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Routing based on CID Name
by David S. Dee 24 Jul '15

24 Jul '15

Hello; I've been a freeswitch user for some time, but I am finally getting around to trying to get idea working again. I have an in-house CID lookup routine that works fine... it sets caller_id_name and effective_caller_id_name and those modified names show up fine on my telephones when the calls come in. That lookup routine adds a prefix of TM: if its a telemarketer, or VM: if i want the caller to go directly to voicemail... Basically, if the original CID from my provider shows up as "FOO Marketing", then after the cid-extension runs, the caller_id_name and effective_caller_name to "TM:FOO Marketing".... What is -not- working is the routing. I have an extension right after the CID Lookup that I want to do the routing directly to voicemail. Watching the debug logs, it appears that the field is already parsed and is set to the *original* callerid name that was available when the call came in... Trying different variations of field="caller_id_name" and field="${caller_id_name}" (and same with effective_caller_id_name) still yields no joy. Any guidance? The snippets from my dialplan are below. Thanks! <extension name="CID Inbound Lookup" continue="true"> <condition field="ani" expression="(\d{10})"> <action application="log" data="INFO In CID Inbound Lookup '$1' -- The Second Time"/> <action application="log" data="INFO CIDLookup(before): CID Num: '${caller_id_number}' CID Name: '${caller_id_name}' Orig CID Name: '${original_caller_id_name}' Dest: '${destination_number}'"/> <action application="set" data="url=num=${caller_id_number}&name=${url_- encode(${caller_id_name})}&did=${destination_number}"/> <action application="log" data="INFO CIDLookup(before): url: '${url}'"/> <action application="curl" data=" http://MYDOMAIN.com/MYCIDLOOKUP.php?${url}"/[1]> <action application="info"/> <action application="log" data="INFO CIDLookup(after ): curl_response_code: ${curl_response_code}"/> <action application="log" data="INFO CIDLookup(after ): curl_response_data: ${curl_response_data}"/> <action application="set" data="caller_id_name=${curl_response_data}"/> <action application="set" data="effective_caller_id_name=${curl_response_data}"/> <action application="log" data="INFO CIDLookup(after ): CID Num: '${caller_id_number}' CID Name: '${caller_id_name}' Orig CID Name: '${original_caller_id_name}' Dest: '${destination_number}'"/> <action application="log" data="INFO CIDLookup(after ): caller_id_name: '${caller_id_name}', original_caller_id_name: '${original_caller_id_name}', effective_caller_id_name: '${effective_caller_id_name}'" />  <action application="event" data="Event-Name=CUSTOM,Event- Subclass=CallAnnounce,ani=${destination_number},cidnum=${caller_id_numb- er},cidname=${caller_id_name}"/> </condition> </extension> and here are my test cases that are *right* after the above... but yet can't pick up the change in variables. *ONE* of these (I expect '3b' or '1') should fire... none do. <extension name="cid_test_0" continue="true"> <condition field="original_caller_id_name" expression="TM" > <action application="log" data="INFO cid_test_0: : '${original_caller_id_name}'"/> </condition> </extension> <extension name="cid_test_1" continue="true"> <condition field="caller_id_name" expression="TM" > <action application="log" data="INFO cid_test_1: : '${caller_id_name}'"/> </condition> </extension> <extension name="cid_test_2" continue="true"> <condition field="effective_caller_id_name" expression="TM" > <action application="log" data="INFO cid_test_2: : '${effective_caller_id_name}'"/> </condition> </extension> <extension name="cid_test_3" continue="true"> <condition field='${caller_id_name}' expression="TM" > <action application="log" data="INFO cid_test_3: : '${caller_id_name}'"/> </condition> </extension> <extension name="cid_test_3b" continue="true"> <condition field="${caller_id_name}" expression="TM" > <action application="log" data="INFO cid_test_3b: : '${caller_id_name}'"/> </condition> </extension> <extension name="cid_test_4" continue="true"> <condition field='${effective_caller_id_name}' expression="TM" > <action application="log" data="INFO cid_test_4: : '${effective_caller_id_name}'"/> </condition> </extension> <extension name="cid_test_5" continue="true"> <condition field='${effective_caller_id_name}' expression="TM" > <action application="log" data="INFO cid_test_5: : '${effective_caller_id_name}'"/> </condition> </extension> Links: 1. http://www.dsba.net/voip/cidlookup.php?${url}"/

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Build ERROR VS 2013
by Alexander Haugg 23 Jul '15

