cant access mailbox if userid is non numeric
under windows, i faced the following problem with FS. if i change the user_id to a non numeric value then the user can register to the FS but can access his mail box by calling 1005. as soon as i change the user_id to 1005 then dialing that i cana ccess my mailbox, y is this so? y cant the user have his name as user_id rather a number? <include> <user id="demo" mailbox="1005"> <params> </params> <variables> <variable name="accountcode" value="1005"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Ankur"/> <variable name="effective_caller_id_number" value="1005"/> </variables> </user> </include> -- View this message in context: http://www.nabble.com/cant-access-mailbox-if-userid-is-non-numeric-tp1972429... Sent from the Freeswitch-users mailing list archive at Nabble.com.
I believe the problem might lie in the dialplan that you are using to get the user into the mailbox. If you can resubmit this section from the user directory plus the snippet from the dialplan that pertains to this, that would make it easier to help you. Alternatively, you can stop by the #freeswitch channel on freenode and we can help in real time then. -- Yossi Neiman Cartis Solutions, Inc. http://www.cartissolutions.com xbipin wrote:
under windows, i faced the following problem with FS. if i change the user_id to a non numeric value then the user can register to the FS but can access his mail box by calling 1005. as soon as i change the user_id to 1005 then dialing that i cana ccess my mailbox, y is this so? y cant the user have his name as user_id rather a number?
<include> <user id="demo" mailbox="1005"> <params>
</params> <variables> <variable name="accountcode" value="1005"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Ankur"/> <variable name="effective_caller_id_number" value="1005"/> </variables> </user> </include>
-- Yossi Neiman President Cartis Solutions, Inc. P) 630-259-8100 http://www.cartissolutions.com
the dialplan is the standard config that comes with FS, no changes to it Yossi Neiman wrote:
I believe the problem might lie in the dialplan that you are using to get the user into the mailbox. If you can resubmit this section from the user directory plus the snippet from the dialplan that pertains to this, that would make it easier to help you. Alternatively, you can stop by the #freeswitch channel on freenode and we can help in real time then.
-- Yossi Neiman Cartis Solutions, Inc. http://www.cartissolutions.com
xbipin wrote:
under windows, i faced the following problem with FS. if i change the user_id to a non numeric value then the user can register to the FS but can access his mail box by calling 1005. as soon as i change the user_id to 1005 then dialing that i cana ccess my mailbox, y is this so? y cant the user have his name as user_id rather a number?
<include> <user id="demo" mailbox="1005"> <params>
</params> <variables> <variable name="accountcode" value="1005"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Ankur"/> <variable name="effective_caller_id_number" value="1005"/> </variables> </user> </include>
-- Yossi Neiman President Cartis Solutions, Inc. P) 630-259-8100 http://www.cartissolutions.com
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Hit the F8 key when you're on the FreeSWITCH console and try calling voicemail when you have the userid set as alphanumeric instead of numeric, and send me the log of the debug information. Again, this will be greatly easier if you join the IRC channel (#freeswitch on irc.freenode.net), as we can work with you in real time. -- Yossi Neiman Cartis Solutions, Inc. http://www.cartissolutions.com xbipin wrote:
the dialplan is the standard config that comes with FS, no changes to it
Yossi Neiman wrote:
I believe the problem might lie in the dialplan that you are using to get the user into the mailbox. If you can resubmit this section from the user directory plus the snippet from the dialplan that pertains to this, that would make it easier to help you. Alternatively, you can stop by the #freeswitch channel on freenode and we can help in real time then.
-- Yossi Neiman Cartis Solutions, Inc. http://www.cartissolutions.com
xbipin wrote:
under windows, i faced the following problem with FS. if i change the user_id to a non numeric value then the user can register to the FS but can access his mail box by calling 1005. as soon as i change the user_id to 1005 then dialing that i cana ccess my mailbox, y is this so? y cant the user have his name as user_id rather a number?
<include> <user id="demo" mailbox="1005"> <params>
</params> <variables> <variable name="accountcode" value="1005"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Ankur"/> <variable name="effective_caller_id_number" value="1005"/> </variables> </user> </include>
-- Yossi Neiman President Cartis Solutions, Inc. P) 630-259-8100 http://www.cartissolutions.com
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-- Yossi Neiman President Cartis Solutions, Inc. P) 630-259-8100 http://www.cartissolutions.com
You can also override this via the ${sip_mailbox} variable instead of the dialed_ext that is a variable that is magically set for you based on the mailbox = attr. /b On Sep 29, 2008, at 9:55 AM, xbipin wrote:
the dialplan is the standard config that comes with FS, no changes to it
Yossi Neiman wrote:
I believe the problem might lie in the dialplan that you are using to get the user into the mailbox. If you can resubmit this section from the user directory plus the snippet from the dialplan that pertains to this, that would make it easier to help you. Alternatively, you can stop by the #freeswitch channel on freenode and we can help in real time then.
