July 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

api commands not working in outbound socket
by jaflong jaflong 24 Jul '15

24 Jul '15

Hi list I have a php script run through ivrd. api commands are returned with "ERR command not found" with done through sendRecv or api functions any suggestion why it fails? example 1 --------- code: $command = "api uuid_getvar db39795e-31e1-11e5-9af9-cfe09f60c9f7 variable_sip_from_user"; $conn->sendRecv($command); tcpdump: api uuid_getvar db39795e-31e1-11e5-9af9-cfe09f60c9f7 variable_sip_from_user Content-Type: command/reply Reply-Text: -ERR command not found example 2 --------- code: $command = "api version"; $conn->sendRecv($command); tcpdump: api version Content-Type: command/reply Reply-Text: -ERR command not found example 3 --------- code: $command = "version"; $conn->api($command); tcpdump: api version Content-Type: command/reply Reply-Text: -ERR command not found

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force-register-domain
by Nikolay Zaytsev 24 Jul '15

24 Jul '15

Hello,everyone. Ive tried to use a1-hash instead of username and password.(when i use plain username and password everything works fine). However, freeswitch does not allow users to register,when i use domain which is different from ip_address of the Freeswitch server. I am using domain - nick.sfedu. Domain in vars.xml,directory/default.conf is set as nick.sfedu and force-register-domain in internal profile is set to the same value. When I use plain username and password, i can see the following: Call-ID: NGJhMDU5NTM0MDllZWY2ODllZjFjYzMwYzAzMzQxMTc User: 11007(a)nick.sfedu Contact: "Nick" <sip:11007@10.132.144.157:49610 ;rinstance=fd2236fab37d0ba1;fs_nat=yes;fs_path=sip%3A11007%4010.132.144.157%3A49610%3Brinstance%3Dfd2236fab37d0ba1> Agent: X-Lite 4.7.1 74247-a844961e-W6.1 Status: Registered(UDP-NAT)(unknown) EXP(2015-07-24 14:55:08) EXPSECS(144) Host: fsw1-test IP: 10.132.144.157 Port: 49610 Auth-User: 11007 Auth-Realm: 10.160.1.23 MWI-Account: 11007(a)nick.sfedu where 10.160.1.23 is ip address of the Freeswitch server,but it should be nick.sfedu,as far as i understand. Best regards, Nikolay Zaytsev

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ReINVITE - 488 Not Acceptable Here
by Sergio García 24 Jul '15

24 Jul '15

Hello all, I am using *FreeSwitch* as a *WebRTC gateway* thanks to its Websocket support (in my case WSS), but I'm facing this strange problem. Audio and Video calls are working perfectly fine, but when I try to set a call On Hold, FreeSwitch replies with "488 Not Acceptable Here" error to the ReINVITE I'm sending. The only error I can see in the logs is: * [ERR] sofia.c:7280 Reinvite Codec Error!* The only difference between the original INVITE and this ReINVITE is that I try to set IP address to *0.0.0.0*, port to *0* and media attribute to *inactive*. I don't understand what part of the SDP, FS doesn't "like". Attached you can find a more detailed log file. Thank you very much in advance. Regards, -- *Sergio García Ramos * VoIP Engineer @ Quobis <http://www.quobis.com/> | e: sergio.garcia(a)quobis.com | t: +34902999465

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Record in WAV IEEE float format
by Zoltán Szabó 24 Jul '15

24 Jul '15

Hi, I would like to record in Wave IEEE float signed 32bit, 8000hz 256kbps, mono format. For this I can set the record_stereo to false in the session and tried to modify mod_sndfile.c to configure the format, so far I tried like: if (mode & SFM_WRITE) { context->sfinfo.channels = handle->channels; context->sfinfo.samplerate = handle->samplerate; context->sfinfo.format = SF_FORMAT_WAV | SF_FORMAT_FLOAT; } But I receive some weird data when trying to load up the wav in goldwave. The format looks fine but the data is not. Do you have any suggestion? Many thanks, Zoltan

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Strange mod_sofia logic
by Igor Olhovskiy 24 Jul '15

24 Jul '15

Hi! Found really strange behavior, but may be it's correct. I'm using FusionPBX as a FreeSwitch GUI, but problem more to FS. So, when I'm setting context of profile 'internal' to 'public' (yes, it's incorrect, but possible) this profile stop accept registrations. Answering "SIP/2.0 405 Method Not Allowed" and gives list of methods like INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,PRACK,NOTIFY,PUBLISH,SUBSCRIBE But when changing to 'default, all works correct and list of methods are INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE So the question is: how changing of context can inflict on methods supported? And how to control methods supported in more obvious way? (For ex, disable REFER) FreeSwitch version is 1.4.20 -- Best regards, Igor

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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

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(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

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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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