No subject

<snip>
Replaces: pickup-9582-c0a80101-d-4 at 192.168.101.102
<snip>

This Replaces header refers to RFC3891 which is not yet supported in
Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA)

This INVITE fails with :
<snip>
chan_sip.c: Trying to pick up 7792 at subs
<snip>
app_directed_pickup.c: No target channel found for 7792.


If I'm dialing *87792 instead of using BLF, then I'm entering the dialplan
part in which there is Pickup(${EXTEN:2}@PICKUPMARK) and the call is
correctly pickup.



So my understanding is :
when upgrading from 1.6.1 to 1.6.2, Asterisk must somehow advertise a newly
supported SIP capability which is now used by ST2030S hardphones to build
Pickup requests.

My question is :
- is my understanding correct ?
- if positive, is there a way to tame asterisk to behave appropropriately ?

Regards

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Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hello,<br><br>I&#39;m using Thomson/Technicolor ST2030S hardphones with Ast=
erisk 1.6.<br>Changing from 1.6.1.18 to 1.6.2.6, I can see a change in Pick=
up&#39;s behaviour and I&#39;m a bit confused about it.<br><br>With 1.6.2.6=
, when extension 7791 is calling extension 7792, I can see INVITE messages =
coming in and out Asterisk.<br>
I can also see a NOTIFY message advertising this call to subscriber 7793, f=
or instance.<br>Here is an extract of this message :<br><br>NOTIFY sip:7793=
@192.168.101.102:5060;user=3Dphone SIP/2.0<br>&lt;snip&gt;<br>Call-ID: <a h=
ref=3D"mailto:7019-c0a80101-d-3 at 192.168.101.102">7019-c0a80101-d-3 at 192.168.=
101.102</a><br>
&lt;snip&gt;<br>Content-Length: 212<br><br><br>From then, if BLF 7792 on ex=
tension 7793 is pressed, then an INVITE message is send with :<br>INVITE si=
p:*87792 at 192.168.101.240:5060;user=3Dphone SIP/2.0<br>&lt;snip&gt;<br>Repla=
ces: <a href=3D"mailto:pickup-9582-c0a80101-d-4 at 192.168.101.102">pickup-958=
2-c0a80101-d-4 at 192.168.101.102</a><br>
&lt;snip&gt;<br><br>This Replaces header refers to RFC3891 which is not yet=
 supported in Asterisk (see <a href=3D"http://www.voip-info.org/wiki/view/A=
sterisk+SLA">http://www.voip-info.org/wiki/view/Asterisk+SLA</a>)<br><br>
This INVITE fails with :<br>&lt;snip&gt;<br>chan_sip.c: Trying to pick up 7=
792 at subs<br>&lt;snip&gt;<br>app_directed_pickup.c: No target channel found =
for 7792.<br><br><br>If I&#39;m dialing *87792 instead of using BLF, then I=
&#39;m entering the dialplan part in which there is Pickup(${EXTEN:2}@PICKU=
PMARK) and the call is correctly pickup.<br>
<br><br><br>So my understanding is :<br>when upgrading from 1.6.1 to 1.6.2,=
 Asterisk must somehow advertise a newly supported SIP capability which is =
now used by ST2030S hardphones to build Pickup requests.<br><br>My question=
 is :<br>
- is my understanding correct ?<br>- if positive, is there a way to tame as=
terisk to behave appropropriately ?<br><br>Regards<br><br><br>

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