-- Yossi Neiman Cartis Solutions, Inc. http://www.cartissolutions.com
xbipin wrote:
under windows, i faced the following problem with FS. if i change the user_id to a non numeric value then the user can register to the FS but can access his mail box by calling 1005. as soon as i change the user_id to 1005 then dialing that i cana ccess my mailbox, y is this so? y cant the user have his name as user_id rather a number?
<include> <user id="demo" mailbox="1005"> <params>
</params> <variables> <variable name="accountcode" value="1005"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Ankur"/> <variable name="effective_caller_id_number" value="1005"/> </variables> </user> </include>
-- Yossi Neiman President Cartis Solutions, Inc. P) 630-259-8100 http://www.cartissolutions.com
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what needs to be changed here, im a bit confused <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="dialed_ext=$1"/> <action application="export" 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 $${domain} ${dialed_ext}"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <anti-action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <anti-action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <anti-action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <anti-action application="set" data="transfer_ringback=${us-ring}"/> <anti-action application="set" data="call_timeout=30"/> <!-- <anti-action application="set" data="sip_exclude_contact=${network_addr}"/> --> <anti-action application="set" data="hangup_after_bridge=true"/> <!--<anti-action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> --> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/> <anti-action application="db" data="insert/last_dial_ext/${dialed_ext}/${uuid}"/> <anti-action application="bridge" data="user/${dialed_ext}@$${domain}"/> <anti-action application="answer"/> <!--<anti-action application="send_display" data="Voicemail for ${dialed_ext}"/>--> <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension> Brian West-3 wrote:
You can also override this via the ${sip_mailbox} variable instead of the dialed_ext that is a variable that is magically set for you based on the mailbox = attr.
/b
On Sep 29, 2008, at 9:55 AM, xbipin wrote:
the dialplan is the standard config that comes with FS, no changes to it
Yossi Neiman wrote:
I believe the problem might lie in the dialplan that you are using to get the user into the mailbox. If you can resubmit this section from the user directory plus the snippet from the dialplan that pertains to this, that would make it easier to help you. Alternatively, you can stop by the #freeswitch channel on freenode and we can help in real time then.
-- Yossi Neiman Cartis Solutions, Inc. http://www.cartissolutions.com
xbipin wrote:
under windows, i faced the following problem with FS. if i change the user_id to a non numeric value then the user can register to the FS but can access his mail box by calling 1005. as soon as i change the user_id to 1005 then dialing that i cana ccess my mailbox, y is this so? y cant the user have his name as user_id rather a number?
<include> <user id="demo" mailbox="1005"> <params>
</params> <variables> <variable name="accountcode" value="1005"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Ankur"/> <variable name="effective_caller_id_number" value="1005"/> </variables> </user> </include>
-- Yossi Neiman President Cartis Solutions, Inc. P) 630-259-8100 http://www.cartissolutions.com
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You have some options here. I highly recommend you check out the changes I have done to the default configs btw. <anti-action application="voicemail" data="default $${domain} $ {user_data(${dialed_ext}@$${domain} param real_mailbox)}"/> And just set the param of the real_mailbox for that user and it'll go snag it out of the directory for that user. Same for the check version: <action application="voicemail" data="check default $${domain} $ {user_data(${dialed_ext}@$${domain} param real_mailbox)}"/> Now if you really want to get crazy you can do something like this: <action application="set" data="real_mailbox=${user_data(${dialed_ext}@ $${domain} param real_mailbox)}"/> <action application="voicemail" data="default $${domain} ${cond($ {real_mailbox} == "" ? ${real_mailbox} : ${dialed_ext})}"/> That would use the real_mailbox param on the user or the dialed_ext... I didn't test that but it looks correct. Chances are I have done nothing but confuse you. :( /b On Sep 29, 2008, at 11:19 AM, xbipin wrote:
what needs to be changed here, im a bit confused
<extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9]) $"> <action application="set" data="dialed_ext=$1"/> <action application="export" 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 $${domain} ${dialed_ext}"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <anti-action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <anti-action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.$ {strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <anti-action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <anti-action application="set" data="transfer_ringback=${us-ring}"/> <anti-action application="set" data="call_timeout=30"/> <!-- <anti-action application="set" data="sip_exclude_contact=${network_addr}"/> --> <anti-action application="set" data="hangup_after_bridge=true"/> <!--<anti-action application="set" data = "continue_on_fail = NORMAL_TEMPORARY_FAILURE ,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> --> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/> <anti-action application="db" data="insert/last_dial_ext/${dialed_ext}/${uuid}"/> <anti-action application="bridge" data="user/${dialed_ext}@$$ {domain}"/> <anti-action application="answer"/> <!--<anti-action application="send_display" data="Voicemail for ${dialed_ext}"/>--> <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension>
Brian West-3 wrote:
You can also override this via the ${sip_mailbox} variable instead of the dialed_ext that is a variable that is magically set for you based on the mailbox = attr.
