Hello I want to follow up on this example from YEARS ago. I had tried using the variable "destination_number" but that didn't work, and I figured that it was because the To: header doesn't have the destination_number but has just the URI, so I thought I'd use sip_to_user instead.

We have calls coming in with the following info in the INVITE

From: "16469NNNNNN" <sip:16469NNNNNN;phone-context=+1@67.1x.xxx.xxx:5060;user=phone>;tag=SDru6fc01-gK0c10a887.
To: <sip:011390NNNNNNNNNN;phone-context=+39@xxx.xxx.xxx.xxx:5060;user=phone>.
(N and x are obviously being masked for privacy)

I use this info in the dialplan like so

<include>
  <extension name="DIDtest">
    <condition field="ani" expression="^(\+?|\+1?|1?)(6469NNNNNN).*$" break="never">
        <action application="set" data="effective_caller_id_number=$2"/>
        <action application="set" data="effective_caller_id_name=$2"/>
    </condition>
    <condition field="${sip_to_user}" expression="^(\+?|\+1?|1?|011?)(390NNNNNNNNNN).$" break="never">
        <action application="set" data="continue_on_fail=false"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="domain_name=$${domain}"/>
        <action application="set" data="bypass_media=true"/>
        <action application="bridge" data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$2"/>
    </condition>
  </extension>
</include>

However, the calls aren't passing the condition in this dialplan and thus not being forwarded to "blade2" server. In fact, even the 011 is not being stripped off. 

What am I doing wrong?

Thanks
\RR 


On Thu, Nov 13, 2008 at 10:52 AM, Peter P GMX <Prometheus001@gmx.net> wrote:
Thanks,

this works

Best regards
Peter

Anthony Minessale schrieb:
> here's another hint.
>
> use field="${sip_to_user}"
>
>
> On Thu, Nov 13, 2008 at 3:43 AM, Peter P GMX <Prometheus001@gmx.net
> <mailto:Prometheus001@gmx.net>> wrote:
>
>     I checked your hint. The variable sip_to_user has the right value
>     - but
>     it is evaluated as empty at the time it is parsed.
>
>     Info gives me:
>     .
>     .
>     variable_sip_to_user: [xxxxxxxxx8910]
>     variable_sip_to_uri: [xxxxxxxxx8910@my.ip.addre.ss]
>
>
>     Example1,:
>     I check with condition field="destination_number"
>     Match Result
>     2008-11-13 10:24:55 [DEBUG] mod_dialplan_xml.c:117 parse_exten()
>     Regex:
>     [Inbound QSC trunk] destination_number(xxxxxxxxx89) =~
>     /^(xxxxxxxxx89)$/
>     This works but gives only the DID of the trunk (10 is missing at
>     the end).
>
>     Example2:
>     I check with condition field="sip_to_user"
>     Match Result
>     2008-11-13 10:30:42 [DEBUG] mod_dialplan_xml.c:117 parse_exten()
>     Regex:
>     [Inbound QSC trunk] sip_to_user() =~ /^(xxxxxxxxx89[0-9.])$/
>     2008-11-13 10:30:42 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex
>     mismatch
>     sip_to_user() is empty
>     Example3:
>     I check with condition field="variable_sip_to_user"
>     Match Result
>     2008-11-13 10:39:53 [DEBUG] mod_dialplan_xml.c:117 parse_exten()
>     Regex:
>     [Inbound QSC trunk] variable_sip_to_user() =~ /^(xxxxxxxxx89[0-9.])$/
>     2008-11-13 10:39:53 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex
>     mismatch
>     variable_sip_to_user() is empty
>
>     So at the time of evaluation "sip_to_user" resp.
>     "variable_sip_to_user"
>     is empty.
>
>     What can I do?
>
>     Best regards
>     Peter
>
>
>     Anthony Minessale schrieb:
>     > route the call to the "info" app and look for the one that has it.
>     >
>     >
>     > On Wed, Nov 12, 2008 at 1:36 PM, Peter P GMX
>     <Prometheus001@gmx.net <mailto:Prometheus001@gmx.net>
>     > <mailto:Prometheus001@gmx.net <mailto:Prometheus001@gmx.net>>>
>     wrote:
>     >
>     >     Dear all,
>     >
>     >     when I get an incoming call from my SIP provider I do receive an
>     >     invite
>     >     on the trunk number, e.g. in Germany 0xxxxx9. However I have an
>     >     extension block of 0xxxxx90 to 0xxxxx999.
>     >     In the dialplan I checked the condition
>     field="destination_number" but
>     >     this compares to the number of the trunk 0xxxxx9, so I do
>     not get the
>     >     direct inward dialling extensions.
>     >
>     >     The SIP to-header contains the direct inward dialling number
>     >     0xxxxx90 to
>     >     0xxxxx999 but I have no success in accessing it in the dialplan.
>     >
>     >     I tried sip_to_user, sip_to_uri, sip_req_user, sip_req_uri,
>     but no
>     >     success. They are all empty.
>     >
>     >     Any idea which variable to parse?
>     >
>     >     Best regards
>     >     Peter
>     >
>     >
>     >
>     >
>     >
>     >
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>     >
>     > --
>     > Anthony Minessale II
>     >
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>     <mailto:PAYPAL%253Aanthony.minessale@gmail.com>>
>     > IRC: irc.freenode.net <http://irc.freenode.net>
>     <http://irc.freenode.net> #freeswitch
>     >
>     > FreeSWITCH Developer Conference
>     > sip:888@conference.freeswitch.org
>     <mailto:sip%3A888@conference.freeswitch.org>
>     > <mailto:sip%3A888@conference.freeswitch.org
>     <mailto:sip%253A888@conference.freeswitch.org>>
>     > iax:guest@conference.freeswitch.org/888
>     <http://iax:guest@conference.freeswitch.org/888>
>     > <http://iax:guest@conference.freeswitch.org/888>
>     > googletalk:conf+888@conference.freeswitch.org
>     <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org>
>     > <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org
>     <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org>>
>     > pstn:213-799-1400
>     >
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> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale@hotmail.com
> <mailto:MSN%3Aanthony_minessale@hotmail.com>
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> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888@conference.freeswitch.org
> <mailto:sip%3A888@conference.freeswitch.org>
> iax:guest@conference.freeswitch.org/888
> <http://iax:guest@conference.freeswitch.org/888>
> googletalk:conf+888@conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
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