April 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Using Softphone on same machine as FreeSwitch?
by LearningFS 24 Apr '12

24 Apr '12

Hi guys, I am on Ubuntu 10.04 inside a VirtualBox virtual machine (my host OS is Windows 7). I am running FreeSwitch on Ubuntu. However, when I try to connect with any softphone within my Ubuntu, it cannot connect to FreeSwitch and FS does not receive the connection requests on command prompt. I tried connecting to port 5060 on the "inet addr:" shown in ifconfig. I actually was able to get an android phone to connect to my windows public IP address after setting up port forwarding, but is it possible for me to get an Ubuntu softphone to connect as well so I can keep all my testing in one machine? Also, I want to test out mod_sms. I have X-Lite on my Windows machine, I guess this is sufficient. Thanks and sorry if this was answered before. I tried searching for it. -- View this message in context: http://old.nabble.com/Using-Softphone-on-same-machine-as-FreeSwitch--tp3364… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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mod_lumenvox ?
by Anita Hall 24 Apr '12

24 Apr '12

Whatever happened to mod_lumenvox ? regards, Anita

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Freeswitch and Opensips
by Miha 24 Apr '12

24 Apr '12

Hi to all, I have configured opensips that works like load_balancer (wiki on FS page). So, we have SBC, trunk is made to Opensips, than opensips is load_balacing to FSs. When you call from FS, call is going to Opensips, then opensips redirect the call to SBC, if call must go outside. My question: FSs boxes are all the same (same dialplan, directory configuration, etc.). What happens if call is made to a user, which is registered on one of the FS box? I guess I must change dialplan and set that all trafic FS send to opensips (not internally), because if someone calls to a local user, but user is registered on other FS box, FS will behave as user is not registered. So, will opensips know which user is registered on which FS box when opensips will be doing load_balacing? Thanks for explenation. Regards, Miha

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detect tone
by dinh thuc 24 Apr '12

24 Apr '12

Hi I have a question about detect tone: I have 2 freeswitch server and I use mod_spandsp to detect tone: 1. I use SIP Phone call to mod_spandsp, it make call to the second freeswitch. 2. Second freeswitch will answer and genarate tone like <action application="playback" data="tone_stream://%%(250,250,480,620);%(500,500,480,620)"/> 3. mod_spandsp will start_detect_tone I have a problem: I create make call from mod_spandsp to second freeswitch I will have 2 session ( channels) 1st session: SIP Phone call to mod_spandsp 2st session: mod_spandsp call to the second freeswitch I try to dectect 1st session, then I can dectect tone But I try to dectect 2st session, then I see that dectect tone don't run What is the problem with 2st session ? Whycan't I detect tone ? And I also problem is I can't detect all tone that I send from second freeswitch and It's too difficult to detect tone B&R Dinh Thuc

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Re: [Freeswitch-users] Incoming fax stopped working when removing mod_fax...
by Peter Olsson 24 Apr '12

24 Apr '12

Have you loaded mod_spandsp? This is the replacement for mod_fax. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Yehavi Bourvine Skickat: den 24 april 2012 10:41 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] Incoming fax stopped working when removing mod_fax... Hi, I am trying to upgrade from a half year olf FS version to the latest one and face some problems with fax. The fax arrives from the PSTN through an Audiocodes MSBG-1,000 gateway. With the old version, when I have mod_fax loaded, FS detects that this is a fax call, sends a re-INVITE to the MSBG and enters T.38 mode; this works ok. With the old version, when mod_fax is unloaded, FS enters T.38 mode without sending re-INVITE, thus the fax fails. The latest FS version (where mod_fax is not available anymore) works as the second situation above and fax fails. I guess I am missing something in the configs, but I can't find what... I have in my dialplan: <action application="set" data="fax_enable_t38_request=true"/> <action application="set" data="fax_enable_t38=true"/> and I've also tried to add: <action application="set" data="execute_on_answer_6=t38_gateway self"/> but nothing helps. Any idea what I am doing wrong? Thanks! __Yehavi: !DSPAM:4f96658d32762460221281!

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Incoming fax stopped working when removing mod_fax...
by Yehavi Bourvine 24 Apr '12

24 Apr '12

Hi, I am trying to upgrade from a half year olf FS version to the latest one and face some problems with fax. The fax arrives from the PSTN through an Audiocodes MSBG-1,000 gateway. With the old version, when I have mod_fax loaded, FS detects that this is a fax call, sends a re-INVITE to the MSBG and enters T.38 mode; this works ok. With the old version, when mod_fax is *unloaded*, FS enters T.38 mode *without *sending re-INVITE, thus the fax fails. The latest FS version (where mod_fax is not available anymore) works as the second situation above and fax fails. I guess I am missing something in the configs, but I can't find what... I have in my dialplan: <action application="set" data="fax_enable_t38_request=true"/> <action application="set" data="fax_enable_t38=true"/> and I've also tried to add: <action application="set" data="execute_on_answer_6=t38_gateway self"/> but nothing helps. Any idea what I am doing wrong? Thanks! __Yehavi:

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mod_httapi
by Tom Parrott 24 Apr '12

24 Apr '12

Hi, Does anyone have a short guide or blog post on using the mod_httapi. The wiki page is a bit sparse on details of how to actually use this new module, and have not been able to make much progress with it. One of my main questions is what is the "action" property of the work element for. The wiki says it "Change the new target url." http://wiki.freeswitch.org/wiki/Mod_httapi But what does that mean? Thanks Tom

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(no subject)
by Mikael Nordman 24 Apr '12

24 Apr '12

Hi, I have a asr dialogue with two consecutive play_and_detect_speech calls. The say: part is empty and the detect: part uses nuance grxml grammar. In the second play_and_detect invocation freeswitch looses some of the userplane data, thus causing the speech detection to fail. >From the userplane data forwarded by freeswitch to the Nuance NSS server, about one second of speech is missing in the beginning of the phrase If I add say: with some content, then all works fine. What could cause this loss of userplane data ? Cheers /Mike

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Speex file format
by Bernard Fluixa 23 Apr '12

23 Apr '12

Hello, I cannot play ".spx" (speex) file in commands such as originate… &playback(file.spx). I double-checked that mod_speex is loaded OK. Freeswitch returns 2012-04-19 11:35:40.532068 [ERR] switch_core_file.c:122 Invalid file format [spx] for [test.spx]! I converted file to wav format and could play it (speexdec test.spx test.wav) with playback command. I probably missed a point here but I would like to know how I can play speex files without having to convert them to wav first. Anyone? Thank you Bernard

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per user/call session_timer
by Serge S. Yuriev 23 Apr '12

23 Apr '12

Hello, Can I disable session timers on per user or per call basis? -- wbr, Serge

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