/b
On Sep 29, 2008, at 9:55 AM, xbipin wrote:
the dialplan is the standard config that comes with FS, no changes to it
Yossi Neiman wrote:
I believe the problem might lie in the dialplan that you are using to get the user into the mailbox. If you can resubmit this section from the user directory plus the snippet from the dialplan that pertains to this, that would make it easier to help you. Alternatively, you can stop by the #freeswitch channel on freenode and we can help in real time then.
-- Yossi Neiman Cartis Solutions, Inc. http://www.cartissolutions.com
xbipin wrote:
under windows, i faced the following problem with FS. if i change the user_id to a non numeric value then the user can register to the FS but can access his mail box by calling 1005. as soon as i change the user_id to 1005 then dialing that i cana ccess my mailbox, y is this so? y cant the user have his name as user_id rather a number?
<include> <user id="demo" mailbox="1005"> <params>
</params> <variables> <variable name="accountcode" value="1005"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Ankur"/> <variable name="effective_caller_id_number" value="1005"/> </variables> </user> </include>
-- Yossi Neiman President Cartis Solutions, Inc. P) 630-259-8100 http://www.cartissolutions.com
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i tried it but still doesnt work, the below is the log the suerid is bipin and the mailbox is 1001 2008-09-29 21:07:16 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel sofia/internal/bipin@sip.xbipin.com [52b5650d-3d2f-1f44-b36b-05bcd1f9c47 d] 2008-09-29 21:07:16 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing bip in->1001 in context default 2008-09-29 21:07:17 [INFO] switch_ivr_async.c:1481 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 1 execute_extension::dx XML features 2008-09-29 21:07:17 [INFO] switch_ivr_async.c:1481 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 2 record_session::C:Program Files (x86)FreeSWITCH/recordings /bipin.2008-09-29-21-07-17.wav 2008-09-29 21:07:17 [INFO] switch_ivr_async.c:1481 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 3 execute_extension::cf XML features 2008-09-29 21:07:18 [WARNING] mod_dptools.c:1972 user_outgoing_channel() can't f ind user [1001@sip.xbipin.com] 2008-09-29 21:07:18 [ERR] switch_ivr_originate.c:958 switch_ivr_originate() Cann ot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] 2008-09-29 21:07:18 [INFO] mod_dptools.c:1814 audio_bridge_function() Originate Failed. Cause: SUBSCRIBER_ABSENT 2008-09-29 21:07:18 [NOTICE] switch_channel.c:1426 switch_channel_perform_mark_p re_answered() Ring-Ready sofia/internal/bipin@sip.xbipin.com! 2008-09-29 21:07:18 [NOTICE] sofia_glue.c:2032 sofia_glue_tech_media() Pre-Answe r sofia/internal/bipin@sip.xbipin.com! 2008-09-29 21:07:18 [NOTICE] mod_dptools.c:606 answer_function() Channel [sofia/ internal/bipin@sip.xbipin.com] has been answered 2008-09-29 21:07:21 [NOTICE] sofia.c:2588 sofia_handle_sip_i_state() Hangup sofi a/internal/bipin@sip.xbipin.com [CS_EXECUTE] [NORMAL_CLEARING] 2008-09-29 21:07:21 [ERR] mod_sndfile.c:175 sndfile_file_open() Error Opening Fi le [C:\Program Files (x86)\FreeSWITCH\storage\voicemail\default\sip.xbipin.com\( null)\msg_52b5650d-3d2f-1f44-b36b-05bcd1f9c47d.wav] [System error : The system c annot find the file specified. ] 2008-09-29 21:07:21 [NOTICE] switch_core_session.c:812 switch_core_session_threa d() Session 3 (sofia/internal/bipin@sip.xbipin.com) Ended 2008-09-29 21:07:21 [NOTICE] switch_core_session.c:814 switch_core_session_threa d() Close Channel sofia/internal/bipin@sip.xbipin.com [CS_HANGUP] Brian West-3 wrote:
You have some options here. I highly recommend you check out the changes I have done to the default configs btw.
