Hi all, I use freeswitch as voip server for other ipbx system. My freeswitch server is connected to gateway. Incoming calls works. I have extension, use as voip trunk for other server. Inbound rule redirect to this extension and it's works, but we don't received called number on extension (sip trunk). Caller ---- freeswitch ---- extension (other ipbx) On extension we never receive original caller number, and like that on this ipbx we can't forward inbound rules. Here my public.xml <extension name="xx3278" continue="true"> <condition field="context" expression="public" /> <condition field="destination_number" expression="^xxx278$" > <action application="set" data="dialed_ext=xxx3278"/> <action application="set" data="effective_callee_id_name=xxx3278" /> <action application="export" data="effective_callee_id_number=xxx3278" /> <action application="transfer" data="12345 XML default" /> </condition> </extension> On extension we only receive incomming call from extension 12345, which is my extension on freeswitch. Tks advance for your help
First, there is no need to call both "set" and "export" on the same variable, because "export" will set the local variable as well as exporting it to the other leg, unless you do something fancy in the "export". Second, we need to see the extension that is used for 12345 in default, because that may be also setting some variables. Third, I'm new to FreeSwitch, so listen to somebody more experienced first, should they also reply. :-) -- Steve On Tue, Mar 26, 2013 at 10:49 AM, Erwan Hermouet <ehermouet@bluetel.fr>wrote:
**
I use freeswitch as voip server for other ipbx system. My freeswitch server is connected to gateway. Incoming calls works. I have extension, use as voip trunk for other server. Inbound rule redirect to this extension and it’s works, but we don’t received called number on extension (sip trunk).* ***
** **
Caller ---- freeswitch ---- extension (other ipbx)****
On extension we never receive original caller number, and like that on this ipbx we can’t forward inbound rules.****
** **
Here my public.xml****
** **
<extension name="xx3278" continue="true">****
<condition field="context" expression="public" />****
<condition field="destination_number" expression="^xxx278$" >****
<action application="set" data="dialed_ext=xxx3278"/>****
<action application="set" data="effective_callee_id_name=xxx3278" /> ****
<action application="export" data="effective_callee_id_number=xxx3278" />****
<action application="transfer" data="12345 XML default" />****
</condition>****
</extension>****
** **
On extension we only receive incomming call from extension 12345, which is my extension on freeswitch.****
** **
Tks advance for your help****
**
Hi Tks for your reply Here my default <extension name="12345"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-% H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT, NO_ROUTE_DESTINATION"/> --> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numb er}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name } var sip_secure_media)}"/>--> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="{sip_invite_domain=$${domain}}user/${dialed_extension}@${domain_name}" /> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension> De : freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Steven Schoch Envoyé : mardi 26 mars 2013 20:59 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound First, there is no need to call both "set" and "export" on the same variable, because "export" will set the local variable as well as exporting it to the other leg, unless you do something fancy in the "export". Second, we need to see the extension that is used for 12345 in default, because that may be also setting some variables. Third, I'm new to FreeSwitch, so listen to somebody more experienced first, should they also reply. :-) -- Steve On Tue, Mar 26, 2013 at 10:49 AM, Erwan Hermouet <ehermouet@bluetel.fr> wrote: I use freeswitch as voip server for other ipbx system. My freeswitch server is connected to gateway. Incoming calls works. I have extension, use as voip trunk for other server. Inbound rule redirect to this extension and its works, but we dont received called number on extension (sip trunk). Caller ---- freeswitch ---- extension (other ipbx) On extension we never receive original caller number, and like that on this ipbx we cant forward inbound rules. Here my public.xml <extension name="xx3278" continue="true"> <condition field="context" expression="public" /> <condition field="destination_number" expression="^xxx278$" > <action application="set" data="dialed_ext=xxx3278"/> <action application="set" data="effective_callee_id_name=xxx3278" /> <action application="export" data="effective_callee_id_number=xxx3278" /> <action application="transfer" data="12345 XML default" /> </condition> </extension> On extension we only receive incomming call from extension 12345, which is my extension on freeswitch. Tks advance for your help
Extension 12345 doesn't match a "transfer 12345 XML default". It will transfer and look through the conditions. Make one of the following changes. Option 1: <condition field="destination_number" expression="^(10[01][0-9])$|^(12345)$"> Option 2: (put 1000 to 1019). <action application="transfer" data="1000 XML default" /> On Tue, Mar 26, 2013 at 1:47 PM, Erwan Hermouet <ehermouet@bluetel.fr>wrote:
Hi****
** **
Tks for your reply****
** **
Here my default****
** **
<extension name="12345">****
<condition field="destination_number" expression="^(10[01][0-9])$">* ***
<action application="set" data="dialed_extension=$1"/>****
<action application="export" data="dialed_extension=$1"/>****
<!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->****
<action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>****
<action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> ****
<action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>****
<action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/>****
<action application="set" data="ringback=${us-ring}"/>****
<action application="set" data="transfer_ringback=$${hold_music}"/> ****
<action application="set" data="call_timeout=30"/>****
<!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> -->****
<action application="set" data="hangup_after_bridge=true"/>****
<!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->****
<action application="set" data="continue_on_fail=true"/>****
<action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> ****
<action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>** **
<action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>****
<!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>-->****
<action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> ****
<action application="bridge" data="{sip_invite_domain=$${domain}}user/${dialed_extension}@ ${domain_name}"/>****
<action application="answer"/>****
<action application="sleep" data="1000"/>****
<action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>****
</condition>****
</extension>****
** **
*De :* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *De la part de* Steven Schoch *Envoyé :* mardi 26 mars 2013 20:59 *À :* FreeSWITCH Users Help *Objet :* Re: [Freeswitch-users] callee id inbound****
** **
First, there is no need to call both "set" and "export" on the same variable, because "export" will set the local variable as well as exporting it to the other leg, unless you do something fancy in the "export".****
** **
Second, we need to see the extension that is used for 12345 in default, because that may be also setting some variables.****
** **
Third, I'm new to FreeSwitch, so listen to somebody more experienced first, should they also reply. :-)****
** **
-- ****
Steve****
** **
On Tue, Mar 26, 2013 at 10:49 AM, Erwan Hermouet <ehermouet@bluetel.fr> wrote:****
** **
I use freeswitch as voip server for other ipbx system. My freeswitch server is connected to gateway. Incoming calls works. I have extension, use as voip trunk for other server. Inbound rule redirect to this extension and it’s works, but we don’t received called number on extension (sip trunk).* ***
****
Caller ---- freeswitch ---- extension (other ipbx)****
On extension we never receive original caller number, and like that on this ipbx we can’t forward inbound rules.****
