June 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

TERMINATOR_USED_VARIABLE
by Alex 20 Jun '12

20 Jun '12

I note that in switch_ivr_play_say.c there's an intriguing reference to SWITCH_READ_TERMINATOR_USED_VARIABLE on line 2015: if ((min_digits == 0) && (strlen(digit_buffer) == 0) && switch_channel_get_variable(channel, SWITCH_READ_TERMINATOR_USED_VARIABLE) != 0) It seems to me that this suggests that the terminator used in PAGD is stored, at least temporarily in a channel variable. Is there any way I can get my mitts on the value of this following a call to PAYG?

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FreeSWITCH Conf Call Today: SIP 101
by Michael Collins 20 Jun '12

20 Jun '12

Hey all, Don't forget that at 1pm EDT, 10am PDT we have the FreeSWITCH community conference call. Darren Schreiber will be doing the SIP 101 discussion. We look forward to having you all with us. -Michael

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gateway registration using TLS
by Mi Ke 20 Jun '12

20 Jun '12

Hi All, Is it possible to explicitly tell sofia to register as a client using TLS? Here's my TLS-related part of sofia config: <profiles> <profile name="external"> <domains> <domain name="all" alias="true" parse="false"/> </domains> <gateways> <gateway name="xxx"> <param name="register" value="true"/> <param name="username" value="yyy"/> <param name="realm" value="remote_ip_here:5061"/> <param name="password" value="zzz"/> <param name="register-transport" value="tcp"/> <param name="expire-seconds" value="3600"/> <param name="retry-seconds" value="60" /> </gateway> </gateways> <settings> ... <param name="context" value="external"/> <param name="sip-port" value="5060"/> <param name="tls" value="true"/> <param name="tls-only" value="false"/> <param name="tls-bind-params" value="transport=tls"/> <param name="contact-params" value="tport=tcp"/> <param name="tls-version" value="sslv23"/> <param name="tls-cert-dir" value="/usr/local/freeswitch/conf/ssl"/> <param name="tls-sip-port" value="5061"/> <param name="dialplan" value="XML"/> ... </settings> </profile> </profiles> But that does not work... tport_connected( 0x7f9b44014110): events CONNECTED tport_send_event(0x7f9b44014110) - ready to send to (tcp/remote_ip_here:5061 http://212.58.166.54:5061 ) tport_vsend(0x7f9b44014110): 600 bytes of 600 to tcp/remote_ip_here:5061 http://212.58.166.54:5061 tport_vsend returned 600 tport_set_events(0x7f9b44014110): events IN tport(0x7f9b44014110): reset timer tport_wakeup(0x7f9b44014110): events IN HUP ERR tport_release(0x7f9b44014110): 0x7f9b4401a460 by 0x7f9b4402ca70 with (nil) nta: REGISTER (29733430): Connection reset by peer (104) with tcp/[remote_ip_here]:5061 nua(0x7f9b440196a0): event r_register 503 Service Unavailable Any ideas what I'm doing wrong? Thanks / MiKe

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lua library
by Alex 20 Jun '12

20 Jun '12

Got some fairly common lua functions that I'd like to use as a library. I note that there is a "require" feature in lua, but that requires modification of the environment variables. I was wondering if there isn't a simpler "include" type mechanism where I can specify the full path of a lua script?

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How to configure FreeSwitch to use specific transport when offering calls to a SIP Gateway
by Venkatesh 20 Jun '12

20 Jun '12

Hi ! I am new to freeswitch. Apologies if this question was already answered in the mailing list. I have a simple IVR application that answers a call and transfers the caller to a specific number. The transfer target requires that I offer the call to a gateway. I want to configure freeswitch to use tcp as the transport for the transfer target. The incoming call arrives over UDP. I configured the following in the gateway configuration: <param name="contact-params" value="transport=tcp"/> However, I don't see this configuration making FS use TCP. The call continues to use UDP. In addition, FS adds two contact parameters (transport=tcp; transport=udp) in the outgoing invite. Would appreciate if you can let me know if I am missing something. Venkatesh

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Clarification needed on setting up group calling parameters
by toddb＠toddbailey.net 20 Jun '12

20 Jun '12

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Re: [Freeswitch-users] Gtalk does not ring local phones when making outbound calls
by mtle 20 Jun '12

20 Jun '12

Hi Brian, I am a novice to both FS and VoIP. I just follow the wiki to setup Google voice. I thought "ring_ready" will do. Could you please give me a pointer to the wiki?Thanks. Google doesn't generate a ringback, you'll have to generate that yourself. Please see the wiki on how to generate a ringback before audio is negotiated.Brian Foster Endigo Computer LLCSent from a mobile device. On Jun 19, 2012 2:30 AM, "curriegrad2004" <curriegrad2004(a)gmail.com> wrote: Do you mean a ringback when you're calling out via google voice? On Mon, Jun 18, 2012 at 1:48 PM, mtle <getj2k(a)yahoo.com> wrote: Hi,I have successfully compiled and installed FS from the latest git as at today. Everything seems OK. The only issue is that I do not get any rings at all with gtalk. I wonder if anyone have had this problem lately and how to fix that? Here's my dialplan. <extension name="gv_out"> <condition field="destination_number" expression="^1?(\d{10})$"> <action application="set" data="hangup_after_bridge=true"/> <action application="ring_ready"/>   <action application="bridge" data="dingaling/gtalk/+1$1(a)voice.google.com"/> </condition> </extension> Thanks. _________________________________________________________________________ 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 Join Us At ClueCon - Aug 7-9, 2012 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 _________________________________________________________________________ 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 Join Us At ClueCon - Aug 7-9, 2012 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 --- On Tue, 6/19/12, freeswitch-users-request(a)lists.freeswitch.org <freeswitch-users-request(a)lists.freeswitch.org> wrote: From: freeswitch-users-request(a)lists.freeswitch.org <freeswitch-users-request(a)lists.freeswitch.org> Subject: FreeSWITCH-users Digest, Vol 72, Issue 174 To: freeswitch-users(a)lists.freeswitch.org Date: Tuesday, June 19, 2012, 11:19 AM Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. FaxxBochs hackers? (Sean Devoy) 2. Call Recording file format(s) (Adelia C.) 3. Re: Gtalk does not ring local phones when making outbound calls (Brian Foster) 4. Re: Multicast RTP collector (Brian Foster) 5. Re: FaxxBochs hackers? (Brian Foster) _______________________________________________ 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

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uuid_simplify from Lua Script
by Ken Armstrong 20 Jun '12

20 Jun '12

Hi All, Can someone give me an example of how to use uuid_simplify from a lua scripts. Ken

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how to implement real time in freeswitch ?
by Samira Mh 20 Jun '12

20 Jun '12

hi guys, i am going to implement REAL TIME on freeswitch like (ARA) Asterisk Realtime Arcitecture how can i do that in freeswitch? plese guide using link address thanks so much

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passing hylafax events via mod_spandsp
by Anita Hall 20 Jun '12

20 Jun '12

Hi I am using Hylafax and soft modems emulated via mod_spandsp to receive and send Fax, details given here http://wiki.freeswitch.org/wiki/HylaFax Is there a way to pass on the HylaFax events back to FreeSWITCH via mod_spandsp ? In particular, I would like to communicate the status of incoming Fax like success / failure / error etc back to FreeSWITCH. Thanks. regards, Anita

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