23 Jul '15

Hi, At the moment they'r bild errors for mod_conference, mod_shout, mod_rtmp and mod_verto. The same errors on "Relese" or "Debug" build for "win32" and "x64". And the "libspeex" project need the "/FS" compiler flag! Freeswitch version is from today: ? git rev-parse HEAD 17f8002936cb1797dc8bd46b4bad52800ebfb5cb ? git describe v1.5.final-1094-g17f8002 mod_conference: ============== Error 4 error C1083: Cannot open include file: 'mod_conference.h': No such file or directory D:\tmp\freeswitch\src\mod\applications\mod_conference\mod_conference.c 42 1 mod_conference mod_shout: ========== Error 4 error C1083: Cannot open include file: 'mpg123.h': No such file or directory D:\tmp\freeswitch\src\mod\formats\mod_shout\mod_shout.c 36 1 mod_shout mod_rtmp: ========= Error 4 error C1083: Cannot open include file: 'openssl/hmac.h': No such file or directory d:\tmp\freeswitch\src\mod\endpoints\mod_rtmp\handshake.h 30 1 mod_rtmp mod_verto: ========= Error 5 error C2079: '<unnamed-tag>::recv_addr6' uses undefined struct 'sockaddr_in6' d:\tmp\freeswitch\src\mod\endpoints\mod_verto\mcast\mcast.h 75 1 mod_verto Error 4 error C2079: '<unnamed-tag>::send_addr6' uses undefined struct 'sockaddr_in6' d:\tmp\freeswitch\src\mod\endpoints\mod_verto\mcast\mcast.h 74 1 mod_verto Thanks a lot!

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I have a php script run by ivrd, is it possible to subscribe to events on this outbound eslconnection
by jaflong jaflong 23 Jul '15

23 Jul '15

I have a php script run by ivrd, is it possible to subscribe to events on this outbound eslconnection

2 6

(no subject)
by Artur Mega 23 Jul '15

23 Jul '15

When new incoming call comes to tr2.xxxxxxx.ru, new session is being created and accounting begins. Further, call forwarding to another server, to fs2.xxxxxxx.ru, and new session is being created again. 2015-07-23 11:22:42.514095 [NOTICE] switch_channel.c:1054 *New Channel* sofia/external/73472460000@tr2.xxxxxxx.ru:5080 [ 26777290306690-93831426613199(a)192.168.217.156] ... 2015-07-23 11:22:42.574084 [INFO] mod_xml_radius.c:1123 mod_xml_radius: *Accounting Start success* 2015-07-23 11:22:42.574084 [DEBUG] switch_core_state_machine.c:164 sofia/external/73472460000@tr2.xxxxxxx.ru:5080 Standard ROUTING 2015-07-23 11:22:42.574084 [INFO] mod_dialplan_xml.c:558 Processing 73472460000 <73472460000>->79373057071 in context public Dialplan: sofia/external/73472460000@tr2.xxxxxxx.ru:5080 parsing [public->originate_leg] continue=true Dialplan: sofia/external/73472460000@tr2.xxxxxxx.ru:5080 Absolute Condition [originate_leg] Dialplan: sofia/external/73472460000@tr2.xxxxxxx.ru:5080 Action export(nolocal:h323-call-origin=originate) Dialplan: sofia/external/73472460000@tr2.xxxxxxx.ru:5080 parsing [public->public_mult] continue=false Dialplan: sofia/external/73472460000@tr2.xxxxxxx.ru:5080 Regex (PASS) [public_mult] destination_number(79373057071) =~ /^(79373057071)$/ break=on-false Dialplan: sofia/external/73472460000@tr2.xxxxxxx.ru:5080 Action set(hangup_after_bridge=true) Dialplan: sofia/external/73472460000@tr2.xxxxxxx.ru:5080 Action bridge(sofia/internal/79373057071@fs2.xxxxxxx.ru:5070) 2015-07-23 11:22:42.574084 [DEBUG] switch_core_state_machine.c:214 (sofia/external/73472460000@tr2.xxxxxxx.ru:5080) State Change CS_ROUTING -> CS_EXECUTE ... 2015-07-23 11:22:42.574084 [NOTICE] switch_channel.c:1054 *New Channel* sofia/internal/79373057071@fs2.xxxxxxx.ru:5070 [e059ad7f-b2f8-4beb-b1f3-249bc431cfc4] ... 2015-07-23 11:22:42.634059 [INFO] mod_xml_radius.c:1123 mod_xml_radius: * Accounting Start success* Thus, one call is being counted twice (for incoming to tr2, and then for outgoing from tr2 to fs2). But we need to make mod_xml_radius counts this call only once, how can we handle it? Thanks

1 0

BLF Implementation
by bhavik patel 23 Jul '15

23 Jul '15

Hi, I am trying to implement BLF in Grandstream (GXP1400) with freeswitch 1.4.20. For that , i used "<param name="send-presence-on-register" value="true"/>" in sip profile. But Not getting any NOTIFY request to sip phone,and BLF is not working. Can any one suggest me how to enable this function in Freeswitch ? Any help would be much appreciated. -- Thanks, Bhavik Patel