****
Here my public.xml****
****
<extension name="xx3278" continue="true">****
<condition field="context" expression="public" />****
<condition field="destination_number" expression="^xxx278$" >****
<action application="set" data="dialed_ext=xxx3278"/>****
<action application="set" data="effective_callee_id_name=xxx3278" /> ****
<action application="export" data="effective_callee_id_number=xxx3278" />****
<action application="transfer" data="12345 XML default" />****
</condition>****
</extension>****
****
On extension we only receive incomming call from extension 12345, which is my extension on freeswitch.****
****
Tks advance for your help****
** **
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list 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
Tks for your reply I dont understand this . my extension number is 12345, and did xxx2378. Tks Option 2: (put 1000 to 1019). <action application="transfer" data="1000 XML default" /> De : freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Nick Vines Envoyé : mardi 26 mars 2013 22:00 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound Extension 12345 doesn't match a "transfer 12345 XML default". It will transfer and look through the conditions. Make one of the following changes. Option 1: <condition field="destination_number" expression="^(10[01][0-9])$|^(12345)$"> Option 2: (put 1000 to 1019). <action application="transfer" data="1000 XML default" /> On Tue, Mar 26, 2013 at 1:47 PM, Erwan Hermouet <ehermouet@bluetel.fr> wrote: Hi Tks for your reply Here my default <extension name="12345"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-% H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT, NO_ROUTE_DESTINATION"/> --> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numb er}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name } var sip_secure_media)}"/>--> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="{sip_invite_domain=$${domain}}user/${dialed_extension}@${domain_name}" /> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension> De : freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Steven Schoch Envoyé : mardi 26 mars 2013 20:59 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound First, there is no need to call both "set" and "export" on the same variable, because "export" will set the local variable as well as exporting it to the other leg, unless you do something fancy in the "export". Second, we need to see the extension that is used for 12345 in default, because that may be also setting some variables. Third, I'm new to FreeSwitch, so listen to somebody more experienced first, should they also reply. :-) -- Steve On Tue, Mar 26, 2013 at 10:49 AM, Erwan Hermouet <ehermouet@bluetel.fr> wrote: I use freeswitch as voip server for other ipbx system. My freeswitch server is connected to gateway. Incoming calls works. I have extension, use as voip trunk for other server. Inbound rule redirect to this extension and its works, but we dont received called number on extension (sip trunk). Caller ---- freeswitch ---- extension (other ipbx) On extension we never receive original caller number, and like that on this ipbx we cant forward inbound rules. Here my public.xml <extension name="xx3278" continue="true"> <condition field="context" expression="public" /> <condition field="destination_number" expression="^xxx278$" > <action application="set" data="dialed_ext=xxx3278"/> <action application="set" data="effective_callee_id_name=xxx3278" /> <action application="export" data="effective_callee_id_number=xxx3278" /> <action application="transfer" data="12345 XML default" /> </condition> </extension> On extension we only receive incomming call from extension 12345, which is my extension on freeswitch. Tks advance for your help _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list 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
Tks for your reply. I want to add that my extension name is 12345 and my extension number on freeswitch is 12345 too. So i try like this <extension name="12345"> <condition field="destination_number" expression="^(10[01][0-9])$|^(12345)$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-% H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT, NO_ROUTE_DESTINATION"/> --> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numb er}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name } var sip_secure_media)}"/>--> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="{sip_invite_domain=$${domain}}user/${dialed_extension}@${domain_name}" /> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension> So now log said my transfert is impossible because 2013-03-27 08:37:37.713058 [INFO] mod_dptools.c:2393 Originate Failed. Cause: SUBSCRIBER_ABSENT 2013-03-27 08:37:37.713058 [DEBUG] switch_ivr_originate.c:3440 Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT] 2013-03-27 08:37:37.713058 [ERR] switch_ivr_originate.c:2632 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] Tks advance for your help De : freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Nick Vines Envoyé : mardi 26 mars 2013 22:00 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound Extension 12345 doesn't match a "transfer 12345 XML default". It will transfer and look through the conditions. Make one of the following changes. Option 1: <condition field="destination_number" expression="^(10[01][0-9])$|^(12345)$"> Option 2: (put 1000 to 1019). <action application="transfer" data="1000 XML default" /> On Tue, Mar 26, 2013 at 1:47 PM, Erwan Hermouet <ehermouet@bluetel.fr> wrote: Hi Tks for your reply Here my default <extension name="12345"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-% H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT, NO_ROUTE_DESTINATION"/> --> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numb er}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name } var sip_secure_media)}"/>--> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="{sip_invite_domain=$${domain}}user/${dialed_extension}@${domain_name}" /> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension> De : freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Steven Schoch Envoyé : mardi 26 mars 2013 20:59 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound First, there is no need to call both "set" and "export" on the same variable, because "export" will set the local variable as well as exporting it to the other leg, unless you do something fancy in the "export". Second, we need to see the extension that is used for 12345 in default, because that may be also setting some variables. Third, I'm new to FreeSwitch, so listen to somebody more experienced first, should they also reply. :-) -- Steve On Tue, Mar 26, 2013 at 10:49 AM, Erwan Hermouet <ehermouet@bluetel.fr> wrote: I use freeswitch as voip server for other ipbx system. My freeswitch server is connected to gateway. Incoming calls works. I have extension, use as voip trunk for other server. Inbound rule redirect to this extension and its works, but we dont received called number on extension (sip trunk). Caller ---- freeswitch ---- extension (other ipbx) On extension we never receive original caller number, and like that on this ipbx we cant forward inbound rules. Here my public.xml <extension name="xx3278" continue="true"> <condition field="context" expression="public" /> <condition field="destination_number" expression="^xxx278$" > <action application="set" data="dialed_ext=xxx3278"/> <action application="set" data="effective_callee_id_name=xxx3278" /> <action application="export" data="effective_callee_id_number=xxx3278" /> <action application="transfer" data="12345 XML default" /> </condition> </extension> On extension we only receive incomming call from extension 12345, which is my extension on freeswitch. Tks advance for your help _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list 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
