January 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] feature list
by Bob Smith 19 Jan '12

19 Jan '12

How about someone puts a link to the changelog at the top of that page ? ;-) >we should probably update that =p >there is also the changelog file in the build root which is pretty big but >sort of up to date.

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Loosing control when bridging calls using Lua
by Michael Lutz 19 Jan '12

19 Jan '12

Hi All, I am struggling with some code, I have (based on the wiki examples) the following piece of lua code: session:setHangupHook("HangUpDialAbort") legA = session legB = freeswitch.Session("sofia/external/{MYDESTINATION}@{MYEXTERNALGATEWAY}") dispoB = "None" while(legA:ready() and legB:ready() and dispoB ~= "ANSWER") do if ( not legA:ready() ) then freeswitch.consoleLog("NOTICE","It appears that leg_a disconnected...\n") else dispoB = legB:getVariable("endpoint_disposition") end end if ( legA:ready() and legB:ready() ) then freeswitch.bridge(legA,legB) while(legA:ready() and legB:ready()) do freeswitch.consoleLog("NOTICE","We´re in the loop!\n") end else freeswitch.consoleLog("NOTICE","It appears that " .. dialA .. " or " .. dialB .. " disconnected...\n") end Question 1: It seems freeswitch.bridge() is a blocking function, Is there a way to use it in a non-blocking way? (so it will reach my while loop checking for legA and B to be ready) I need control of my code while the two legs are bridged. Question 2: session:setHangupHook("HangUpDialAbort") How can I pass session object to my hangup hook? I need to abort the dial attempt in my hangup hook when legA disconnects. Or are there other (better) ways of doing this? Question3: Is it possible to set different hangup hooks for different session (legs)? (would need that when question 2 leads to a 'no-go' ...) Thanks for your help, Michael Lutz

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PHP ESL Custom Events Event-Name sent twice
by Gerald Weber 19 Jan '12

19 Jan '12

Hello, i'm trying to send custom events using PHP ESL Library. (running FreeSWITCH Version 1.0.head (git-c37c029 2012-01-11 21-35-19 -0600)) My PHP Code: <?php require_once('ESL.php'); $sock = new ESLconnection('192.168.20.73', '8021', 'ClueCon'); $e = new ESLevent("CUSTOM","CONFIG::AGENT_LIST"); $e->addHeader("Agents","2022"); $sock->sendEvent($e); ?> When I connect to fs using telnet and run the php script from another xterm, I can see the following output: Content-Length: 657 Content-Type: text/event-json { "Event-Subclass": "CONFIG::AGENT_LIST", "Event-Name": "SOCKET_DATA", "Core-UUID": "b3ac8858-4019-11e1-910a-9b04baf8e5ea", "FreeSWITCH-Hostname": "freeswitch.local", "FreeSWITCH-Switchname": "freeswitch.local", "FreeSWITCH-IPv4": "192.168.20.73", "FreeSWITCH-IPv6": "::1", "Event-Date-Local": "2012-01-16 15:20:23", "Event-Date-GMT": "Mon, 16 Jan 2012 14:20:23 GMT", "Event-Date-Timestamp": "1326723623776184", "Event-Calling-File": "mod_event_socket.c", "Event-Calling-Function": "read_packet", "Event-Calling-Line-Number": "1188", "Command": "sendevent CUSTOM", "Event-Name": "CUSTOM", "Agents": "2022", "ZMQ-Msg-Cnt": "3508" } Looks ok, but i don't get why "Event-Name" is sent twice. Shouldn't there be an Event-Name: CUSTOM only ? Is there a way to avoid this or am i missing something ? thx & regards, gw

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bridging calls using mod_managed
by Michael Lutz 19 Jan '12

