote:<br>
> Just a suggestion: have you tried more recent versions of Asterisk<br>
> with IAX2? I'm uncertain what version you're using, and if it's<br>
> 1.2.4, that's getting to be quite old and the problems that you<br>
> reference may already be solved in more recent updates.<br>
><br>
> In addition, if you're set on SIP, there are features in newer<br>
> versions of Asterisk which allow you to both set and read SIP headers,<br>
> so you can insert values in those headers between Asterisk instances<br>
> which could then be used by the dialplan to split your calls apart<br>
> into different contexts or behaviors.<br>
><br>
> See function "SIP_HEADER" and application "SIPAddHeader" for the most<br>
> recent versions of Asterisk.<br>
><br>
> JT<br>
><br>
><br>
> On Mar 6, 2009, at 11:29 AM, tracinet wrote:<br>
><br>
>> That stinks... We are migrating to SIP from IAX2 at the moment and<br>
>> running into the same exact problem. No way to control the<br>
>> destination context unless you use the "fromuser". Of course that<br>
>> is rendering Caller ID useless as you pointed out.<br>
>><br>
>> I am still researching this though, if I find anything I will post<br>
>> it here...<br>
>><br>
>><br>
>> On Fri, Mar 6, 2009 at 2:13 PM, Adam Robins<br>
>> <<a href="mailto:arobins@pharmacentra.com">arobins@pharmacentra.com</a>> wrote:<br>
>> no<br>
>><br>
>><br>
>> From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>
>> ] On Behalf Of tracinet<br>
>> Sent: Friday, March 06, 2009 2:08 PM<br>
>> To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
>> Subject: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using<br>
>> SIP<br>
>><br>
>><br>
>><br>
>> On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins<br>
>> <<a href="mailto:arobins@pharmacentra.com">arobins@pharmacentra.com</a>> wrote:<br>
>><br>
>><br>
>> I am switching from IAX2 to SIP for my inter-Asterisk transport due to<br>
>> assorted quality issues following the 1.2.4 upgrade.<br>
>><br>
>> On the server that SENDS the call, I have the following in SIP.CONF:<br>
>><br>
>> [192.168.1.2_OB]<br>
>> type=peer<br>
>> fromuser=OB<br>
>> host=192.168.1.2<br>
>><br>
>> And in EXTENSIONS.CONF<br>
>><br>
>> exten => 91NXXNXXXXXX,1,Dial(SIP/${EXTEN}@192.168.1.2_OB)<br>
>><br>
>><br>
>> On the RECEIVING Server in SIP.CONF:<br>
>><br>
>> [OB]<br>
>> type=user<br>
>> context=longdistance<br>
>><br>
>><br>
>> I am not using a REGISTER statement on the receiving server.<br>
>><br>
>> My problem is that the only way I can seem to get the call delivered<br>
>> into the proper SIP context on the receiving box is to use the<br>
>> "fromuser=OB" on the sending machine. I tried using "username=OB",<br>
>> but<br>
>> then it delivers into the default context. I don't want to use<br>
>> "fromuser" because it overrides the callerid.<br>
><br>
<br>
</div></div>You should be able to solve the callerid issue by using the sendrpid and<br>
trustrpid prompts.<br>
<br>
Hans<br>
<div><div></div><div class="h5"><br>
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</div></div></blockquote></div><br><br>Hans,<br>Thanks for the tip - sendrpid and trustrpid is SOOOOO much more elegant!!!!<br><br>Works like a charm!!!<br><br>Thanks!<br><br>Pedro<br>