- FreeSWITCH-users - FreeSWITCH Mailing Lists

Goto Voicemail When Network Down
by Dan Lan 21 Sep '11

21 Sep '11

Hi, I install the Freeswitch without much modification. I have an extension, and with incoming DID transfer to that extension. Some Test Scenarios 1. When the extension is registered, the incoming call will go to VM after 30 secs, if no one answer the call (This is normal) 2. When the extension is not registered, the incoming call will go directly to VM 3. However, if the endpoint is registered, but for some reason, the endpoint's network is down. The incoming call will just go to blank scilience. after about 60 secs then I can hear the busy signal. >From the packet capture, I can see FS try to reach the end point by sending INVITE during the 60 secs, but because the end point is not responding, so the FS keep sending INVITE. I am just wondering, are there anyway to break the loop? see, if the FS send INVITE without response for some certain time, then go to Voice Mail without the caller waiting and dont know what to do? Dan

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Authenticate any user free and by demand?
by Dmitry Kravchenko 20 Sep '11

20 Sep '11

Is it possible to configure freeswitch to authenticate any SIP user connected to it, regardless it's password and presense in freeswitch configs, receiving it's name from SIP phone and allowing calls to that names?

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Making changes to fifo.conf.xml requires restarting FS?
by Yungwei Chen 20 Sep '11

20 Sep '11

Hi, I noticed that making changes to fifo.conf.xml requires FS to be restarted, and reloadxml command doesn't seem to reload fifo.conf.xml. Am I missing something? Thanks.

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Manipulate sip error codes
by Chris Martineau 20 Sep '11

20 Sep '11

Hi, Is it possible to translate B leg sip error codes so that the A leg always gives a fixed error code? i.e make call and get 408 for instance from the far end but would like 486 to be sent on the a leg. Need this as one of our suppliers retransmits a call multiple times unless they recieve a 403 or 486. No major issue but just clogs the error cdrs with loads of duplicate records otherwise. Any ideas would be greatly appreciated. Many thanks Chris

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Stuck registration.
by Igor Borisov 20 Sep '11

20 Sep '11

Hello! Trying to resolve problem with text messages passing between SIP users I've noticed strange registration: freeswitch@internal> sofia status profile internal Call-ID: ArOr7DiiGiLEUoA0..mgUbX30vJ7p3V2 User: 9001(a)static.XX.XX.XX.X.clients.your-server.de Contact: "Alexander" <sip:9001@YY.YY.YY.YY:52121;transport=tls> Agent: Bria iPhone 1.3.1 Status: Registered(TLS)(unknown) EXP(2011-08-22 18:07:11) EXPSECS(-2506676) Host: static.34.160.4.46.clients.your-server.de IP: YY.YY.YY.YY Port: 52121 Auth-User: 9001 Auth-Realm: static.XX.XX.XX.X.clients.your-server.de MWI-Account: 9001(a)static.XX.XX.XX.X.clients.your-server.de This registration is expired long time ago but still exists in the list. Registration from correct user with login 9001 exists too. I suppose it can be cause of messaging problems. Profile reload or FreeSwitch restart doesn't help - registration remains in the list. How can I get rid of it and avoid its appearance in the future? Regards, Igor A. Borisov -- С уважением, Борисов Игорь

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DTMF issue when using execute_extension with play_and_get_digits
by Nagalenoj H. 20 Sep '11

20 Sep '11

Hi Friends, Facing an issue when using bind_meta_app and execute_extension(with play_and_get_digits) combined. Here is my dialplan, <condition field="destination_number" expression="^4567$" break="on-true"> <action application="play_and_get_digits" data="1 5 3 7000 # '/voice_files/A.wav' '/voice_files/failed.wav' dtmf_received \d+"/> </condition> <condition field="destination_number" expression="^.*$" break="on-true"> <action application="answer"/> <action application="bind_meta_app" data="5 b o execute_extension::4567 XML default"/> <action application="bridge" data="freetdm/1/a/9952248266"/> <action application="park"/> </condition> So, when callee enters *5, I want the caller to enter a number. I get the extension executed as expected. The caller is able to hear the voice file played and when he enters the digits, it is not received. Digits are not even present in FS log. In the normal cases, there is no issues in getting DTMFs. I don't know, what am I doing wrong here. Kindly, help me to resolve this. -- Regards, Nagalenoj H.

