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Outbound call with password
by baskar 30 Dec '13

30 Dec '13

Hi All, How to fetch password from MySQL through dialplan in freeswitch. In MySQL i have table name with "ext_passwd" in that 1001 password is 4587. How to enable outbound call with password protected. If i need to make a outbound call from extension say 1001 to some toll free number. When i dial it should prompt for password. If password successful outbound should be enabled or else it should be Hangup. As off now there is no restriction on outbound calls. Through prefix calls are routed to make outbound calls. Can some one assist me how to fetch data from MySQL in dialplan. Thanks, N.Baskar -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Outbound-call-with-password-t… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Help with Registration Failed with status DNS Error [503]
by Jim Lynch 29 Dec '13

29 Dec '13

Every few days FS stops working with that error. Here's a sofia log level all 9 log. http://pastebin.com/CmhY84W9 DNS works fine from the command line. It's Ubuntu 12.04 FS stable. A restart of FS resolves the problem but that can't be a permanent fix. Thanks, Jim.

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answer via .net esl is getting ignored
by Yisroel M. Olewski 29 Dec '13

29 Dec '13

Hello everyone This is my first posting via the mailing list I'm trying my hand at freeswitch esl with .net I've managed to get a hold of all incoming events, but can't seem to "pick up the phone" my first goal is to pick up the phone and read out some text via tts I've scoured the web, tried everything I can find or think of but no luck here's my code Private Sub winMain_Loaded(sender As Object, e As RoutedEventArgs) Handles Me.Loaded ThreadPool.QueueUserWorkItem(New WaitCallback(Sub(stateInfo) FreeSwitch = New ESLconnection("10.0.0.1", 8021, "mypass") Dim eslEvent = FreeSwitch.SendRecv("event plain ALL") FreeSwitch.SendRecv("event plain DTMF") If eslEvent IsNot Nothing Then AddEvent(eslEvent) While FreeSwitch.Connected eslEvent = FreeSwitch.RecvEvent AddEvent(eslEvent) End While End If End Sub)) End Sub Private Sub AddEvent(e As ESLevent) Dim dd = TextCollection(e.Serialize(""), ": ", vbLf) Dim eventname = dd("event-name") 'Dim uid = dd("Unique-ID") Dim uid = dd("Channel-Call-UUID") If dd("Caller-Network-Addr") <> "" Then Dim dcx = DBEntities() Dim ses = dcx.Sessions.SingleOrDefault(Function(x) x.ServerCode = uid) Dim ChannelState = dd("Channel-State") Dim ChannelCallState = dd("Channel-Call-State").ToLower Select Case eventname Case "CHANNEL_STATE" If ses Is Nothing Then ses = New Session dcx.Sessions.AddObject(ses) End If ses.CallerHost = dd("Caller-Network-Addr") ses.CallerName = UrlDecoded(dd("Caller-Caller-ID-Name")) ses.CallerPhone = dd("Caller-Caller-ID-Number") ses.CallerUser = dd("Caller-Username") ses.CalleeUser = dd("Caller-Destination-Number") ses.ServerCode = uid If {"cs_destroy", "cs_hangup"}.Contains(ChannelState) Then ses.HungUpOn = Now Case "CHANNEL_CALLSTATE" If ChannelCallState = "ringing" Then Dim evn = FreeSwitch.Execute("answer", String.Empty, uid) Dim res = evn.Serialize("") End If Case "CHANNEL_DESTROY" If ses IsNot Nothing Then ses.HungUpOn = Now End Select dcx.SaveChanges() End If End Sub the answer line is getting reached, and the result is "+OK" but the phone still keeps ringing please note that I'm unsure which is the correct uuid, but trying both of them gave me identical results As per advise from Nuwan Wijerathne, I put the "answer" action in the dialplan, and tried just the playback from code (just a wav file), but the result is the same. Freeswitch ignores entirely my commands. It picks up, and immediately hangs up because it has finished the whole dialplan I'd appreciate your help. I'm pretty new to freeswitch and esl Please advise Sincerely, ymo

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Problem with mod_sms
by peetee23＠hushmail.com 27 Dec '13

27 Dec '13

Dear All, I have a problem with mod_sms. Configuration of chatplan is as this: Here is FS log: 2013-12-26 21:14:21.841090 [INFO] mod_sms.c:336 Processing text message 1001->1004 in context public Chatplan: 1004 parsing [public->default] continue=false Chatplan: 1004(a)test.local Regex (PASS) [default] to(1004(a)test.local) =~ /^(.*)$/ break=on-false Chatplan: 1004(a)test.local Action send() 2013-12-26 21:14:21.841090 [DEBUG] sofia_presence.c:224 Can't find registered user 1004(a)test.local But user is registered: in fs_cli #sofia_contact 1004 sofia/internal/sip:1004@1.2.3.4:40348;rinstance=72ECC02F;transport=tls;fs_nat=yes;fs_path=sip%3A1004%4014.3.2.1.%3A40348%3Brinstance%3D72ECC02F%3Btransport%3Dtls Out of default configuration I have just forced TLS connection (tlsv1). Dont you have an idea where could be problem? Thanks everyone for help PT