<anti-action application="voicemail" data="default $${domain} $ {user_data(${dialed_ext}@$${domain} param real_mailbox)}"/>
And just set the param of the real_mailbox for that user and it'll go snag it out of the directory for that user.
Same for the check version:
<action application="voicemail" data="check default $${domain} $ {user_data(${dialed_ext}@$${domain} param real_mailbox)}"/>
Now if you really want to get crazy you can do something like this:
<action application="set" data="real_mailbox=${user_data(${dialed_ext}@ $${domain} param real_mailbox)}"/> <action application="voicemail" data="default $${domain} ${cond($ {real_mailbox} == "" ? ${real_mailbox} : ${dialed_ext})}"/>
That would use the real_mailbox param on the user or the dialed_ext... I didn't test that but it looks correct.
Chances are I have done nothing but confuse you. :(
/b
On Sep 29, 2008, at 11:19 AM, xbipin wrote:
what needs to be changed here, im a bit confused
<extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9]) $"> <action application="set" data="dialed_ext=$1"/> <action application="export" 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 $${domain} ${dialed_ext}"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <anti-action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <anti-action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.$ {strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <anti-action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <anti-action application="set" data="transfer_ringback=${us-ring}"/> <anti-action application="set" data="call_timeout=30"/> <!-- <anti-action application="set" data="sip_exclude_contact=${network_addr}"/> --> <anti-action application="set" data="hangup_after_bridge=true"/> <!--<anti-action application="set" data = "continue_on_fail = NORMAL_TEMPORARY_FAILURE ,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> --> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/> <anti-action application="db" data="insert/last_dial_ext/${dialed_ext}/${uuid}"/> <anti-action application="bridge" data="user/${dialed_ext}@$$ {domain}"/> <anti-action application="answer"/> <!--<anti-action application="send_display" data="Voicemail for ${dialed_ext}"/>--> <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension>
Brian West-3 wrote:
You can also override this via the ${sip_mailbox} variable instead of the dialed_ext that is a variable that is magically set for you based on the mailbox = attr.
/b
On Sep 29, 2008, at 9:55 AM, xbipin wrote:
the dialplan is the standard config that comes with FS, no changes to it
Yossi Neiman wrote:
I believe the problem might lie in the dialplan that you are using to get the user into the mailbox. If you can resubmit this section from the user directory plus the snippet from the dialplan that pertains to this, that would make it easier to help you. Alternatively, you can stop by the #freeswitch channel on freenode and we can help in real time then.
-- Yossi Neiman Cartis Solutions, Inc. http://www.cartissolutions.com
xbipin wrote:
under windows, i faced the following problem with FS. if i change the user_id to a non numeric value then the user can register to the FS but can access his mail box by calling 1005. as soon as i change the user_id to 1005 then dialing that i cana ccess my mailbox, y is this so? y cant the user have his name as user_id rather a number?
<include> <user id="demo" mailbox="1005"> <params>
</params> <variables> <variable name="accountcode" value="1005"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Ankur"/> <variable name="effective_caller_id_number" value="1005"/> </variables> </user> </include>
-- Yossi Neiman President Cartis Solutions, Inc. P) 630-259-8100 http://www.cartissolutions.com
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im not using it by myself but actually my softphone calls my server whicha ctually has openSBC as my session border controller which is in upper registration mode so its proxys all the calls and then passes it over to FS so that might be doing it. from my dialer i simply call 1001. if i dont use opensbc then i cant use voip at all coz my isp blocks voip calls and opensbc does the needed encryption and decryption. is there anyway round it? Brian West-3 wrote:
Why are you calling with the @sign? REMOVE THAT.