That is not your extension. What I mean is that you gave us a dialplan extension *named *12345, but the *name* of the extension is only meaningful when you're looking at a log file. Just because an extension is named 12345 does not mean its actions will get executed when 12345 is dialed. The important part of an extension is the condition. In this case the condition of that extension is: <condition field="destination_number" expression="^(10[01][0-9])$"> This means that condition will only be used if the number dialed (the number to which you transfer the call) is between 1000 and 1019. Since you transferred to 12345, the condition does not match, so that extension will be skipped, and processing will continue to other extensions. You have to find the extension that is actually getting processed. The easiest way is to look in the log file (or the CLI). You should see lines that start with "Dialplan:" and say PASS or FAIL. The last one that PASSed is probably the extension that ended up getting used. Once you find that, we can look at it to check the caller_id variables. -- Steve On Tue, Mar 26, 2013 at 1:47 PM, Erwan Hermouet <ehermouet@bluetel.fr>wrote:
Hi****
** **
Tks for your reply****
** **
Here my default****
** **
<extension name="12345">****
<condition field="destination_number" expression="^(10[01][0-9])$">* ***
<action application="set" data="dialed_extension=$1"/>****
<action application="export" data="dialed_extension=$1"/>****
<!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->****
<action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>****
<action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> ****
<action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>****
<action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/>****
<action application="set" data="ringback=${us-ring}"/>****
<action application="set" data="transfer_ringback=$${hold_music}"/> ****
<action application="set" data="call_timeout=30"/>****
<!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> -->****
<action application="set" data="hangup_after_bridge=true"/>****
<!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->****
<action application="set" data="continue_on_fail=true"/>****
<action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> ****
<action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>** **
<action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>****
<!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>-->****
<action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> ****
<action application="bridge" data="{sip_invite_domain=$${domain}}user/${dialed_extension}@ ${domain_name}"/>****
<action application="answer"/>****
<action application="sleep" data="1000"/>****
<action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>****
</condition>****
</extension>
Here the log file 2013-03-27 08:42:05.744114 [DEBUG] switch_core_state_machine.c:157 sofia/internal/06xxxxx@voip.vivaction.net Standard EXECUTE 2013-03-27 08:42:05.744114 [DEBUG] mod_sofia.c:239 sofia/internal/06xxxxx@voip.vivaction.net SOFIA EXECUTE 2013-03-27 08:42:05.744114 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/06xxxx@voip.xxxxx) State EXECUTE 2013-03-27 08:42:05.744114 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/06xxxx@voip.xxxxx) Running State Change CS_EXECUTE 2013-03-27 08:42:05.744114 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/06xxxx@voip.xxxxx) State ROUTING going to sleep 2013-03-27 08:42:05.744114 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/06xxxx@voip.xxxx[BREAK] 2013-03-27 08:42:05.744114 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/06xxxx@voip.ccccc) State Change CS_ROUTING -> CS_EXECUTE Dialplan: sofia/internal/06xxxx@voip.xxx Action transfer(12345 XML default) Dialplan: sofia/internal/06xxxx@voip.xxxx Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) Dialplan: sofia/internal/06xxxx@voip.xxxx Action sleep(1000) Dialplan: sofia/internal/06xxxx@voip.xxxx Action answer() Dialplan: sofia/internal/06xxxx@voip.xxxx Action bridge({sip_invite_domain=87.x.x.x}user/${dialed_extension}@${domain_name}) Dialplan: sofia/internal/06xxxx@voip.xxxx Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/06xxxx@voip.xxxx Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/06xxxx@voip.xxxx Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/06xxxx@voip.xxxx Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe r}) Dialplan: sofia/internal/06xxxx@voip.xxxx Action set(continue_on_fail=true) Dialplan: sofia/internal/06xxxx@voip.xxxx Action set(hangup_after_bridge=true) Dialplan: sofia/internal/06xxxx@voip.xxxx Action set(call_timeout=30) Dialplan: sofia/internal/06xxxx@voip.xxxx Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/06xxxx@voip.xxxx Action set(ringback=${us-ring}) Dialplan: sofia/internal/06xxxx@voip.xxxx Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: sofia/internal/06xxxx@voip.xxxx Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/06xxxx@voip.xxxx Action bind_meta_app(2 b s record_session::/opt/freeswitch/recordings/${caller_id_number}.${strftime(%Y -%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/internal/06xxxx@voip.xxxx Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/06xxxx@voip.xxxx Action export(dialed_extension=) Dialplan: sofia/internal/06xxxx@voip.xxxx Action set(dialed_extension=) Dialplan: sofia/internal/06xxxx@voip.xxxx Regex (PASS) [12345] destination_number(12345) =~ /^(10[01][0-9])$|^(12345)$/ break=on-false Dialplan: sofia/internal/06xxxx@voip.xxxx parsing [default->12345] continue=false Tks advance for your trying to help me J De : freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Steven Schoch Envoyé : mercredi 27 mars 2013 00:25 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound That is not your extension. What I mean is that you gave us a dialplan extension named 12345, but the name of the extension is only meaningful when you're looking at a log file. Just because an extension is named 12345 does not mean its actions will get executed when 12345 is dialed. The important part of an extension is the condition. In this case the condition of that extension is: <condition field="destination_number" expression="^(10[01][0-9])$"> This means that condition will only be used if the number dialed (the number to which you transfer the call) is between 1000 and 1019. Since you transferred to 12345, the condition does not match, so that extension will be skipped, and processing will continue to other extensions. You have to find the extension that is actually getting processed. The easiest way is to look in the log file (or the CLI). You should see lines that start with "Dialplan:" and say PASS or FAIL. The last one that PASSed is probably the extension that ended up getting used. Once you find that, we can look at it to check the caller_id variables. -- Steve On Tue, Mar 26, 2013 at 1:47 PM, Erwan Hermouet <ehermouet@bluetel.fr> wrote: Hi Tks for your reply Here my default <extension name="12345"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-% H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT, NO_ROUTE_DESTINATION"/> --> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numb er}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name } var sip_secure_media)}"/>--> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="{sip_invite_domain=$${domain}}user/${dialed_extension}@${domain_name}" /> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension>