19 Jan '12

Hi, Is there someone who can explain me why after executing a bridge function in mod.managed, the hangup hook is executed immediately, and my main code execution continues. And, even stranger, Context.session.bridged() returns false for both legs! This code snippet is the example, it skips the while loop (check for bridge) but does detect the answered(), even session.ready() returns false... FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api(); ManagedSession leg_b = new ManagedSession("{ignore_early_media=true,origination_caller_id_number=" + orig + ",originate_timeout=20}sofia/external/" + route); if (leg_b.Ready()) { if (leg_b.answered()) { leg_a.answer(); string apiResult = fsApi.ExecuteString(string.Format("uuid_bridge {1} {0}", leg_a.GetUuid(), leg_b.GetUuid())); while (leg_a.bridged() && leg_b.bridged()) { leg_a.sleep(500, 0); // Slow the loop down... } } } Thanks, Michael Lutz

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Issues with Linksys ATA and DTMF
by obbyone 19 Jan '12

19 Jan '12

Hi, On my freeswitch server, I have created a number 9910 that has to recieve messages in DTMF. As I call that number with a softphone (xlite) on a PC, everything works, but as I try to call and send DTMF from an analog phone plugged into an ATA linksys spa 3000, freeswitch answers, but no DTMF is recieved. The linksys is configured to send DTMF in inband mode. Do you have any idea how I can handle this issue ? Thanks, Tony -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Issues-with-Linksys-ATA-and-D… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Freeswitch & double NAT Configuration (Was: USER_NOT_REGISTERED with external profile)
by Alexis de BRUYN [Mailinglists] 19 Jan '12

19 Jan '12

Hi Everybody, I am trying to get a conversation between two natted clients and a natted freeswitch server. I am playing with a fresh install, the external profile and default directory (1000 & 1001 users). I can see that my clients are registered on the server logs (on the 5080 port). All others needed ports are opened too. I have followed the instructions from the wiki to get a double nat configuration, but my client phone are not ringing (and I cannot hear anything). All is working fine with defaults in a non nat configuration (with internal profile). Is there anything special to set up in the directory / user configuration ? Can anyone give me some hints with his natted configuration ? I am stuck. Thanks for your help. Regards, -------- Original Message -------- Subject: [Freeswitch-users] USER_NOT_REGISTERED with external profile Date: Mon, 16 Jan 2012 23:37:20 +0100 From: Alexis de BRUYN [Mailinglists] <alexis.mailinglist(a)de-bruyn.fr> Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Hi Everybody, I am trying to use FreeSwitch in a (double) NAT Configuration from a fresh snapshot install (Debian Squeeze, server outside from my client LAN) with default directory for users. When I call 1001 from 1000 (or 1000 from 1001) , I cannot hear the callee, the phone doesn't ring and automatically hangup, I see in the console : 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] However, 1000 and 1001 are registered (from the same LAN) : sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 192.168.1.6 SIP-IP 192.168.1.6 URL sip:mod_sofia@192.168.1.6:5080 BIND-URL sip:mod_sofia@192.168.1.6:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM CODECS OUT PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 3 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: aU.1ZlEZkW7nlHpwp9ig1lt7A3weNnwm User: 1001(a)X.Y.Z.T Contact: "Freeswitch" <sip:1001@A.B.C.D:50193> Agent: Bria iOS 2.0.0 Status: Registered(UDP)(unknown) EXP(2012-01-16 23:16:00) EXPSECS(934) Host: phone IP: A.B.C.D Port: 50193 Auth-User: 1001 Auth-Realm: X.Y.Z.T MWI-Account: 1001(a)X.Y.Z.T Call-ID: 3c27b7424837-bsfi877ujb2n User: 1000(a)X.Y.Z.T Contact: "freeswitch" <sip:1000@192.168.0.69:2048;line=skgk4hug> Agent: snom300/8.4.32 Status: Registered(UDP)(unknown) EXP(2012-01-17 00:01:24) EXPSECS(3658) Host: phone IP: A.B.C.D Port: 62061 Auth-User: 1000 Auth-Realm: X.Y.Z.T MWI-Account: 1000(a)X.Y.Z.T Total items returned: 2 ================================================================================================= All necessary ports are opened/forwarded on the server. I See on the 1000 configuration that this is the local ip address which is set as contact. Is there any other setups to do in the directory ? Or other parameters in the external profile ? Thanks for your help ! Regards, -- Alexis de BRUYN Mail : alexis.mailinglist(a)de-bruyn.fr _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)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(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -- Alexis de BRUYN Mail : alexis.mailinglist(a)de-bruyn.fr