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Break unaswered bridge
by Alexey Melnichuk 23 Jul '15

23 Jul '15

This is simple dialplan that I try to implement <extension> <condition field="destination_number" expression="^\d+$" break="on-true"> <action application="bind_digit_action" data="conf,*1,exec:execute_extension,xfer"/> <action application="digit_action_set_realm" data="conf"/> <action application="conference" data="100@default"/> </condition> <condition field="destination_number" expression="^xfer$" break="on-true"> <action application="bind_digit_action" data="xfer,*1,?????????"/> <action application="digit_action_set_realm" data="xfer"/> <action application="digit_action_set_realm" data="xfer"/> <action application="read" data="2 11 'tone_stream://%(10000,0,350,440)' num 30000 #"/> <action application="bridge" data="user/${num}@${domain_name}"/> </condition> </extension> I have two problem with this dialplan 1. in secon `bind_digit_action` I can not use `exec` only `api`. e.g. <action application="bind_digit_action" data="xfer,*1,exec:info"/> does not work. application executes only when I return from first `exec:execute_extension` 2. I can not cancel bridge before answer I can not use transfer because I do not whant leave conference (e.g. user has `end_conference=true` flag). I try use uuid_kill from Lua like "xfer,*1,lua:kill_legs.lua ${uuid}" -- kill_legs.lua local uuid = argv[1] local sql = ("select uuid from channels where call_uuid='%s' and uuid<>'%s'"):format(uuid, uuid) dbh:query(sql, function(row) local res = api:executeString("uuid_kill " .. row.uuid) end) But this is does not work because it kills only active channels, but for dial-string like `loopback/111/my.domain, [leg_delay_start=20]loopback/222/my.domain` second channels creates only after 20 sec. And because I may have more then one channels I can not use `originate_uuid` In documentation I found `break` application but as I say I can not use it. `uuid_break` api break only play audio.

1 0

Websocket creation logging
by Oleg Stolyar 22 Jul '15

22 Jul '15

Hi guys, I am experiencing what looks like a memory leak on my FS servers but I suspect that there is a bug in my WebRTC client code that keep establishing new websockets and not closing them properly which causes FS to require more and more memory. Is there a way in FS to log creation and closing of websockets? Thank you Oleg

2 7

Enterprise originate not honouring fs_path in 302 redirect
by Asgaroth 22 Jul '15

22 Jul '15

Hi All, First of all, I'm a relatively new user of FreeSWITCH so, please bear with me if I'm asking a silly question. I have users registering through multiple kamailio loadbalancer/registrars, so when we build a bridge command to multiple local users we send the initial invite(s) to the kamailio lookup server which then responds with a 302 redirect message specifying the contact details of the user. What I am trying to do is get FreeSWITCH to call a particular user via a particular loadbalancer based on information recieved in the 302 redirect. From what I have read in the archives, we can achieve this with the fs_path directive on the bridge command. However, how/where do I specify the fs_path in the contact header in the 302 redirect in such a manner that FreeSWITCH will honour the fs_path directive. I originally tried setting a fs_path parameter in the contact of the 302 redirect message, something like the following: Contact: <sip:subscriber@1.2.3.4:5678;rinstance=9af71926212a25cf;transport=UDP;received=sip:5.6.7.8:9012;fs_path=sip:1.2.3.5> However, FreeSWITCH ignores the fs_path directive here and attempts to send this call directly to 1.2.3.4, instead of sending it via 1.2.3.5. Is this at all possible using the enterprise originate (:_: seperator) feature of the bridge command? We're attempting this with FreeSWITCH v 1.4.20 ("official" rpm packages from the repo) Any info/tips/suggestions/beatings much appreciate :) Thanks

4 6

Disable 401 and 407 for internal profile
by Admin 22 Jul '15

22 Jul '15

Hi, I want to configure FS to not challenge calls that come in from an internal profile (internal network devices), meaning disable SIP '401 Unauthorized' and '407 Proxy Authentication Required' for calls that belong to internal profile. I changed 'auth-calls' parameter to 'false', in internal.xml file but FS still sends 401 method when a user attempts to register and it sends 407 when an invite comes in. It seems the parameter is for ACL though. Is there another parameter to change in order to disable SIP 401 and 407 methods? Thank you.

6 13

Disable 401 and 407 for internal profile
by Admin 22 Jul '15

22 Jul '15

I have had the dialplan configured for users 1000-1999 ("destination_number" expression="^(1[0-9][0-9][0-9])$"> for months. I have run thousands of calls using the same dialplan and users config. files. To disable auth., i made two changes in internal.xml. I set 'auth-calls' value to 'false' and removed the line that contains 'apply-inbound-acl'. I didn't make any changes to neither dialplan nor the users config. files.

2 1