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 INVITEFrom: "16469NNNNNN" <sip:16469NNNNNN;phone-context=+1@67.1x.xxx.xxx:5060;user=phone>;tag=SDru6fc01-gK0c10a887.(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 <mailto: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:MSN%253Aanthony_minessale@hotmail.com>>> 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
> >
> >
> >
> >
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users@lists.freeswitch.org
> <mailto:Freeswitch-users@lists.freeswitch.org>
> > <mailto:Freeswitch-users@lists.freeswitch.org
> <mailto:Freeswitch-users@lists.freeswitch.org>>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
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> > http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > 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>
> > <mailto:MSN%3Aanthony_minessale@hotmail.com
> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com> <mailto:PAYPAL%253Aanthony.minessale@gmail.com>>
> <mailto:PAYPAL%3Aanthony.minessale@gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale@gmail.com
> > IRC: irc.freenode.net <http://irc.freenode.net>> <mailto:sip%253A888@conference.freeswitch.org>>
> <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
> > iax:guest@conference.freeswitch.org/888> <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org>>
> <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
> > pstn:213-799-1400
> >
> ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users@lists.freeswitch.org
> <mailto:Freeswitch-users@lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
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> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users@lists.freeswitch.org
> <mailto:Freeswitch-users@lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
>
>
>
> --
> 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>
> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
> <mailto:PAYPAL%3Aanthony.minessale@gmail.com>
> 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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> Freeswitch-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
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