You mean the SIP TO you receive does not contain a @domain part ? Hmmm, I am sure some guys around will have some comments to do about that :) David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 25/05/2010 à 21:48, RR a écrit :
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 > > > > > > > _______________________________________________ > 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 > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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 <mailto:MSN%253Aanthony_minessale@hotmail.com>> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> > <mailto:PAYPAL%3Aanthony.minessale@gmail.com <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
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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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