March 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Dialplan bridge results in choppy audio
by Peter Olsson 14 Mar '12

14 Mar '12

Yeah - that's the reason for bad audio. If you want to be sure - never use virtual stuff, at least not for production systems. The main issue is that timing in virtualized environments are not good enough. If you Goggle it, you will find loads of information about this. Much related to Asterisk, but it's the same problem for all VoIP systems. /Peter Från: Adam Kelloway [mailto:adam.kelloway@newpace.ca] Skickat: den 14 mars 2012 13:42 Till: FreeSWITCH Users Help Kopia: Peter Olsson Ämne: Re: Re: [Freeswitch-users] Dialplan bridge results in choppy audio These are virtual machines. Would this have a negative effect? On 3:59 PM, Peter Olsson wrote: Are you running on real hardware, or virtual machines? /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Adam Kelloway Skickat: den 13 mars 2012 17:31 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] Dialplan bridge results in choppy audio Hi there, Using the dialplan tool, I am trying to bridge a call on one freeswitch instance to another freeswitch instance on the same network. The call is bridged as expected, but the audio is very choppy. The first few seconds is basically inaudible, and after that the quality is better but still quite poor. In searching previous threads, I could only find audio problems related to using loopback, but nothing for the bridge dialplan tool. Also note that the ptime is 20 on both legs. Am I missing something here? Is there another way I should be transferring a call to another freeswitch instance? Here is what the bridge looks like: EXECUTE sofia/profile_1_external/userA(a)hostA.com<mailto:sofia/profile_1_external/userA@hostA.com> bridge(sofia/internal/userB@hostB) Both instances are running 1.0.head-git-72bb196 2012-03-12 16-07-56 -0700 Thanks, Adam _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@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<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Adam -- [NewPace Logo] Adam Kelloway Software Engineer, NewPace phone +1 (902) 406-8375 x1031 email Adam.Kelloway(a)NewPace.com<mailto:Adam.Kelloway@newpace.com> aim<aim:GoIm?screenname=Adam.Kelloway@newpace.com>/msn<msnim:chat?contact=Adam.Kelloway@newpace.com> Adam.Kelloway<aim:GoIm?screenname=Adam.Kelloway@newpace.ca>@NewPace.ca !DSPAM:4f608f8332762193319106!

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FreeSWITCH Conference Call Today
by Michael Collins 14 Mar '12

14 Mar '12

Hello all, Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_03_14 Ken Rice will be discussing the feature freeze and versioning. Please join the discussion to learn more about how you can help the project get up to version 1.2 as quickly as possible! -Michael

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stip variable
by Miha Zoubek 14 Mar '12

14 Mar '12

Hi, how can I strip this sofia/internal/18005551212(a)tf.voipmich.com, so that I would have only number 18005551212? This return enum. Thanks! Miha

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make[1]: *** [mod_shout-install] Error 1
by Владислав Гришин 14 Mar '12

14 Mar '12

I have FreeSWITCH Version 1.0.head (git-d827cfe 2012-03-04 17-48-30 -0600) on Centos 5.5 >From src directory I run command first time [root@FS-A freeswitch]# make mod_shout-install : --2012-03-11 13:59:40-- http://files.freeswitch.org/downloads/libs/libshout-2.2.2.tar.gz Resolving files.freeswitch.org... 67.201.31.224 Connecting to files.freeswitch.org|67.201.31.224|:80... connected. HTTP request sent, awaiting response... 200 OK Length: 478582 (467K) [application/x-gzip] Saving to: `libshout-2.2.2.tar.gz' 100%[======================================================================= ===================================================>] 478,582 969K/s in 0.5s 2012-03-11 13:59:40 (969 KB/s) - `libshout-2.2.2.tar.gz' saved [478582/478582] /bin/sh: patch: command not found make[3]: *** [/usr/local/src/freeswitch/libs/libshout-2.2.2] Error 127 make[2]: *** [install] Error 1 make[1]: *** [mod_shout-install] Error 1 make: *** [mod_shout-install] Error 2 [root@FS-A freeswitch]# second time [root@FS-A freeswitch]# make mod_shout-install : Making all in doc Compiling /usr/local/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c... mkdir .libs Compiling /usr/local/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c ... Creating mod_shout.so... installing mod_shout.so [root@FS-A freeswitch]# All ok Vladislav Grishin