OK I correct to this -> <extension name="12345"> <condition field="destination_number" expression="^(10[01][0-9])$|^(12345)$"> <action application="set" data="dialed_ext=$1"/> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <action application="set" data="voicemail_authorized=${sip_authorized}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default $${domain} ${dialed_ext}"/> <anti-action application="ring_ready"/> <anti-action application="set" data="call_timeout=10"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="bridge" data="USER/12345@$${domain}"/> <anti-action application="answer"/> <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension> Now i receive call on extension but without calle_id. Tks De : freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Erwan Hermouet Envoyé : mercredi 27 mars 2013 08:49 À : 'FreeSWITCH Users Help' Objet : Re: [Freeswitch-users] callee id inbound Here the log file 2013-03-27 08:42:05.744114 [DEBUG] switch_core_state_machine.c:157 sofia/internal/06xxxxx@voip.vivaction.net Standard EXECUTE 2013-03-27 08:42:05.744114 [DEBUG] mod_sofia.c:239 sofia/internal/06xxxxx@voip.vivaction.net SOFIA EXECUTE 2013-03-27 08:42:05.744114 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/06xxxx@voip.xxxxx) State EXECUTE 2013-03-27 08:42:05.744114 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/06xxxx@voip.xxxxx) Running State Change CS_EXECUTE 2013-03-27 08:42:05.744114 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/06xxxx@voip.xxxxx) State ROUTING going to sleep 2013-03-27 08:42:05.744114 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/06xxxx@voip.xxxx[BREAK <mailto:sofia/internal/06xxxx@voip.xxxx%5bBREAK> ] 2013-03-27 08:42:05.744114 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/06xxxx@voip.ccccc) State Change CS_ROUTING -> CS_EXECUTE Dialplan: sofia/internal/06xxxx@voip.xxx Action transfer(12345 XML default) Dialplan: sofia/internal/06xxxx@voip.xxxx Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) Dialplan: sofia/internal/06xxxx@voip.xxxx Action sleep(1000) Dialplan: sofia/internal/06xxxx@voip.xxxx Action answer() Dialplan: sofia/internal/06xxxx@voip.xxxx Action bridge({sip_invite_domain=87.x.x.x}user/${dialed_extension}@${domain_name} <mailto:%7bsip_invite_domain=87.x.x.x%7duser/$%7bdialed_extension%7d@$%7bdom ain_name%7d> ) Dialplan: sofia/internal/06xxxx@voip.xxxx Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/06xxxx@voip.xxxx Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/06xxxx@voip.xxxx Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/06xxxx@voip.xxxx Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe r}) Dialplan: sofia/internal/06xxxx@voip.xxxx Action set(continue_on_fail=true) Dialplan: sofia/internal/06xxxx@voip.xxxx Action set(hangup_after_bridge=true) Dialplan: sofia/internal/06xxxx@voip.xxxx Action set(call_timeout=30) Dialplan: sofia/internal/06xxxx@voip.xxxx Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/06xxxx@voip.xxxx Action set(ringback=${us-ring}) Dialplan: sofia/internal/06xxxx@voip.xxxx Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: sofia/internal/06xxxx@voip.xxxx Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/06xxxx@voip.xxxx Action bind_meta_app(2 b s record_session::/opt/freeswitch/recordings/${caller_id_number}.${strftime(%Y -%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/internal/06xxxx@voip.xxxx Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/06xxxx@voip.xxxx Action export(dialed_extension=) Dialplan: sofia/internal/06xxxx@voip.xxxx Action set(dialed_extension=) Dialplan: sofia/internal/06xxxx@voip.xxxx Regex (PASS) [12345] destination_number(12345) =~ /^(10[01][0-9])$|^(12345)$/ break=on-false Dialplan: sofia/internal/06xxxx@voip.xxxx parsing [default->12345] continue=false Tks advance for your trying to help me J De : freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Steven Schoch Envoyé : mercredi 27 mars 2013 00:25 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound That is not your extension. What I mean is that you gave us a dialplan extension named 12345, but the name of the extension is only meaningful when you're looking at a log file. Just because an extension is named 12345 does not mean its actions will get executed when 12345 is dialed. The important part of an extension is the condition. In this case the condition of that extension is: <condition field="destination_number" expression="^(10[01][0-9])$"> This means that condition will only be used if the number dialed (the number to which you transfer the call) is between 1000 and 1019. Since you transferred to 12345, the condition does not match, so that extension will be skipped, and processing will continue to other extensions. You have to find the extension that is actually getting processed. The easiest way is to look in the log file (or the CLI). You should see lines that start with "Dialplan:" and say PASS or FAIL. The last one that PASSed is probably the extension that ended up getting used. Once you find that, we can look at it to check the caller_id variables. -- Steve On Tue, Mar 26, 2013 at 1:47 PM, Erwan Hermouet <ehermouet@bluetel.fr> wrote: Hi Tks for your reply Here my default <extension name="12345"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-% H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT, NO_ROUTE_DESTINATION"/> --> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numb er}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name } var sip_secure_media)}"/>--> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="{sip_invite_domain=$${domain}}user/${dialed_extension}@${domain_name}" /> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension>
On Wed, Mar 27, 2013 at 1:31 AM, Erwan Hermouet <ehermouet@bluetel.fr>wrote:
<condition field="destination_number" expression="^(10[01][0-9])$|^(12345)$">
The problem with that condition is you have 2 expressions in parenthesis. Thus, if the destination number is from 1000 to 1019, it will set $1 to the number. However, if the destination number is 12345, then it will set $2 to that number, and leave $1 unset. You can fix that problem by changing it to this: <condition field="destination_number" expression="^(10[01][0-9]|12345)$">**** -- Steve
.Tks for your reply I do it, but i never see called number on sip tarce. Here the sip that i receive 3 Dev(3):[sip:12345@blue.fr:5060 / 12345]: PBX contact is public IP: <sip:12345@82.x.x.x:5060> 27-mars-2013 19:40:39.105 [CM503012]: Inbound office hours rule (unnamed) for 10000 forwards to DN:999 27-mars-2013 19:40:39.105 [Flow] Looking for inbound target: called=12345; caller="067xxxx" <sip:067xxxxx@:5060> 27-mars-2013 19:40:39.105 CallerNameAddr: "067xxxxx4"sip:067xxxxx;nf=e I must see called number before freeswitch Like here 7931 and not 12345 Max-Forwards: 70 Contact: <sip:025xxxxx@192.168.0.243:5060> To: <sip:7931@192.168.0.242> From: <sip:0253xxxxx@192.168.0.243:5060>;tag=486d6569e9 Call-ID: c62f94408c1f3c7c Tks advance for your help De : freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Steven Schoch Envoyé : mercredi 27 mars 2013 19:30 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound On Wed, Mar 27, 2013 at 1:31 AM, Erwan Hermouet <ehermouet@bluetel.fr> wrote: <condition field="destination_number" expression="^(10[01][0-9])$|^(12345)$"> The problem with that condition is you have 2 expressions in parenthesis. Thus, if the destination number is from 1000 to 1019, it will set $1 to the number. However, if the destination number is 12345, then it will set $2 to that number, and leave $1 unset. You can fix that problem by changing it to this: <condition field="destination_number" expression="^(10[01][0-9]|12345)$"> -- Steve
Hi all nobody know ? It's very urgent please :) tks for advance for your help Le 2013-03-27 19:44, Erwan Hermouet a écrit :
.Tks for your reply
I do it, but i never see called number on sip tarce.
Here the sip that i receive
3 Dev(3):[sip:12345@blue.fr:5060 / 12345]: PBX contact is public IP: <sip:12345@82.x.x.x:5060>
27-mars-2013 19:40:39.105 [CM503012]: Inbound office hours rule (unnamed) for 10000 forwards to DN:999
27-mars-2013 19:40:39.105 [Flow] Looking for inbound target: called=12345; caller="067xxxx" <sip:067xxxxx@:5060>
27-mars-2013 19:40:39.105 CallerNameAddr: "067xxxxx4"sip:067xxxxx;nf=e [1]
I must see called number before freeswitch
Like here 7931 and not 12345
Max-Forwards: 70
Contact: <sip:025xxxxx@192.168.0.243:5060>
To: <sip:7931@192.168.0.242>
From: <sip:0253xxxxx@192.168.0.243:5060>;tag=486d6569e9
Call-ID: c62f94408c1f3c7c
Tks advance for your help
DE : freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] DE LA PART DE Steven Schoch ENVOYÉ : mercredi 27 mars 2013 19:30 À : FreeSWITCH Users Help OBJET : Re: [Freeswitch-users] callee id inbound
On Wed, Mar 27, 2013 at 1:31 AM, Erwan Hermouet <ehermouet@bluetel.fr [2]> wrote:
<condition field="destination_number" expression="^(10[01][0-9])$|^(12345)$">
The problem with that condition is you have 2 expressions in parenthesis. Thus, if the destination number is from 1000 to 1019, it will set $1 to the number. However, if the destination number is 12345, then it will set $2 to that number, and leave $1 unset.
You can fix that problem by changing it to this:
<condition field="destination_number" expression="^(10[01][0-9]|12345)$">
On Thu, Mar 28, 2013 at 6:51 AM, <ehermouet@bluetel.fr> wrote:
nobody know ?
