Hello With or without the extension parameter, it's exactly the same. Thanks On 01/05/2015 20:04, Stanislav Sinyagin wrote:

Remove the extension parameter and see if it helps.

On May 1, 2015 6:11 PM, "Tanguy" <phenix@vfemail.net <mailto:phenix@vfemail.net>> wrote:

Hello,

My provider did not send correct DID number in the INVITE packet but i can use "To" argument

INVITE sip:gw+0a96c3d3-0b0e-4864-b9ec-759fa4422429@92.222.18.xxx:5080;transport=udp;gw=0a96c3d3-0b0e-4864-b9ec-759fa4422429 <mailto:sip:gw+0a96c3d3-0b0e-4864-b9ec-759fa4422429@92.222.18.xxx:5080;transport=udp;gw=0a96c3d3-0b0e-4864-b9ec-759fa4422429> SIP/2.0. Call-ID: 25016-VB-188fd96b-526e3dbd4@sip.ovh.fr <mailto:25016-VB-188fd96b-526e3dbd4@sip.ovh.fr>. Contact: <sip:10.7.1.60:5060 <http://10.7.1.60:5060>>. Content-Type: application/sdp. CSeq: 403749831 INVITE. From: "0967212xxx" <sip:0967212xxx@sip.ovh.fr;user=phone> <mailto:sip:0967212xxx@sip.ovh.fr;user=phone>;tag=25016-VE-188fd96c-18bb43586. Max-Forwards: 27. Record-Route: <sip:91.121.129.20:5060;lr>. *To: <sip:0557590xxx@10.7.1.60;user=phone> <mailto:sip:0557590xxx@10.7.1.60;user=phone>.* Via: SIP/2.0/UDP 91.121.129.20:5060;branch=z9hG4bK-WGZO-1fe73949-2df58378.

Using asterisk i can bypass the issue using something like exten => s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1) but i am unable to do the same under freeswitch.

My trunk configuration seems correct, as you can see i used auto_to_user, but the destination number remains 0033972480xxx when i call 0557590xxx.

2015-05-01 18:02:12.235827 [INFO] mod_dialplan_xml.c:635 Processing 0967212xxx <0967212xxx>->0033972480xxx in context public

<include> <gateway name="0a96c3d3-0b0e-4864-b9ec-759fa4422429"> <param name="username" value="0033972480xxx"/> <param name="password" value="xxxxxxx"/> <param name="proxy" value="sip.ovh.fr <http://sip.ovh.fr>"/> <param name="expire-seconds" value="800"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> <param name="extension" value="auto_to_user"/> <param name="context" value="public"/> </gateway> </include>

I tried to edit my inbound dialplan manually, it works using <condition field="${sip_to_user}" expression="0557590xxx" > but i prefer a proper way to do this because i will also use telcos with normal invite packets

I how i can copy $sip_to_header to destination for this specific trunk ?

Please note that i use fusionpbx.

Best regards, sorry for my bad English

Hello Chris The file ( generated by fusionpbx ) is located in /usr/local/freeswitch/conf/sip_profiles/external/v_0a96c3d3-0b0e-4864-b9ec-759fa4422429.xml, i also tried to force a number in this field but it did not change anything. Thanks On 01/05/2015 22:11, Chris Tunbridge wrote:

Tanguy,

setting the auto_to_user should set the destination number field to the correct one according to the wiki

"Note: /extension/ parameter influence the contents of channel variable /Caller-Destination-Number/ and /destination_number/. If it is blank, /Caller-Destination-Number/ will always be set to gateway's username. If it has a value, /Caller-Destination-Number/ will always be set to this value. If it has value /auto_to_user/, /Caller-Destination-Number/ will be populated with value /${sip_to_user}/ which means the real dialed number in case of an inbound call."

from: https://wiki.freeswitch.org/wiki/Sofia.conf.xml

where is this config that you listed currently located? (full file path please)

Hello I'm not sure to understand why i should use a so strange trick. Using "internal" profile for external trunk can't be a security leak ? If my provider change something about the 10.7.6.60 gateway, the trunk will no longer work until i update my configuration ? Nevertheless i tried to apply your advice, so i moved by gateway in internal profile and i created a new user /usr/local/freeswitch/conf/dialplan/provider_inbound_calls/user.xml <include> <user id="provider_gw1" cidr="10.7.1.60/32" > <params> <param name="password" value="xxxxxx"/> </params> </user> <variables> <variable name="user_context" value="provider_inbound_calls"/> </variables> </include> Unfortunately it did not work, but i have the good DID number in [0557590xxx@10.7.1.60] 2015-05-02 18:30:15.940354 [DEBUG] sofia.c:9015 IP 91.121.129.20 Rejected by acl "domains". Falling back to Digest auth. 2015-05-02 18:30:15.940354 [WARNING] sofia_reg.c:2827 Can't find user [anonymous@10.7.1.60] from 91.121.129.20 You must define a domain called '10.7.1.60' in your directory and add a user with the id="anonymous" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2015-05-02 18:30:15.940354 [WARNING] sofia_reg.c:1687 SIP auth failure (INVITE) on sofia profile 'internal' for *[0557590xxx@10.7.1.60] *from ip 91.121.129.20 2015-05-02 18:30:15.940354 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/0967212xxx@sip.ovh.fr [BREAK] 2015-05-02 18:30:15.940354 [NOTICE] sofia.c:2063 Hangup sofia/internal/0967212xxx@sip.ovh.fr [CS_NEW] [CALL_REJECTED] Just for fun i created a user anonymous in domain 10.7.1.60 and reloaded with reloadxml but i still have the same message. /usr/local/freeswitch/conf/dialplan/10.7.1.60/user.xml <include> <user id="anonymous" cidr="10.7.1.60/32" > <params> <param name="password" value="xxxxx"/> </params> </user> <variables> <variable name="user_context" value="10.7.1.60"/> </variables> </include> Thanks On 02/05/2015 08:17, Sergey Safarov wrote:

