- FreeSWITCH-users - FreeSWITCH Mailing Lists

how to use mod_erlang ?
by daqiang wang 19 Feb '10

19 Feb '10

hello ! I test mod_erlang from : http://wiki.freeswitch.org/wiki/Mod_erlang_event but when I input : > {foo, freeswitch@localhost } ! {api, status, ""}. I received (test@wangdq-laptop)2> =ERROR REPORT==== 18-Feb-2010::11:10:13 === Error in process <0.41.0> on node 'test@wangdq-laptop' with exit value: {badarg,[{erlang,list_to_existing_atom,["freeswitch@wangdq-laptop"]},{dist_util,recv_challenge,1},{dist_util,handshake_we_started,1}]} and at freeswitch console : [NOTICE] mod_erlang_event.c:1720 Ignorable error in ei_accept - probable bad client version, bad cookie or bad nodename and I think the mod_erlang config file is : <configuration name="erlang_event.conf" description="Erlang Socket Client"> <settings> <param name="listen-ip" value="0.0.0.0"/> <param name="listen-port" value="8031"/> <param name="cookie" value="ClueCon"/> <param name="shortname" value="true"/>  <param name="apply-inbound-acl" value="lan"/>  <param name="encoding" value="string"/>   </settings> </configuration> ~ Thanks !

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Voicemail quality
by Edward Stevenson 18 Feb '10

18 Feb '10

I have V1.0.4 running on a production server. It's working quite well, except for voicemail retrieval over a satellite internet connection. Voice calls over satellite sound fine, other than the 600ms delay. I'm using the Howler G729 module for G729 transcoding. I've noticed that in voice calls, the bandwidth over the satellite is a steady 24 kbps in both directions. When accessing voicemail, the bandwidth fluctuates with voice in the call. What I mean by that is, pauses in audio in the voicemail, or in the IVR, cause the bandwidth of the call to drop off momentarily. It's almost like it's using a variable bit rate. The audio sounds like there's packet loss. Pops and garbled speach. This is not noticable over a landline. I'm using the built in voicemail. Anyone else had any similar issues? -- View this message in context: http://old.nabble.com/Voicemail-quality-tp27251411p27251411.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Skypiax snd-dummy, One way audio
by Max Bridgewater 18 Feb '10

18 Feb '10

Hi Skypiax Lovers, I'm trying to get Skypiax running on CentOS5.3 but I keep having problems with snd-dummy. My problem is that I have sound only in one direction (inbound). I suspect the sound devices are not set properly. In particular, I can't find "hw:dummy" in the Skype options (Static Build 2.1). Here are the sound devices I see in Skype options: a) Dummy, Dummy, PCM Default Audio 0default:CARD=Dummy) b) HDA Intel, ALC662 Analog (hw:0,0) c) Dummy, Dummy PCM (hw:1,0) d) hdmi (unlnown) Here is what i notice: 1) modprobe snd-dummy returns nothing; implying to me that the module was loaded properly. 2) lsmod | grep "snd" returns: snd_dummy 15553 0 snd_hda_intel 343537 0 snd_hwdep 12869 1 snd_hda_intel snd_seq_dummy 7877 0 snd_seq_oss 32577 0 snd_seq_midi_event 11073 1 snd_seq_oss snd_seq 49585 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event snd_seq_device 11725 3 snd_seq_dummy,snd_seq_oss,snd_seq snd_pcm_oss 42817 0 snd_mixer_oss 19009 1 snd_pcm_oss snd_pcm 72133 3 snd_dummy,snd_hda_intel,snd_pcm_oss snd_timer 24517 2 snd_seq,snd_pcm snd 55237 10 snd_dummy,snd_hda_intel,snd_hwdep,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer soundcore 11553 1 snd snd_page_alloc 14281 2 snd_hda_intel,snd_pcm Any idea? Thanks, Max.

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tone_detect timeout
by Troy Anderson 18 Feb '10

18 Feb '10

In the wiki about tone_detect, the docs state that the timeout is in seconds (e.g. +2 for 2 seconds from now), but all the examples have +5000, suggesting that it may really be milliseconds. I'd be happy to update the wiki if someone could say if it's seconds or millisconds or otherwise. I tried tracing it back in the code, but got lost looking for the definition of switch_media_bug_t! http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect Thanks! Troy

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freeswitch config expert wanted
by Madovsky 18 Feb '10

18 Feb '10

Hi again, as I'm a researcher and work on a (personal) project since 5 years now, I decided to add a "light" VOIP functionality to my project, but after 3 weeks of learning, I decided to stop protocols headdick and waste of my time (as I never wanted to be in the VOIP dev world), it's definetly not my cup of tea. So, if anybody in this emailist wants to help me (I pay of course) to configure freeeswitch as I'm expecting for (I'm sure my config request is really easy but I have not the right informations and example until now) it would be great, and maybe I will take my retirement more early... ;) Thanks Franck Chionna infos at madovsky dot org

