- FreeSWITCH-users - FreeSWITCH Mailing Lists

How to dial an extension in Freeswitch after the call in connected?
by Frankie Yiu 07 Jun '11

07 Jun '11

Hi there, I have a question regarding having FreeSwitch dials an extension number after the call is connected. Here is the condition: Freeswitch would get a call information from database and make an outbound call. Once it is connected, and if there is an extension that we want to reach at that number, we would wait 5 second and dial that extension number. Right now I am using C# code and I don't see any API would allow me to dial an extension. Could someone please help me? Thanks, Frankie

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SIP Response Headers
by Eric Beard 07 Jun '11

07 Jun '11

Hello, I am trying to set SIP response headers using something like this in the dial plan: <action application="set" data="sip_ph_X-MyVariable=true"/> I haven't been able to get it into the SIP headers, however. I have a simple setup for testing: Soft Phone -> Freeswitch -> Gateway I want Freeswitch to add the header to the SIP response sent back to the soft phone. Any ideas what I might be missing? I have also tried the sip_rh_ variant. Thanks, ----------------------- Eric Z. Beard, CTO Loop LLC w (877) 850-2010 x9249 m (727) 776-2768 eric(a)loopfx.com<mailto:eric@loopfx.com>

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caller added to conference cannot be heard until members speak
by elijah 07 Jun '11

07 Jun '11

Following the example in http://wiki.freeswitch.org/wiki/Conference_Add_Call_Example I have established a conference in which a moderator dials the number of a 3rd member he wishes to conference in. Problem is, the newly added 3rd member cannot be heard until after one of the original members speaks. This effect causes the new member to be saying "hello" without anyone hearing. Can you recommend a solution to get this 3rd member mixed into the conference upon answer?

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how to use mod_mp4
by pooja 07 Jun '11

07 Jun '11

has anyone used mod_mp4 to play a video. if i use play_mp4 it says invalid application play_mp4. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/how-to-use-mod-mp4-tp6410226p… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] Freeswitch dialplan help
by awais nazir 07 Jun '11

07 Jun '11

Thank you Actually I don't want to respond 503 on all cases. I just want to mask response code 480 with 503. So that originator can never get 480. Other cause codes should go naturally. On Tue, Jun 7, 2011 at 11:35 PM, Kristian Kielhofner <kris(a)kriskinc.com> wrote: Set continue_on_fail=true and use respond 503: <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="blah"/> <action application="respond" data="503"/> On Tue, Jun 7, 2011 at 12:05 PM, awais nazir <awais-nazeer(a)hotmail.com> wrote: > Hi

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A DTMF issue
by Yungwei Chen 07 Jun '11

07 Jun '11

Hi, When testing the following javascript program, I notice a problem that it takes longer and longer for freeswitch to respond to DTMF input for some reason. Please refer to http://pastebin.freeswitch.org/16435 for debug logs. Notice that the difference in time between any adjacent pairs of the following is increasing as time goes by. Am I missing something here? switch_ivr_play_say.c:1279 Codec Activated L16@8000hz 1 channels 20ms switch_rtp.c:3302 RTP RECV DTMF 2:960 Thanks. Here's the javascript program: var dtmf_digits = ""; function on_dtmf_28(session, type, digits, arg) { if (type == "dtmf") { dtmf_digits = digits.digit; console_log("dtmf_digits=" + dtmf_digits + "\n"); return false; // barge-in / done } return true; } while (true) { dtmf_digits = ""; session.flushDigits(); session.streamFile("/usr/local/freeswitch/sounds/long-prompt.wav", on_dtmf_28, false); if(dtmf_digits.length == 0) { /* no input */ console_log("no input\n"); session.speak('flite', 'kal', 'no input!', null); } else if(dtmf_digits == "1" || dtmf_digits == "2") { console_log("match "+dtmf_digits+"\n"); session.speak('flite', 'kal', 'you pressed ' + dtmf_digits+ '!', null); } else { /* no match */ console_log("no match\n"); session.speak('flite', 'kal', "no match!", null); } }

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Freeswitch 1.0.6 doesn't have mod_hash.c?
by jesse 07 Jun '11

07 Jun '11

but my fs running time complains : switch_core_session.c:1731 Invalid Application hash. I couldn't find mod_hash.c in src package... any idea? thanks!

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mod_com_g729 version for freeswitch 1.0.6
by rex.alex 07 Jun '11

07 Jun '11

Hello, I just purchased a g729 license for testing purpose and based on http://files.freeswitch.org/g729/INSTALL.txt file, installed fsg729-194-installer. My freeswitch version is 1.0.6. I have unloaded the mod_g729 and when I am trying to load mod_com_g729, it shows the following error. Please assist. freeswitch@internal> load mod_com_g729 -ERR [module load file routine returned an error] freeswitch@internal> 2011-06-07 03:03:08.294418 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_com_g729.so **Trying to load an out of date module, please rebuild the module.** Thanks, Rex. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-com-g729-version-for-free… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Questions
by Fabian Borot 07 Jun '11

07 Jun '11

Hello We are giving Freeswitch a try in our organization to provide H323 <-> SIP functionality. We have 2 main questions/requests before starting installing/configuring: 1- What would be the most recommended module for H323? I see there are two [opal and mod_h323] both both say that are in development stages. 2- Does FS fully supports virtual interfaces [eth0:1]? I mean: 2- 1 listening on several virtual interfaces, 2-2 replying from the virtual interface that the request came in instead of replying from the primary interface 2-3 and also we need to specify the interface to send the call out instead of using the primary interface Thank you Fabian

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FS does not uses the IP/PORT resolved using STUN.
by DEVARAHALLI, PRAKASH (PRAKASH) 07 Jun '11

07 Jun '11

I configured FS to use the STUN server to find the public IP address used by the NAT. FS sends the request to STUN and receives the reply from the STUN Server , But it does not uses this info ( IP and port obtained from STUN) neither while negotiation ( in SDP) nor in the RTP. My testbed: ----------- < FreeSwitch > ---- < NAT > ---+----<NAT>--- < SIP Phone Client > | | [ STUN Server] File conf/sip_profiles/internal.xml has below configuration. <param name="ext-rtp-ip" value="stun:120.120.120.120"/> ; This is the local STUN server. <param name="ext-sip-ip" value="stun:120.120.120.120"/> <param name="stun-auto-disable" value="false"/> <param name="rtp-ip" value="auto"/> Thanks Prakash

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