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feature list
by bob smith 19 Jan '12

19 Jan '12

Hi everyone. Does anyone have a feature list for freeswitch? I have not been able to find a good one anywhere ! Pls let me know ! B

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Unable to export a variable to b leg (callcenter)
by Yungwei Chen 19 Jan '12

19 Jan '12

Hi, With the extension below, I am able to export caller_id_name to b leg without problems. However, exporting another variable doesn't seem to work because I don't see it in the corresponding CDRs (cdr.variables). It looks like only known variables are exportable. Is there way to export a custom variable in this case? Thanks. <extension name="test"> <condition field="destination_number" expression="^test1$" break="on-true"> <action application="export" data="sid=${sid}" />  <action application="export" data="caller_id_name=${origination_caller_id_name}" />  <action application="answer" data=""/> <action application="callcenter" data="support@default"/> </condition> </extension>

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choppy sound on voice recording and conference
by Anestis Mavro 18 Jan '12

18 Jan '12

Hello, I have a problem with one of my sip trunks. The provider uses g729 with a ptime of 40ms and all incoming calls that are being recorded or go to the conference have a big delay and choppy sound, like a mismatch of ptime. I have already tried to put rtp-autofix-timing=false into the profiles and even changed the inbound-codec.. to g729@40i, but nothing helps. We have done a test with the provider switching for a test call to 20ms and the sound was perfect. The issue now is that the provider can't keep this setting only for us, he wants to send with 40ms. Is there a way to configure it on Freeswitch? Thank you Anestis

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mod_sofia displaying on Avaya 9630 as caller id contact
by Bernard Murphy 18 Jan '12

18 Jan '12

Hi, I have a soft phone calling an Avaya 9630 both registered on the same freeswitch and have a problem with the Avaya displaying the inbound contact as 'mod_sofia' instead of the originating extension number. The invite from the originating extension and the invite to the Avaya are both pasted below and you will see theat freeswitch includes mod_sofia as the contact uri. Can this be corrected in anyway so that the Avaya phone displays the incoming extension number ? thanks murph* ORIGINAL INVITE* INVITE sip:1000@192.168.2.42 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31;rport;branch=z9hG4bKc0a8021f000000234f166fd50000386400000044 Content-Length: 337 Contact: <sip:1001@192.168.2.31:5060> Call-ID: BC96029E-D214-4CBF-BCF5-BD0A260757C1(a)192.168.2.31 Content-Type: application/sdp CSeq: 2 INVITE From: "unknown"<sip:1001@192.168.2.42>;tag=1383828710312 Max-Forwards: 70 To: <sip:1000@192.168.2.42> User-Agent: SJphone/1.60.289a (SJ Labs) Proxy-Authorization: Digest username="1001",realm="192.168.2.42",nonce="6ff3bebc-b5c5-446a-8625-e89dcd2dd861",uri=" sip:1000@192.168.2.42 ",response="1754cba9b8bc72cfac8c69ff0bee8a21",algorithm="MD5",cnonce="1383887920463",qop="auth",nc="00000001" *INVITE FROM SOFIA* INVITE sip:1000@192.168.2.30;avaya-sc-enabled;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.2.42;rport;branch=z9hG4bKjtc19v5754K6c Max-Forwards: 69 From: "Extension 1001" <sip:1001@192.168.2.42>;tag=QvNQrZya6vHjS To: <sip:1000@192.168.2.30;avaya-sc-enabled;transport=udp> Call-ID: f73b031a-bc45-122f-388f-000c291b7d13 CSeq: 23099169 INVITE Contact: <sip:*mod_sofia*@192.168.2.42:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.6-hacked-20120115T023659Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 427 X-FS-Support: update_display Remote-Party-ID: "Extension 1001" <sip:1001@192.168.2.42 >;party=calling;screen=yes;privacy=off

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