/b
On Sep 29, 2008, at 12:09 PM, xbipin wrote:
Channel sofia/internal/bipin@sip.xbipin.com
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according to what discussion we had in irc, atleast with the default config and the changes u suggested, im supposed to atleast be able to hear my own messages from my mailbox but i cant, it gives me record message everytime so atleast this portion can be solved by checking caller id based on authorization and if its the same as dialed extension then go to own mailbox Brian West-3 wrote:
Why are you calling with the @sign? REMOVE THAT.
/b
On Sep 29, 2008, at 12:09 PM, xbipin wrote:
Channel sofia/internal/bipin@sip.xbipin.com
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Hi, As some of you are probably aware, I've had a really hard time getting asterisk to work with FS. The effective_caller_id_number and the accountcode were not getting logged or passed through on outgoing calls. I finally solved the problem, but attribute it to some unexpected behavior of the acl setting. I'm curious as to whether this is the intended behavior, or a bug. In my sip profile, I had apply-inbound-acl set to a list that contained the IP of my asterisk box. Apparently, when this happens, FS was allowing ANY call from my asterisk box without registration. Subsequently, since FS was just blindly accepting the call with no registration, it didn't match anything in the directory and no accountcode or caller-id was set. By simply removing the inbound-acl setting, I was able to have everything working perfectly. My understanding was that the goal of the inbound-acl was to limit which IP FS would accept SIP requests from. I was not aware that it would then accept any call blindly without any associated user. Interesting.... If any of the FS developers are interested in the exact configuration file that I'm using, please contact me directly and I'll forward the file to you. -Noah
The ACL is a way to specify a group of trusted machines and the system will bypass auth on those calls... If you need something from the directory don't use the ACL... If you don't want FS to respond to SIP from unknown IP Addresses that's a more appropriate job for your firewall software (iptables?)
From: Noah Silverman <noah@allresearch.com> Reply-To: <freeswitch-users@lists.freeswitch.org> Date: Tue, 30 Sep 2008 00:41:17 -0700 To: <freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Unexpected acl behavior. Feature or bug?
Hi,
As some of you are probably aware, I've had a really hard time getting asterisk to work with FS.
The effective_caller_id_number and the accountcode were not getting logged or passed through on outgoing calls.
I finally solved the problem, but attribute it to some unexpected behavior of the acl setting. I'm curious as to whether this is the intended behavior, or a bug.
In my sip profile, I had apply-inbound-acl set to a list that contained the IP of my asterisk box. Apparently, when this happens, FS was allowing ANY call from my asterisk box without registration. Subsequently, since FS was just blindly accepting the call with no registration, it didn't match anything in the directory and no accountcode or caller-id was set.
By simply removing the inbound-acl setting, I was able to have everything working perfectly.
My understanding was that the goal of the inbound-acl was to limit which IP FS would accept SIP requests from. I was not aware that it would then accept any call blindly without any associated user. Interesting....
If any of the FS developers are interested in the exact configuration file that I'm using, please contact me directly and I'll forward the file to you.
-Noah
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That makes sense. However it might make sense for me to add something to the wiki about this. It isn't documented anywhere that an "allow" in the acl will bypass the directory and registration. On a separate topic, I was just reading a post of yours from February where you describe your LCR platform. Would you be willing to share some of your setup and/or software for this? Thanks, -N On Sep 30, 2008, at 12:47 AM, Ken Rice wrote:
The ACL is a way to specify a group of trusted machines and the system will bypass auth on those calls...
If you need something from the directory don't use the ACL...
If you don't want FS to respond to SIP from unknown IP Addresses that's a more appropriate job for your firewall software (iptables?)
From: Noah Silverman <noah@allresearch.com> Reply-To: <freeswitch-users@lists.freeswitch.org> Date: Tue, 30 Sep 2008 00:41:17 -0700 To: <freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Unexpected acl behavior. Feature or bug?
Hi,
As some of you are probably aware, I've had a really hard time getting asterisk to work with FS.
The effective_caller_id_number and the accountcode were not getting logged or passed through on outgoing calls.
I finally solved the problem, but attribute it to some unexpected behavior of the acl setting. I'm curious as to whether this is the intended behavior, or a bug.
In my sip profile, I had apply-inbound-acl set to a list that contained the IP of my asterisk box. Apparently, when this happens, FS was allowing ANY call from my asterisk box without registration. Subsequently, since FS was just blindly accepting the call with no registration, it didn't match anything in the directory and no accountcode or caller-id was set.
By simply removing the inbound-acl setting, I was able to have everything working perfectly.