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TLS error?
by Federico Beffa 27 Dec '13

27 Dec '13

Hi, I'm trying to setup FS as a gtalk client. I'm working on Linux Ubuntu 10.04 and have cloned the git FS repository about 1 week ago. I can compile everything with no errors. However, after configuring client.xml and starting FS I see the following error message in the freeswitch.log file: 2010-07-20 23:38:56.392853 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/freeswitch/mod/mod_ dingaling.so **/opt/freeswitch/mod/mod_dingaling.so: undefined symbol: gnutls_global_init** TLS comes from the following Ubuntu packages: libgnutls26, libgnutls-dev. I would appreciate any suggestion on how to investigate further the problem. Could it be a gnutls library version incompatibility? A bug in FS? In the ubuntu gnutls packages? Or simply an error in the configuration? Thanks for you help! Fede

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Problems about STUN and one way audio
by alex 27 Dec '13

27 Dec '13

Hi, all I have a server that has installed the FreeSwitch software, and the server has a public IP addr. My clients are android phones, the CSipSimple softphone runs on them. There are some issues when these phone call each other. When phone A calls phone B, only the phone B can listen the audio coming from phone A, but the phone A cannot. What's more, the phone B always need some delays to listen the audio from phone A. Therefore, I guess I need set a STUN server to deal with the above problem. So I read related web pages, such as http://wiki.freeswitch.org/wiki/NAT_Traversal. But, How to configure the STUN server in FreeSwitch is ambiguous. Then, I tried to make some configurations as below: vars.xml. <X-PRE-PROCESS cmd="set" data="external_rtp_ip=a.b.c.d"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=a.b.c.d"/> internal.xml <param name="ext-rtp-ip" value="$${ external_rtp_ip }"/> <param name="ext-sip-ip" value="$${ external_sip_ip }"/> external.xml <param name="manage-presence" value="true"/> <param name="ext-rtp-ip" value="$${ external_rtp_ip }"/> <param name="ext-sip-ip" value="$${ external_sip_ip }"/> But, I didn't find where could I set the port of STUN server. So I failed to launch the STUN server. Thus, Could you tell me How to solve the one-way audio problem, and How to configure the STUN server in FreeSwitch? Thanks and sorry if my English is poor. Best regards!

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Grandstream DP710 & DP715 Working Configuration
by Sean Devoy 27 Dec '13

27 Dec '13

Hi All, I didn't see any references to these GrandStream DECT phones in the archives or wiki. I was able to get them working with some minor problems. The "Primary Sip Server" and "Secondary Sip Server" fields in the setup work great with IP Addresses. However, they do not like MY DNS names as they a have underscores in them. My FS is multi-tenant so connecting by name is quite critical. GrandStream identified the problem as the underscore and says it will be fixed in the next firmware release. Anyway, here is a quick review of the phones. The DP715 is the "base station" for the DECT handsets and comes with one handset. It can support 5 handsets and 4 concurrent calls. Each handset is a unique SIP Registration. The voice quality is decent with no background noise. I was a little disappointed that the handsets have no speaker phone or headset options, but that is not so critical in home environments. The price is REALLY good, roughly $65 for DP715 and $35 for each additional DP710. I have only just started looking into these, but that is FAR cheaper than the CISCO offering. Even though the CISCO devices are much more expensive, they offer much more like an FXO and FXS port on the base. I hope to try them soon. Overall I give the DP715/DP710 3 stars for features (out of 5) and 4 stars for performance. My Directory settings for these phones are: <user id="XXXXXX">  <variables> <variable name="outbound_caller_id_name" value="XXXXXXXXXXXXXXXX"/> <variable name="outbound_caller_id_number" value="XXXXXXXXXXX"/> <variable name="internal_caller_id_name" value="XXXXXXXXXXXXXXXX"/> <variable name="internal_caller_id_number" value="221"/> <variable name="user_context" value="from-internal-XXXXXX"/> <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> </variables> <params> <param name="password" value="XXXXXXXX"/> <param name="manage-shared-appearance" value-="false" />" <param name="NDLB-force-rport" value="safe"/> </params> </user>

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fisheye shows No activity found for a few days
by Seven Du 26 Dec '13

26 Dec '13

Just found Fisheye shows No activity found. for a few days, any problem on the back end? -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

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ORIGINATOR_CANCEL and dialplan proccessing
by Anton Kvashenkin 26 Dec '13

26 Dec '13

Hello guys, I need to run system command after getting ORIGINATOR_CANCEL status message. Judging by http://jira.freeswitch.org/browse/FS-3561 i can get it working by adding <action application="set_zombie_exec"/> to my dialplan. But it proceedes anyway even after successfully hanging up call. My snippet of dialplan: http://pastebin.freeswitch.org/21771

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TLS Audiocodes
by Szasz Szabolcs 26 Dec '13

26 Dec '13

Hi, Did anybody set up TLS between Freeswitch and Audiodes MP11X ? I got to work TLS between freeswitch and a softphone (phonerlite), but I have problem with Audiocodes during the TLS authentication. I've loaded the certification but it still doesn't work. Can I debug the tls in freeswitch? Please help me. Thank you in advance. Szasz Szabolcs

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