Try 1) link gateway to "internal" profile; 2) create dialplan with name "provider_inbound_calls" and add required extensions; 3) create user "provider_gw1" in directory with attribute cidr="10.7.1.60/32 <http://10.7.1.60/32>" (value from you example), with random value in param "password", and "provider_inbound_calls" value in variable "user_context" After it you can make inbound call from. If provider has several gateways, add user record in directory for each gateway.

Sergey

Hello Sergey I tried to change the cidr attribute with my gateway public IP but this change nothing. My provider did not provide a different SIP accound for each DID number. I actually have 2 DID numbers on this test platform but if i migrate my production servers from asterisk to freeswitch i will manage about 50 DID numbers for this provider and probably more in the future. I also tried to fetch a minimal configuration from freeswitch source ( $somewhere/freeswitch.git/conf/minimal/ ) to replace the default configuration provided by the fusionpbx installer, then i just copied my original trunk configuration in /usr/local/freeswitch/conf/sip_profiles/external I did not set a dialplan for testing everything but it's seems to work better: when i call 0557590xxx i got the following message 2015-05-03 09:14:54.587286 [INFO] mod_dialplan_xml.c:635 Processing 0967212xxx <0967212xxx>->*0557590xxx* in context public In comparison with 2015-05-01 18:02:12.235827 [INFO] mod_dialplan_xml.c:635 Processing 0967212xxx <0967212xxx>->*0033972480xxx* in context public There is probably something bad in the default configuration provided by fusionpbx witch prevent auto_to_user to work properly. I will compare configuration files from minimal freeswitch sample configuration and from freeswitch and try to determine where is the problem. This may deserve a but report :-) Thanks On 02/05/2015 20:55, Sergey Safarov wrote:

In log you can see invite arrived from IP address 91.121.129.20 Change IP 10.7.1.60/32 <http://10.7.1.60/32> to 91.121.129.20/32 <http://91.121.129.20/32>

I'm not sure to understand why i should use a so strange trick. Using "internal" profile for external trunk can't be a security leak ? It is not problem. In this case profile used for authenticate call by user "provider_gw1". Real call processing executed in dialplan "provider_inbound_calls". In this dialplan you can allow one DID number and block all other.

If my provider change something about the 10.7.6.60 gateway, the trunk will no longer work until i update my configuration ? If provider can configure username and password for DID call authentication you can remove "cird" attribute. If username and password can't be configured, then say me how much DID number provider assigned for you.

Sergey

Hi Tanguy I has tested case where: 1) gateway linked to public context; 2) gateway has parameter "<param name="context" value="did_processing"/>"; 3) FS configuration has following dialplan. <context name="did_processing"> <extension name="info" continue="true"> <condition> <action application="info" /> </condition> </extension> <extension name="did_processing"> <condition field="${sip_to_user}" expression="^11_my_did_number_22$" break="on-true"> <action application="transfer" data="sales XML defaul" /> </condition> <condition field="${sip_to_user}" expression="^33_my_did_number_44$" break="on-true"> <action application="transfer" data="support XML defaul" /> </condition> <condition> <action application="hangup" data="INCOMING_CALL_BARRED"/> </condition> </extension> </context> It it working correctly. On Sun, May 3, 2015 at 7:25 PM, Tanguy <phenix@vfemail.net> wrote:

Hello

It solved my problem. I restored my yesterday backup and i notice that my inbound calls are now working. Very strange

I tried to remove the "extension" parameter, so inbound calls should not working but they are still working, and i noticed this message in the CLI when i updated my configuration using the fusionpbx GUI

2015-05-03 18:00:27.048335 [WARNING] sofia.c:3310 Ignoring duplicate gateway '0a96c3d3-0b0e-4864-b9ec-759fa4422429'

The gateway is not reloaded, all modifications where ignored but if i reload my external profile manually with *sofia profile external restart * i don't have this kind of message and my gateway is properly reloaded.

It seems to be a bug in fusionpbx, i will make a bug report

Regards

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