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any way to use a hook/event for voicemail notification instead of email?
by Nathan Neulinger 23 Jun '14

23 Jun '14

Seems like it would enable a bit more functionality if the notification were optionally handled as a hook, with the msg id/other data as parameters. In addition to enabling a bit more control of the content beyond the basic templates - this might also enable other functionality - such as triggering automatic transcription. I didn't see anything in the mod_voicemail wiki page, but figured I'd ask if there was any other trick for accomplishing this. From a look over the code, I'm not seeing any obvious way to have the event based codepath trigger but not the email notification unless I've missed something. A minimal means to handle this might be adding a parameter to disable voicemail email notification, but leave in all of the voicemail event generation. Thoughts? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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FreeSwitch running on VM. What all ports need to be forwarded.
by Anurag Rana 23 Jun '14

23 Jun '14

Hi all, I am running FS on a Virtual Machine which in turn is running on Windows server. I want to know what all ports I need to forward from Windows server to VM apart from 1. Sip Port - 5060 2. WebRTC Port - 9066 [ Reason for asking this question <https://jira.freeswitch.org/browse/FS-6618?filter=-2> ] -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences.

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Getting issue while using mod_rtmp
by Akib Sayyed 23 Jun '14

23 Jun '14

guys I am using mod_rtmp in freeswitch I am using flex client from freeswitch git repo. I am able to connect ,login place a call but not able to hear any voice or call is getting disconnected even if context is default it not routing properly here is cli log http://pastebin.com/DvaXECdl -- Akib Sayyed Matrix-Shell akibsayyed(a)gmail.com akibsayyed(a)matrixshell.com Mob:- +91-966-514-2243

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Emulate typical house phone/answering machine
by d-OP 23 Jun '14

23 Jun '14

Is there a way to configure freeswitch to allow a callee to screen a call by eavesdropping on the caller leaving a voicemail and then, if desired, pick up the handset to accept the call/pull it back from voicemail? This was discussed briefly here in Aug of 2012 but there was no resolution. Anybody have a working example they can share? Thanks, -Terry iPhone says Hello World!

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ussd on gsmopen
by Shlomi Agiv 23 Jun '14

23 Jun '14

(moved from a different thread) I get an issue with ussd - sending ussd always returns the text "just a ussd message" whatever i send Anyone encounter that? Thanks, Shlomi

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No audio on both sides. Call is made and received successfully but there is no audio.
by Anurag Rana 22 Jun '14

22 Jun '14

Hi All, I installed FS on *VM*(Ubuntu) running on a window server. I have Twinkle softphones installed on 2 different machines in same network. Both twinkle softphones are registered and connected to FS. When I make a call from one twinkle to other, Call is successfully made and received. But there is no audio on any side. then Call is disconnected after 32 seconds automatically. I logged a defect on JIRA but its closed. They say its a Network issue. I have pasted the *siptrace* log here --> http://pastebin.freeswitch.org/22780 JIRA link -> https://jira.freeswitch.org/browse/FS-6618?filter=-2 Let me know if more trace/log/info is required. Please suggest me what needs to be done in order to mitigate the problem. -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences.

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A leg not able to hear the user busy message using bridge_early_media
by lakshmanan ganapathy 22 Jun '14

22 Jun '14

Hi everyone, I'm using freeswitch stable version 1.2.23 with wanpipe 7.0.8 and libsng_isdn-7.29.0. Recently I faced a issue with regard to early media not been sent to the A leg. The scenario is, 1. There is an incoming call. 2. From dialplan, trying to bridge with another number as follows. <action application="bridge" data="{ignore_early_media=false,bridge_early_media=true}freetdm/1/a/99522xxxxx/> 3. Now when the B leg starts ringing, the A leg hears the "ringback" ( which means bridge_early_media works ). 4. B leg rejects the call. So TELCO sends a PROGRESS message normally saying "The number you are trying to reach is currently busy etc...". 5. The A leg was not able to hear the above said message, and it hears only the "ringback". 6. Then finally TELCO sends a DISCONNECT, and the A leg got hungup. Some one please tell what I'm missing in this?, Is there some other variable to set?. Please help. Attached the freeswitch.log and the pcap files.