It's very urgent please :)
We need more information. Make the call. Then copy the freeswitch.log file (including all the lines from the time the call was first received until it is sent to the ipbx extension). Use http://pastebin.freeswitch.org/. Do not flood the mailing list. Then we can have a detailed look. -- Steve
Sorry i cant connect here Caller is xxxxx1544 anonymous Call to xxxxxx3278 Freeswitch server is 87.1.1.1 Extension server is 82.165.2.2 Extension on server fs is 12345 Tks advance 2013-03-28 18:21:11.236998 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/anonymous@anonymous.invalid) State ROUTING 2013-03-28 18:21:11.236998 [DEBUG] mod_sofia.c:146 sofia/internal/anonymous@anonymous.invalid SOFIA ROUTING 2013-03-28 18:21:11.236998 [DEBUG] switch_core_state_machine.c:77 sofia/internal/anonymous@anonymous.invalid Standard ROUTING 2013-03-28 18:21:11.236998 [INFO] mod_dialplan_xml.c:331 Processing Anonymous <xxxxxx1544>->12345 in context default Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->unloop] continue=false Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->tod_example] continue=true Dialplan: sofia/internal/anonymous@anonymous.invalid Date/Time Match (PASS) [tod_example] break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid Action set(open=true) Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->holiday_example] continue=true Dialplan: sofia/internal/anonymous@anonymous.invalid Date/Time Match (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->global-intercept] continue=false Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [global-intercept] destination_number(12345) =~ /^886$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->group-intercept] continue=false Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [group-intercept] destination_number(12345) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [intercept-ext] destination_number(12345) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->redial] continue=false Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [redial] destination_number(12345) =~ /^(redial|870)$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->global] continue=true Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/anonymous@anonymous.invalid Absolute Condition [global] Dialplan: sofia/internal/anonymous@anonymous.invalid Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/anonymous@anonymous.invalid Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_numbe r}) Dialplan: sofia/internal/anonymous@anonymous.invalid Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/anonymous@anonymous.invalid Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [snom-demo-2] destination_number(12345) =~ /^9001$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [snom-demo-1] destination_number(12345) =~ /^9000$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [eavesdrop] destination_number(12345) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [eavesdrop] destination_number(12345) =~ /^779$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->call_return] continue=false Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [call_return] destination_number(12345) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->del-group] continue=false Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [del-group] destination_number(12345) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->add-group] continue=false Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [add-group] destination_number(12345) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [call-group-simo] destination_number(12345) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->call-group-order] continue=false Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [call-group-order] destination_number(12345) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [extension-intercom] destination_number(12345) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [Local_Extension] destination_number(12345) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->3273] continue=false Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [3273] destination_number(12345) =~ /^(10[01][0-9]|3273)$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->3274] continue=false Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [3274] destination_number(12345) =~ /^(10[01][0-9]|3274)$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid parsing [default->12345] continue=false Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (PASS) [12345] destination_number(12345) =~ /^(10[01][0-9]|12345)$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid Action set(dialed_ext=12345) Dialplan: sofia/internal/anonymous@anonymous.invalid Regex (FAIL) [12345] destination_number(12345) =~ /^xxxxxx1544$/ break=on-false Dialplan: sofia/internal/anonymous@anonymous.invalid ANTI-Action ring_ready() Dialplan: sofia/internal/anonymous@anonymous.invalid ANTI-Action set(call_timeout=10) Dialplan: sofia/internal/anonymous@anonymous.invalid ANTI-Action set(hangup_after_bridge=true) Dialplan: sofia/internal/anonymous@anonymous.invalid ANTI-Action set(continue_on_fail=true) Dialplan: sofia/internal/anonymous@anonymous.invalid ANTI-Action bridge(USER/12345@87.1.1.1) Dialplan: sofia/internal/anonymous@anonymous.invalid ANTI-Action answer() Dialplan: sofia/internal/anonymous@anonymous.invalid ANTI-Action sleep(1000) Dialplan: sofia/internal/anonymous@anonymous.invalid ANTI-Action voicemail(default 87.1.1.1 ${dialed_ext}) 2013-03-28 18:21:11.244016 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/anonymous@anonymous.invalid) State Change CS_ROUTING -> CS_EXECUTE 2013-03-28 18:21:11.244016 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/anonymous@anonymous.invalid [BREAK] 2013-03-28 18:21:11.244016 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/anonymous@anonymous.invalid) State ROUTING going to sleep 2013-03-28 18:21:11.244016 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/anonymous@anonymous.invalid) Running State Change CS_EXECUTE 2013-03-28 18:21:11.244016 