My understanding was that the goal of the inbound-acl was to limit which IP FS would accept SIP requests from. I was not aware that it would then accept any call blindly without any associated user. Interesting....
If any of the FS developers are interested in the exact configuration file that I'm using, please contact me directly and I'll forward the file to you.
-Noah
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The software in question is a custom dialplan replacement module for freeswitch that is available for licensing... As far as the setup goes you set up the sip profiles as normal, ratedecks and other items are loaded into a DB and routing happens automagically...
From: Noah Silverman <noah@allresearch.com> Reply-To: <freeswitch-users@lists.freeswitch.org> Date: Tue, 30 Sep 2008 01:12:00 -0700 To: <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Unexpected acl behavior. Feature or bug?
That makes sense.
However it might make sense for me to add something to the wiki about this. It isn't documented anywhere that an "allow" in the acl will bypass the directory and registration.
On a separate topic, I was just reading a post of yours from February where you describe your LCR platform. Would you be willing to share some of your setup and/or software for this?
Thanks,
-N
On Sep 30, 2008, at 12:47 AM, Ken Rice wrote:
The ACL is a way to specify a group of trusted machines and the system will bypass auth on those calls...
If you need something from the directory don't use the ACL...
If you don't want FS to respond to SIP from unknown IP Addresses that's a more appropriate job for your firewall software (iptables?)
From: Noah Silverman <noah@allresearch.com> Reply-To: <freeswitch-users@lists.freeswitch.org> Date: Tue, 30 Sep 2008 00:41:17 -0700 To: <freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Unexpected acl behavior. Feature or bug?
Hi,
As some of you are probably aware, I've had a really hard time getting asterisk to work with FS.
The effective_caller_id_number and the accountcode were not getting logged or passed through on outgoing calls.
I finally solved the problem, but attribute it to some unexpected behavior of the acl setting. I'm curious as to whether this is the intended behavior, or a bug.
In my sip profile, I had apply-inbound-acl set to a list that contained the IP of my asterisk box. Apparently, when this happens, FS was allowing ANY call from my asterisk box without registration. Subsequently, since FS was just blindly accepting the call with no registration, it didn't match anything in the directory and no accountcode or caller-id was set.
By simply removing the inbound-acl setting, I was able to have everything working perfectly.
My understanding was that the goal of the inbound-acl was to limit which IP FS would accept SIP requests from. I was not aware that it would then accept any call blindly without any associated user. Interesting....
If any of the FS developers are interested in the exact configuration file that I'm using, please contact me directly and I'll forward the file to you.
-Noah
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There is a separate apply-inbound-acl and apply-register-acl And yest the point of apply-inbound-acl is that all matches from that ip will be allowed in blindly. it's the opposite of doing Digest auth. There is still a way to associate an a ip range with a user so when you are let in over acl it will still set all the user settings. it's the special domain acl that was recently added that scans each user in your directory for a cidr= attr and adds it to the domains acl then you can apply-inbound-acl=domains It is possible to get asterisk to register to FS with normal digest auth it's just a little tricky because asterisk does not like domain based authentication which is actually the normal way to do it. They hacked in support for it way late in the development timeline and it's kinda hard to figure out, I still forget how to do it sometimes. it's something to do with a special syntax in the register line. On Tue, Sep 30, 2008 at 2:41 AM, Noah Silverman <noah@allresearch.com>wrote:
Hi,
As some of you are probably aware, I've had a really hard time getting asterisk to work with FS.
The effective_caller_id_number and the accountcode were not getting logged or passed through on outgoing calls.
I finally solved the problem, but attribute it to some unexpected behavior of the acl setting. I'm curious as to whether this is the intended behavior, or a bug.
In my sip profile, I had apply-inbound-acl set to a list that contained the IP of my asterisk box. Apparently, when this happens, FS was allowing ANY call from my asterisk box without registration. Subsequently, since FS was just blindly accepting the call with no registration, it didn't match anything in the directory and no accountcode or caller-id was set.
By simply removing the inbound-acl setting, I was able to have everything working perfectly.
My understanding was that the goal of the inbound-acl was to limit which IP FS would accept SIP requests from. I was not aware that it would then accept any call blindly without any associated user. Interesting....
If any of the FS developers are interested in the exact configuration file that I'm using, please contact me directly and I'll forward the file to you.
-Noah
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
participants (6)
-
Anthony Minessale
-
Brian West
-
Ken Rice
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Noah Silverman
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xbipin
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Yossi Neiman