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No audio on both sides. Call is made and received successfully but there is no audio.
by Phil Quesinberry 22 Jun '14

22 Jun '14

Hi Anuraq, Welcome! Now the fun begins. At first glance it looks like you probably need to forward the RTP ports to FS. I believe the default port range is 16384 to 32768 and that setting should be in <FS-dir>/conf/auload_configs/switch.conf.xml. RTP is what the media (audio/video) goes over. Your two phones are on the same network but FS is on a different subnet, and looking at the SDP, FS is proxying the audio from that different subnet, (the default behavior) so those ports will need to be forwarded. Make sure that those ports are allowed through Ubuntu's firewall as well. You may want to consider tightening up the default RTP port range so the ports you need to forward don't conflict with anything else. I believe that you'll need two for each channel, so change it to whatever works for your intended setup and forward those ports to the FS box, I'd probably reduce the range to 512 or 1024 ports but that's just me. It appears that you're already forwarding SIP port 5060. If you want FS to accept calls from "outside" (A DID for example), then you'll want to forward port 5080 as well with the default setup. If you want to enable TLS, additional SIP profiles, etc. then you'll need to forward additional ports. Cheers, - Phil _____ Hi All, I installed FS on *VM*(Ubuntu) running on a window server. I have Twinkle softphones installed on 2 different machines in same network. Both twinkle softphones are registered and connected to FS. When I make a call from one twinkle to other, Call is successfully made and received. But there is no audio on any side. then Call is disconnected after 32 seconds automatically. I logged a defect on JIRA but its closed. They say its a Network issue. I have pasted the *siptrace* log here --> http://pastebin.freeswitch.org/22780 JIRA link -> https://jira.freeswitch.org/browse/FS-6618?filter=-2 Let me know if more trace/log/info is required. Please suggest me what needs to be done in order to mitigate the problem. -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences.

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First build completed with many warnings
by Keith Goggin 22 Jun '14

22 Jun '14

Hi List, I've managed to build 1.2.24+git~20140617T204847Z~9dde4b1548~64bit (git 9dde4b1 2014-06-17 20:48:47Z 64bit) on Linux Mint 17 AMD E-350 dual core and 3.6 GiB memory, it booted up and shut down OK. I'd like to try again with fewer options enabled in the hope of a quicker build and getting a handle on the very many warnings and possible error messages involved. Can someone suggest some config changes to achieve this outcome? I have used <https://wiki.freeswitch.org/wiki/Installation_Guide#Recommended:_Git> and the wiki generally to get this far but am struggling to understand the warnings and don't know which options to pick for a very light weight and basic SIP server for no more than 5 or so users. I'm not sure about the policy on attachment of logs but I'll risk the first two pages of bootstrap.sh console output. Any help appreciated.

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Re: [Freeswitch-users] Why a call hangs up without warning during recording
by Clive Lansink 22 Jun '14

22 Jun '14

Hello. That variable is new to me but it looks like it is what I need. I'll experiment with it. Thank you. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 -----Original message----- From: Anthony Minessale <anthony.minessale(a)gmail.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Why a call hangs up without warning during recording Reply-to: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Date: Sat, 21 Jun 2014 20:57:56 -0500 try setting record_waste_resources=true on the channel On Sat, Jun 21, 2014 at 8:09 PM, Clive Lansink <clive(a)lansink.co.nz> wrote: > Hello. > > I've noticed in a Freeswitch IVR application I've been working on that > when the customer phones in to record a message, the call is hung up > without warning after about five minutes. This does not happen when I call > in locally. When this happens, the log just shows there was a hangup with > normal clearing. But the customer didn't hang up. The customer just notices > visually that their phone hangs up so the hangup seems to be caused > somewhere between the two systems. > > I figured that during a recording, Freeswitch is not generating any audio > at all, and it could be that something in the public phone system times out > after absolutely no audio for a period of time. I have come up with a work > around by getting Freeswitch to generate a short tone repeatedly during > recording, like 50Hz at -30dB so it is basically inaudible, every thirty > seconds or so. This seems to have solved the problem and now the customer > can record for as long as he wants. > > But I suspect this is really a work-around and not the recommended > approach. I did play with the suppress cng parameter in the sofia > configuration but didn't find any setting that would solve the problem. > > Thankfully my work-around has got me home in the meantime but I would > appreciate any comments on this. Does nobody make long recordings these > days? > > Cheers. > > > Clive Lansink > Email: Clive(a)Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > _________________________________________________________________________ > 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 > > 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 > -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 _________________________________________________________________________ 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 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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