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/anonymous@anonymous.invalid) State EXECUTE 2013-03-28 18:21:11.244016 [DEBUG] mod_sofia.c:239 sofia/internal/anonymous@anonymous.invalid SOFIA EXECUTE 2013-03-28 18:21:11.244016 [DEBUG] switch_core_state_machine.c:157 sofia/internal/anonymous@anonymous.invalid Standard EXECUTE EXECUTE sofia/internal/anonymous@anonymous.invalid set(open=true) 2013-03-28 18:21:11.244016 [DEBUG] mod_dptools.c:854 sofia/internal/anonymous@anonymous.invalid SET [open]=[true] EXECUTE sofia/internal/anonymous@anonymous.invalid hash(insert/87.1.1.1-spymap/xxxxxx1544/ad47b79a-d938-4c72-8eb4-916c3180ce3b) EXECUTE sofia/internal/anonymous@anonymous.invalid hash(insert/87.1.1.1-last_dial/xxxxxx1544/12345) EXECUTE sofia/internal/anonymous@anonymous.invalid hash(insert/87.1.1.1-last_dial/global/ad47b79a-d938-4c72-8eb4-916c3180ce3b) EXECUTE sofia/internal/anonymous@anonymous.invalid set(RFC2822_DATE=Thu, 28 Mar 2013 18:21:11 +0100) 2013-03-28 18:21:11.247026 [DEBUG] mod_dptools.c:854 sofia/internal/anonymous@anonymous.invalid SET [RFC2822_DATE]=[Thu, 28 Mar 2013 18:21:11 +0100] EXECUTE sofia/internal/anonymous@anonymous.invalid set(dialed_ext=12345) 2013-03-28 18:21:11.247026 [DEBUG] mod_dptools.c:854 sofia/internal/anonymous@anonymous.invalid SET [dialed_ext]=[12345] EXECUTE sofia/internal/anonymous@anonymous.invalid ring_ready() 2013-03-28 18:21:11.248025 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/anonymous@anonymous.invalid! 2013-03-28 18:21:11.248025 [DEBUG] switch_core_session.c:666 Send signal sofia/internal/anonymous@anonymous.invalid [BREAK] 2013-03-28 18:21:11.248025 [NOTICE] mod_dptools.c:514 Ring Ready sofia/internal/anonymous@anonymous.invalid! EXECUTE sofia/internal/anonymous@anonymous.invalid set(call_timeout=10) 2013-03-28 18:21:11.249204 [DEBUG] sofia.c:4455 Channel sofia/internal/anonymous@anonymous.invalid entering state [early][180] 2013-03-28 18:21:11.249204 [DEBUG] mod_dptools.c:854 sofia/internal/anonymous@anonymous.invalid SET [call_timeout]=[10] EXECUTE sofia/internal/anonymous@anonymous.invalid set(hangup_after_bridge=true) 2013-03-28 18:21:11.249204 [DEBUG] mod_dptools.c:854 sofia/internal/anonymous@anonymous.invalid SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/anonymous@anonymous.invalid set(continue_on_fail=true) 2013-03-28 18:21:11.249204 [DEBUG] mod_dptools.c:854 sofia/internal/anonymous@anonymous.invalid SET [continue_on_fail]=[true] EXECUTE sofia/internal/anonymous@anonymous.invalid bridge(USER/12345@87.1.1.1) 2013-03-28 18:21:11.255009 [DEBUG] switch_ivr_originate.c:1979 variable string 0 = [sip_invite_domain=87.1.1.1] 2013-03-28 18:21:11.255009 [DEBUG] switch_ivr_originate.c:1979 variable string 1 = [presence_id=12345@87.1.1.1] 2013-03-28 18:21:11.256011 [NOTICE] switch_channel.c:779 New Channel sofia/internal/sip:12345@82.165.2.2:5060 [7a499a0c-774f-4774-96a5-189bc2616226] 2013-03-28 18:21:11.259002 [DEBUG] mod_sofia.c:3950 (sofia/internal/sip:12345@82.165.2.2:5060) State Change CS_NEW -> CS_INIT 2013-03-28 18:21:11.259002 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/sip:12345@82.165.2.2:5060 [BREAK] 2013-03-28 18:21:11.260016 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:12345@82.165.2.2:5060) Running State Change CS_INIT 2013-03-28 18:21:11.260016 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/sip:12345@82.165.2.2:5060) State INIT 2013-03-28 18:21:11.260016 [DEBUG] mod_sofia.c:83 sofia/internal/sip:12345@82.165.2.2:5060 SOFIA INIT 2013-03-28 18:21:11.856065 [DEBUG] sofia_glue.c:689 STUN Success [87.1.1.1]:[31540] 2013-03-28 18:21:11.856065 [DEBUG] sofia_glue.c:693 STUN Not Required ip and port match. [87.1.1.1]:[31540] 2013-03-28 18:21:11.856065 [DEBUG] mod_sofia.c:123 (sofia/internal/sip:12345@82.165.2.2:5060) State Change CS_INIT -> CS_ROUTING 2013-03-28 18:21:11.856065 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/sip:12345@82.165.2.2:5060 [BREAK] 2013-03-28 18:21:11.856065 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/sip:12345@82.165.2.2:5060) State INIT going to sleep 2013-03-28 18:21:11.856065 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:12345@82.165.2.2:5060) Running State Change CS_ROUTING 2013-03-28 18:21:11.856065 [DEBUG] switch_channel.c:1517 (sofia/internal/sip:12345@82.165.2.2:5060) Callstate Change DOWN -> RINGING 2013-03-28 18:21:11.857056 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/sip:12345@82.165.2.2:5060) State ROUTING 2013-03-28 18:21:11.857056 [DEBUG] mod_sofia.c:146 sofia/internal/sip:12345@82.165.2.2:5060 SOFIA ROUTING 2013-03-28 18:21:11.857056 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:12345@82.165.2.2:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2013-03-28 18:21:11.857056 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/sip:12345@82.165.2.2:5060 [BREAK] 2013-03-28 18:21:11.857056 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/sip:12345@82.165.2.2:5060) State ROUTING going to sleep 2013-03-28 18:21:11.857056 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:12345@82.165.2.2:5060) Running State Change CS_CONSUME_MEDIA 2013-03-28 18:21:11.857056 [DEBUG] switch_core_state_machine.c:360 (sofia/internal/sip:12345@82.165.2.2:5060) State CONSUME_MEDIA 2013-03-28 18:21:11.857056 [DEBUG] switch_core_state_machine.c:360 (sofia/internal/sip:12345@82.165.2.2:5060) State CONSUME_MEDIA going to sleep 2013-03-28 18:21:11.857056 [DEBUG] sofia.c:4455 Channel sofia/internal/sip:12345@82.165.2.2:5060 entering state [calling][0] 2013-03-28 18:21:12.205086 [INFO] sofia.c:709 sofia/internal/sip:12345@82.165.2.2:5060 Update Callee ID to "xxxxxx3278" <xxxxxx3278> 2013-03-28 18:21:12.209126 [DEBUG] sofia.c:4455 Channel sofia/internal/sip:12345@82.165.2.2:5060 entering state [proceeding][180] 2013-03-28 18:21:12.209126 [NOTICE] sofia.c:4527 Ring-Ready sofia/internal/sip:12345@82.165.2.2:5060! 2013-03-28 18:21:12.211091 [DEBUG] switch_core_session.c:666 Send signal sofia/internal/anonymous@anonymous.invalid [BREAK] 2013-03-28 18:21:12.211091 [NOTICE] switch_ivr_originate.c:472 Ring Ready sofia/internal/anonymous@anonymous.invalid! 