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Well, in this case you need to test for the status code in the failure
route. If you fork off several INVITEs in your main route, you don't
get into the failure route until all branches failed. So, I still don't
understand how you do it. <br>
Anyway, you want to avoid it, because you have a gateway that does not
follow the RFC...<br>
g-)<br>
<br>
Zappasodi Daniele wrote:
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<title>R: [Serusers] sequential forking and merged request</title>
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<p><font size="2">I have written a module that choose the next client
reading the destination from a db, but I can easily obtain the same
scenario with the basic serial forking example:<br>
<br>
route{<br>
[...]<br>
t_on_failure("1");<br>
t_relay();<br>
}<br>
<br>
failure_route[1] {<br>
append_branch(<a class="moz-txt-link-rfc2396E" href="mailto:sip:number1@mygw.com">"sip:number1@mygw.com"</a>);<br>
log(1,"first redirection\n");<br>
t_on_failure("2");<br>
t_relay();<br>
}<br>
<br>
failure_route[2] {<br>
append_branch(<a class="moz-txt-link-rfc2396E" href="mailto:sip:number2@mygw.com">"sip:number2@mygw.com"</a>);<br>
log(1, "second redirection to the same gateway, but different
number\n");<br>
t_relay();<br>
}<br>
<br>
<br>
<br>
-----Messaggio originale-----<br>
Da: Greger V. Teigre [<a moz-do-not-send="true"
href="mailto:greger@teigre.com">mailto:greger@teigre.com</a>]<br>
Inviato: lun 07/05/2007 15.48<br>
A: Zappasodi Daniele<br>
Cc: <a class="moz-txt-link-abbreviated" href="mailto:serusers@lists.iptel.org">serusers@lists.iptel.org</a><br>
Oggetto: Re: [Serusers] sequential forking and merged request<br>
<br>
Any device implementing UA functionality should follow RFC3261. But I'm<br>
not sure how you manage to fork off two INVITEs to the same gateway? I<br>
would start there.<br>
g-)<br>
<br>
Zappasodi Daniele wrote:<br>
> Hello,<br>
> I have a question about sequential forking and Merged Request.<br>
> If in a forking SER calls two lines of the same client (different
numbers and different Request-URIs) it refuses the second call with 482
(Cisco and Snom multiline phone for example) according with RFC 3261
section 8.2.2.2 because the two INVITEs have the same Call-Id, From tag
and Cseq.<br>
> If this behaviour is reasonable with a phone, it could not be
acceptable for other types of client, first of all SIP-PSTN gateway.<br>
> Is it correct this RFC interpretation? Maybe is this section
inappropriate for gateways? <br>
> Is there something that I can do in the config file?<br>
><br>
><br>
>
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