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Sending message notifications with openzap
by François Legal 18 Feb '10

18 Feb '10

Hello, I was wondering whether or not a facility was available to send message waiting indicator (from mpd_voicemail) using openzap. I know this feature is not available in openzap (I'm in the process of coding it for analog channels) but as message indication can be sent via mod_sofia, I wonder if a frame is available in FS core. Anybody can shed me some light ? François

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Testers
by Brian West 18 Feb '10

18 Feb '10

Here is what I need: Testers for our G729 installer for linux 32bit and 64bit installs. I have tested CentOS 5x, Debian (excluding 4.0 unsupported) Please contact me off list so I can issue you a temp. license and a make sure it all works. Thanks, Brian

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Forbidden using UDP, works with TCP/TLS
by Mark Campbell-Smith 18 Feb '10

18 Feb '10

Hi! I had a user registered using TLS transport. That was working fine but I want to change the ATA over to use UDP instead. All I thought I should have to do was to change the transport and ports used to register in the ATA (SPA3102). However, when I do this, FS responds with Forbidden. When I change the settings back to use TCP or TLS, registration is successful. What would cause FS to respond with forbidden? I do not change the username/password fields in either case. Thanks

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Re: [Freeswitch-users] Other party does not hear voice when connecting with X-Lite
by Steven Brown 18 Feb '10

18 Feb '10

I had the same problem with XLite / Freeswitch a while back that I never fully understood, however the problem vanished when I disabled all codecs on Xlite except G711 uLaw, as I say, no idea what was going on but this might be worth trying. Steve Message: 1 Date: Sun, 14 Feb 2010 09:04:09 -0600 From: Anthony Minessale <anthony.minessale(a)gmail.com> Subject: Re: [Freeswitch-users] Other party does not hear voice when connecting with X-Lite To: freeswitch-users(a)lists.freeswitch.org Message-ID: <191c3a031002140704g705bfc73rd8dd103f3d846062(a)mail.gmail.com > Content-Type: text/plain; charset="iso-8859-1" You need to describe this again its too confusing now. List each device, freeswitch, the phones and which ip and combo of addrs it uses with the topology clearly stated. Your attempt to simplify your explanation is actually making it harder to follow. Also consider a debug/sip trace as well. Include sofia status profile default. Then capture a test call after entering these commands. console loglevel debug. sofa profile internal siptrace on On Feb 14, 2010 4:12 AM, "Vitalii Colosov" <vetali100(a)gmail.com> wrote: No, it is done on the different PCs... Sorry, when I started the topic, I have described the problem how it is visible from PC of my friend. Then I tried to reproduce the same on my own PC, and you are right...I was not able to hear anything as well, not only both party wasn't. Also, from my PC I was NOT able to hear guitar on test number "9999".

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Greetings and a couple of questions
by Doc 18 Feb '10

18 Feb '10

First, to all, greetings. I am just beginning a quest in trying to setup a simple FS box to route my incoming skype account to a SIP ATA (SPA-1001). I have this installed on Ubuntu 8.04 (since I read that it came with a stock tickless kernel with 100HZ tick). I have (as far as I can tell) compiled the alsa 1.0.20 drivers and included the mod_skypiax dummy file. I followed install instructions in the FS wiki for Hardy & FS & skypiax. 1) I am able to see a call come in, and it gets routed to the sample IVR to start. The first thing off is that when I dial from PSTN -> Skype-In, it does not let me push any buttons. If I launch a second skype client, and dial the skype user on FS directly, it works fine. Any ideas? 2) No matter how Skype works, when I hang up, it throws 3 or 4 errors (2010-02-16 20:41:45.384633 [ERR] mod_skypiax.c:826 rev 16619M[(nil)|37 ][ERRORA 826 ][interface1][-1, 5,21] EXIT? sent=-1), and when I try subsequent dials, I get pages and pages of this error, and no audio. Eventually it crashes FS. Any ideas? 3) When dialing extension to extension, or even testing out the IVR, it all works fine - no errors. The only thing it does make me do, and I haven't tracked it down yet is it waits like 30 seconds before responding (unless I press the # after I type the extension number). Any ideas? 4) When I run startskype.sh, I see these errors (and are they worth concerning?) : expected keysym, got XF86KbdLightOnOff: line 70 of pc expected keysym, got XF86KbdBrightnessDown: line 71 of pc expected keysym, got XF86KbdBrightnessUp: line 72 of pc Could not init font path element /usr/share/fonts/X11/cyrillic, removing from list! Could not init font path element /var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list! ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null Thanks for the patience in letting me email to the group. I used Asterisk in the past, and even though I am finding a bit of a learning curve using FS, I am enjoying it. Hopefully someone will have some insight at least into where or how I can settle the above items down, and I can start building a proper dialplan. Thanks, Doc

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