2013-03-28 18:21:12.213103 [DEBUG] sofia.c:4455 Channel sofia/internal/sip:12345@82.165.2.2:5060 entering state [completing][200] 2013-03-28 18:21:12.213103 [DEBUG] sofia.c:4466 Remote SDP: v=0 o=3cxPS 239343763456 175808446465 IN IP4 82.165.2.2 s=3cxPS Audio call c=IN IP4 82.165.2.2 t=0 0 m=audio 5552 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 2013-03-28 18:21:12.213103 [DEBUG] sofia.c:4455 Channel sofia/internal/sip:12345@82.165.2.2:5060 entering state [ready][200] 2013-03-28 18:21:12.213103 [DEBUG] sofia_glue.c:3913 Audio Codec Compare [PCMU:0:8000:20]/[PCMA:8:8000:20] 2013-03-28 18:21:12.213103 [DEBUG] sofia_glue.c:3913 Audio Codec Compare [PCMU:0:8000:20]/[G7221:115:32000:20] 2013-03-28 18:21:12.213103 [DEBUG] sofia_glue.c:3913 Audio Codec Compare [PCMU:0:8000:20]/[G7221:107:16000:20] 2013-03-28 18:21:12.213103 [DEBUG] sofia_glue.c:3913 Audio Codec Compare [PCMU:0:8000:20]/[G722:9:8000:20] 2013-03-28 18:21:12.213103 [DEBUG] sofia_glue.c:3913 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2013-03-28 18:21:12.213103 [DEBUG] sofia_glue.c:2476 Set Codec sofia/internal/sip:12345@82.165.2.2:5060 PCMU/8000 20 ms 160 samples 2013-03-28 18:21:12.213103 [DEBUG] sofia_glue.c:4003 Set 2833 dtmf send payload to 101 2013-03-28 18:21:12.213103 [DEBUG] sofia_glue.c:2716 AUDIO RTP [sofia/internal/sip:12345@82.165.2.2:5060] 87.1.1.1 port 31540 -> 82.165.2.2 port 5552 codec: 0 ms: 20 2013-03-28 18:21:12.215102 [DEBUG] switch_rtp.c:1417 Starting timer [soft] 160 bytes per 20ms 2013-03-28 18:21:12.217095 [DEBUG] sofia_glue.c:2926 Set 2833 dtmf send payload to 101 2013-03-28 18:21:12.217095 [DEBUG] sofia_glue.c:2931 Set 2833 dtmf receive payload to 101 2013-03-28 18:21:12.217095 [DEBUG] switch_channel.c:2590 (sofia/internal/sip:12345@82.165.2.2:5060) Callstate Change RINGING -> ACTIVE 2013-03-28 18:21:12.217095 [DEBUG] switch_channel.c:2602 Send signal sofia/internal/anonymous@anonymous.invalid [BREAK] 2013-03-28 18:21:12.217095 [NOTICE] sofia.c:5026 Channel [sofia/internal/sip:12345@82.165.2.2:5060] has been answered 2013-03-28 18:21:12.337109 [DEBUG] sofia_glue.c:689 STUN Success [87.1.1.1]:[28386] 2013-03-28 18:21:12.337109 [DEBUG] sofia_glue.c:693 STUN Not Required ip and port match. [87.1.1.1]:[28386] 2013-03-28 18:21:12.337109 [DEBUG] sofia_glue.c:2716 AUDIO RTP [sofia/internal/anonymous@anonymous.invalid] 87.1.1.1 port 28386 -> 80.251.101.8 port 36584 codec: 8 ms: 20 2013-03-28 18:21:12.337109 [DEBUG] switch_rtp.c:1417 Starting timer [soft] 160 bytes per 20ms 2013-03-28 18:21:12.339105 [DEBUG] sofia_glue.c:2926 Set 2833 dtmf send payload to 101 2013-03-28 18:21:12.339105 [DEBUG] sofia_glue.c:2931 Set 2833 dtmf receive payload to 101 2013-03-28 18:21:12.339105 [DEBUG] mod_sofia.c:681 Local SDP sofia/internal/anonymous@anonymous.invalid: v=0 o=FreeSWITCH 1364462886 1364462887 IN IP4 87.1.1.1 s=FreeSWITCH c=IN IP4 87.1.1.1 t=0 0 m=audio 28386 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2013-03-28 18:21:12.339105 [DEBUG] switch_core_session.c:666 Send signal sofia/internal/anonymous@anonymous.invalid [BREAK] 2013-03-28 18:21:12.339105 [DEBUG] switch_channel.c:2590 (sofia/internal/anonymous@anonymous.invalid) Callstate Change RINGING -> ACTIVE 2013-03-28 18:21:12.341093 [NOTICE] switch_ivr_originate.c:3315 Channel [sofia/internal/anonymous@anonymous.invalid] has been answered 2013-03-28 18:21:12.341093 [DEBUG] switch_ivr_originate.c:3360 Originate Resulted in Success: [sofia/internal/sip:12345@82.165.2.2:5060] 2013-03-28 18:21:12.341093 [DEBUG] sofia.c:4455 Channel sofia/internal/anonymous@anonymous.invalid entering state [completed][200] 2013-03-28 18:21:12.343102 [DEBUG] switch_ivr_originate.c:3360 Originate Resulted in Success: [sofia/internal/sip:12345@82.165.2.2:5060] 2013-03-28 18:21:12.343102 [DEBUG] switch_core_session.c:666 Send signal sofia/internal/sip:12345@82.165.2.2:5060 [BREAK] 2013-03-28 18:21:12.343102 [DEBUG] switch_core_session.c:666 Send signal sofia/internal/anonymous@anonymous.invalid [BREAK] 2013-03-28 18:21:12.343102 [DEBUG] switch_ivr_bridge.c:1186 (sofia/internal/sip:12345@82.165.2.2:5060) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2013-03-28 18:21:12.343102 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/sip:12345@82.165.2.2:5060 [BREAK] 2013-03-28 18:21:12.343102 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:12345@82.165.2.2:5060) Running State Change CS_EXCHANGE_MEDIA 2013-03-28 18:21:12.343102 [DEBUG] switch_core_state_machine.c:351 (sofia/internal/sip:12345@82.165.2.2:5060) State EXCHANGE_MEDIA 2013-03-28 18:21:12.343102 [DEBUG] mod_sofia.c:552 SOFIA EXCHANGE_MEDIA 2013-03-28 18:21:12.357103 [DEBUG] sofia.c:4455 Channel sofia/internal/anonymous@anonymous.invalid entering state [ready][200] 2013-03-28 18:21:12.377096 [DEBUG] switch_rtp.c:2534 Correct ip/port confirmed. 2013-03-28 18:21:12.377096 [DEBUG] switch_core_session.c:728 Send signal sofia/internal/sip:12345@82.165.2.2:5060 [BREAK] 2013-03-28 18:21:12.377096 [DEBUG] switch_core_session.c:728 Send signal sofia/internal/anonymous@anonymous.invalid [BREAK] 2013-03-28 18:21:12.423088 [DEBUG] sofia.c:4455 Channel sofia/internal/anonymous@anonymous.invalid entering state [ready][200] 2013-03-28 18:21:12.423088 [DEBUG] sofia.c:4466 Remote SDP: v=0 o=cp10 136449127150 136449127151 IN IP4 192.168.59.1 s=SIP Call c=IN IP4 80.251.101.8 t=0 0 m=audio 36584 RTP/AVP 8 b=AS:64 a=rtpmap:8 PCMA/8000/1 a=ptime:20 De : freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Steven Schoch Envoyé : jeudi 28 mars 2013 17:43 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound On Thu, Mar 28, 2013 at 6:51 AM, <ehermouet@bluetel.fr> wrote: nobody know ? It's very urgent please :) We need more information. Make the call. Then copy the freeswitch.log file (including all the lines from the time the call was first received until it is sent to the ipbx extension). Use http://pastebin.freeswitch.org/. Do not flood the mailing list. Then we can have a detailed look. -- Steve
On Thu, Mar 28, 2013 at 10:33 AM, Erwan Hermouet <ehermouet@bluetel.fr>wrote:
Sorry i can’t connect here
What do you mean, you can't connect?