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Sophia authentication without using editing acl.conf.xml
by Joe Flemmings 20 Sep '11

20 Sep '11

I use xml_curl to authenticate sip devices and was wondering if there is a way to do IP authentication without having to edit and reaload acl.conf.xml

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GXW4104 gateway setup for outgoing calls
by ocset 20 Sep '11

20 Sep '11

Hi I have recently bought a Grandstream GXW4104 (4 FXO ports) and need some help setting up a gateway to call out using the GXW4104. I am really out of my depth here and may be looking at this the wrong way so please bear with me. I followed the advice on this website "http://www.timhunt.net/wiki/FreeSwitch:GXW4104" and incoming calls from a PSTN line are working great. Now I need to setup a dialplan so that outgoing calls are routed through the same PSTN line on the GXW4104. I will eventually have 4 PSTN lines with a dialplan to use the first available line (if that is possible). According to the FreeSWITCH 1.0.6 book (and many online posts) I need to create a gateway and a dialplan but all the gateway examples are for SIP accounts. So, the gateway definition seems to need a username and password but the GXW4104 does not have that capability. I found this gateway definition in the freeswitch.xml.fsxml file but am not sure how many of these variables are required. <gateways> <gateway name="example.com"> <param name="username" value="joeuser"/> <param name="password" value="password"/> <param name="from-user" value="joeuser"/> <param name="from-domain" value="example.com"/> <param name="expire-seconds" value="600"/> <param name="register" value="false"/> <param name="retry-seconds" value="30"/> <param name="extension" value="5000"/> <param name="context" value="public"/> </gateway> </gateways> If I define a gateway called "gxw4104", then this is what I think a simple dialplan should look like but I'm not sure of the gateway details in the "bridge" section of the definition. <extension name="gxw4104_out"> <condition field="destination_number" expression="^(\d{10})$"> *<action application="bridge" data="sofia/gateway/gxw4104/........"/> (what should this be???)* </condition> </extension> Am I moving in the right direction and can someone fill in the blanks for me please Thanks in advance!

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[mod_xml_curl] Only accept if Return Password is blank
by rendyfrx 20 Sep '11

20 Sep '11

Hi, I'm trying to setup freeswitch to authenticate SIP client login (SIP register) to my backend database via php, example: username=65972932222, pwd=1234 As I couldn't really find sample return that suit my need, so I tried with below xml return. Problem is if the password is blank, then it will accepted, but if I change to simple password 1234 as xml return below, I will get SIP/2.0 403 Forbidden. xml return : <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory" description="User Directory"> <domain name="10.1.1.47"> <user id="6597293279"> <params> </params> </user> </domain> </section> I tried add in groups, but it doesn't have any impact. The bindings' auth-scheme value is "basic", mod_xml_curl will call another server (LAN) and it will validate and return the xml above. Anybody can advise what I might be missing here? Thanks in advance -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-xml-curl-Only-accept-if-R… Sent from the freeswitch-users mailing list archive at Nabble.com.

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uuid_broadcast not working
by Luis Jimenez 19 Sep '11

19 Sep '11

Hello, list I have this lua script, i try to put music to the aleg while do another task, but it is not working (not audio at all), any advice on what am doing wrong? THanks in advance, here my script: ---------------------------------------------------------------------------------------------- require 'socket' local host = "127.0.0.1" local port = 8021 function set_hold(uuid) local conn = socket.connect(host, port) if (conn) then local cmd = "bgapi uuid_broadcast " .. uuid .. " /opt/freeswitch/sounds/es/dm/dm-promo-recharge-hold.G729 aleg"; freeswitch.consoleLog("info", "CMD: " .. cmd .. "\n") conn:send(string.format("auth ClueCon\r\n\r\n%s\r\n\r\nexit\r\n\r\n", cmd)) line, err = conn:receive() while (not err) do if (line ~= nil) then freeswitch.consoleLog("info", "SCKT: " .. line .. "\n") end line, err = conn:receive() end conn:close() else debug.notice("Error posting to server: " .. host .. ":" .. port) end end function sleep(n) os.execute("sleep " .. tonumber(n)) end session:answer() session:sleep(500); uuid = session:get_uuid(); freeswitch.consoleLog("info", "UUID: " .. uuid .. "\n"); api = freeswitch.API(); set_hold(uuid); freeswitch.consoleLog("info", " - - - RETURNED - - -\n"); for i=1,50 do if not session:ready() then break end sleep(1); end if session:ready() then api:executeString("uuid_break" .. uuid); end freeswitch.consoleLog("info", " - - - FINNISH COUNTING - - -\n"); session:hangup(); ----------------------------------------------------------------------------------

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