****
Caller is xxxxx1544 anonymous****
Call to xxxxxx3278
How do you call xxxxxx3278? Is it through the PSTN? If so, what gateway are you using and how is it routing the call to FS? How does xxxxxx3278 get mapped to 12345? (This is usually done in dialplan/public/*.xml.) You really need to learn how to use http://pastebin.freeswitch.org/. -- Steve
Sorry im new on freeswitch and i dont understand all things J Here the schema of my call My mobile phone number <anonymous> call xxxxxx3278-----à VOIP provider forward to my fs----> ip 87.1.1.1 is my freeswitch serveur - inbound rule redirect to extension 12345 ------à Other ipbx ip 82.165.2.2 connected to FS on extension 12345. Public xml <extension name="AB3278" continue="true"> <condition field="context" expression="public" /> <condition field="destination_number" expression="^xxxxxx3278$" > <action application="set" data="dialed_ext=xxxxxx3278"/> <action application="set" data="effective_callee_id_name=xxxxxx3278" /> <action application="export" data="effective_callee_id_number=xxxxxx3278" /> <action application="transfer" data="12345 XML default" /> </condition> </extension> <extension name="VIVA.10d" > //my voip provider <condition field="destination_number" expression="^(\d{10})$" > <action application="set" data="dialed_ext=$1"/> <action application="set" data="sip_h_X-accountcode=$${accountcode}" /> <action application="set" data="sip_h_X-Tag=" /> <action application="set" data="call_direction=outbound" /> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="effective_caller_id_name=$${outbound_caller_id_name}" /> <action application="set" data="effective_caller_id_number=$${outbound_caller_id_number}" /> <action application="set" data="inherit_codec=true" /> <action application="bridge" data="sofia/gateway/VIVA/$1" /> </condition> </extension> On my other ipbx i receive this To: <sip:12345@82.165.2.2:5060;rinstance=98f9fe5f934b9370>;tag=d41a4c19 But it must be like this To: <sip:xxxxxx2378@82.165.2.2:5060;rinstance=98f9fe5f934b9370>;tag=d41a4c19 I hope its clear J De : freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Steven Schoch Envoyé : jeudi 28 mars 2013 19:27 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound On Thu, Mar 28, 2013 at 10:33 AM, Erwan Hermouet <ehermouet@bluetel.fr> wrote: Sorry i cant connect here What do you mean, you can't connect? Caller is xxxxx1544 anonymous Call to xxxxxx3278 How do you call xxxxxx3278? Is it through the PSTN? If so, what gateway are you using and how is it routing the call to FS? How does xxxxxx3278 get mapped to 12345? (This is usually done in dialplan/public/*.xml.) You really need to learn how to use http://pastebin.freeswitch.org/. -- Steve
Here is how I would solve the problem: If you don't want the 'To' address to be 12345, then don't use that as your extension number. There is no rule that says your extension number has to be less than 10 digits. Make the extension of the ipbx xxxxxx3278. Then your action in the public dialplan would be: <action application="transfer" data="xxxxxx3278 XML default" />**** You would also add an extension in the default dialplan to bridge that number to user/xxxxxx3278. -- Steve
I can t do like this because 3278 is an exemple but i have 10did like that and redirect to 12345. It can be on other place that to...but today it is nowhere on other pbx Tks steve Steven Schoch <schoch+freeswitch.org@xwin32.com> a écrit :
Here is how I would solve the problem: If you don't want the 'To' address to be 12345, then don't use that as your extension number. There is no rule that says your extension number has to be less than 10 digits. Make the extension of the ipbx xxxxxx3278. Then your action in the public dialplan would be:
<action application="transfer" data="xxxxxx3278 XML default" />****
You would also add an extension in the default dialplan to bridge that number to user/xxxxxx3278.
--
Steve
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Hermouet Erwan Responsable technique Bluetel
I understand. I'm sure there must be a parameter you can put in the "user/12345@${domain} action, but I don't know what that is. Someone more knowledgeable will have to answer that. -- Steve
Yes it s that Steven Schoch <schoch+freeswitch.org@xwin32.com> a écrit :
I understand. I'm sure there must be a parameter you can put in the "user/12345@${domain} action, but I don't know what that is. Someone more knowledgeable will have to answer that.
-- Steve
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Hermouet Erwan Responsable technique Bluetel
Nobody know how do add my called number to my extension paramter ? maybe on application="bridge" data=" » I want to export my numer called to paramter like from, or contact or request uri tks De : freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Steven Schoch Envoyé : jeudi 28 mars 2013 21:18 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound I understand. I'm sure there must be a parameter you can put in the "user/12345@${domain} action, but I don't know what that is. Someone more knowledgeable will have to answer that. -- Steve
http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_number http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_name On Mar 28, 2013, at 6:22 PM, Erwan Hermouet <ehermouet@bluetel.fr> wrote:
Nobody know how do add my called number to my extension paramter ? maybe on application="bridge" data=" »
I want to export my numer called to paramter like from, or contact or request uri … tks
De : freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Steven Schoch Envoyé : jeudi 28 mars 2013 21:18 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound
I understand. I'm sure there must be a parameter you can put in the "user/12345@${domain} action, but I don't know what that is. Someone more knowledgeable will have to answer that.
:) tks i arrive to receive value but i want to add variable, because now it's only write value on defautl.xml but all time it will be called number, which is all time different. Le 2013-03-28 23:57, Michael Jerris a écrit :
http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_number [3] http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_name [4]
On Mar 28, 2013, at 6:22 PM, Erwan Hermouet <ehermouet@bluetel.fr [5]> wrote:
Nobody know how do add my called number to my extension paramter ? maybe on application="bridge" data=" »
I want to export my numer called to paramter like from, or contact or request uri … tks
DE : freeswitch-users-bounces@lists.freeswitch.org [1] [mailto:freeswitch-users-bounces@lists.freeswitch.org [2]] DE LA PART DE Steven Schoch ENVOYÉ : jeudi 28 mars 2013 21:18 À : FreeSWITCH Users Help OBJET : Re: [Freeswitch-users] callee id inbound
I understand. I'm sure there must be a parameter you can put in the "user/12345@${domain} action, but I don't know what that is. Someone more knowledgeable will have to answer that.
Links: ------ [1] mailto:freeswitch-users-bounces@lists.freeswitch.org [2] mailto:users-bounces@lists.freeswitch.org [3] http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_number [4] http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_name [5] mailto:ehermouet@bluetel.fr
I found <anti-action application="bridge" data="{origination_caller_id_name='${rdnis}',origination_caller_id_number='${rdnis}'}USER/11111@$${domain}"/> on extension tks all Le 2013-03-29 08:53, ehermouet@bluetel.fr a écrit :
:)
tks i arrive to receive value but i want to add variable, because now it's only write value on defautl.xml but all time it will be called number, which is all time different.
Le 2013-03-28 23:57, Michael Jerris a écrit :
http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_number [3] http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_name [4]
On Mar 28, 2013, at 6:22 PM, Erwan Hermouet <ehermouet@bluetel.fr [5]> wrote:
Nobody know how do add my called number to my extension paramter ? maybe on application="bridge" data=" »
I want to export my numer called to paramter like from, or contact or request uri … tks
DE : freeswitch-users-bounces@lists.freeswitch.org [1] [mailto:freeswitch-users-bounces@lists.freeswitch.org [2]] DE LA PART DE Steven Schoch ENVOYÉ : jeudi 28 mars 2013 21:18 À : FreeSWITCH Users Help OBJET : Re: [Freeswitch-users] callee id inbound
I understand. I'm sure there must be a parameter you can put in the "user/12345@${domain} action, but I don't know what that is. Someone more knowledgeable will have to answer that.
Links: ------ [1] mailto:freeswitch-users-bounces@lists.freeswitch.org [2] mailto:users-bounces@lists.freeswitch.org [3]
http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_number [4] http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_name [5] mailto:ehermouet@bluetel.fr
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list 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
participants (6)
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ehermouet＠bluetel.fr
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Erwan Hermouet
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Hermouet Erwan
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Michael Jerris
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Nick Vines
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Steven Schoch