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Re: [Freeswitch-users] connexion attempts behing a firewall (wtf?)
by Peter Olsson 13 Mar '12

13 Mar '12

If you get USER_NOT_REGISTERED, that means that the user is not registered. Please do a "show registrations", and you will see who is registered to your server. Also - if you want to disable NAT for now (and don't let FS punch holes in your fw) you can also start FS using the "-nonat" switch. /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Bzzz Skickat: den 12 mars 2012 00:39 Till: freeswitch-users(a)lists.freeswitch.org Ämne: Re: [Freeswitch-users] connexion attempts behing a firewall (wtf?) On Sun, 11 Mar 2012 19:21:01 -0400 Brian Foster <bdfoster(a)endigotech.com> wrote: > This happens all the time. You need fail2ban set up properly. There > are articles on both the freeswitch and fusionpbx wikis. It is, and tested as the wiki says:) but attacker isn't banned. This show me this is also linked to my previous post: softphones & fusionpbx say they're registered, but when I try to make a call between them FS throw me USER_NOT_REGISTERED errors, this is why the attacker's not jettisoned: FS don't seem to answer to a correct registration:( > I run a public server > and I get people doing scans all the time. Does it worry me? No. Of > course not. That's what fail2ban and other measures are for. Interesting, could you describe "other measures", PLS? -- Draft beer, not boys! _________________________________________________________________________ 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 !DSPAM:4f5da5e232761141321256!

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Failed to pass arguments to lua script of FreeSwitch ('arg' and 'arg[i]' are nil values)
by king2kin 13 Mar '12

13 Mar '12

Hi all, It's said we could access the arguments passed to lua script by variable 'arg[i]' where i > 0. for example, { lua dummy.lua aa 100 } then, we will have as follows: arg[1] -> aa arg[2] -> 100 However, I always get 'nil' values for 'arg[i]' or 'arg' while running the lua script. How can I pass and retrieve arguments or parameters to a lua script of FreeSwitch?

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Remove INFO and MESSAGE from SIP Invite Header
by Rob Moore 13 Mar '12

13 Mar '12

Hi Guys, I'm looking for a way to remove INFO and MESSAGE from the SIP Invite message freeswitch generates when I attempt to use a certain gateway. The system used by one of our providers to provide a TLS solution doesn't accept these methods and is rejecting a call. I've tried to remove INFO by setting <param name="dtmf-type" value="rfc2833"/> within the gateway configuration and attempted to disable message presence ( <param name="manage-presence" value="false"/>) to remove the MESSAGE method request but neither seem to have any effect. Does anyone have any suggestions on how I can remove these? Thanks Rob

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SIP Invite Contact Header
by Rob Moore 13 Mar '12

13 Mar '12

HI All, Hopefully a quick question here. I'm trying to get TLS working with one of our gateway providers and they are asking that I pass an FQDN instead of the external IP address of our Freeswitch server in our contact header within our sip invite. I've tried setting sip_contact_user but that doesn't seem to have any effect on the contents of the contact header. Plus I would really prefer to set this in the gateway profile so it only effects calls passing over a single gateway. Anyone have any advice on how this can be achieved? Thanks Rob

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Dialplan bridge results in choppy audio
by Adam Kelloway 13 Mar '12

13 Mar '12

Hi there, Using the dialplan tool, I am trying to bridge a call on one freeswitch instance to another freeswitch instance on the same network. The call is bridged as expected, but the audio is very choppy. The first few seconds is basically inaudible, and after that the quality is better but still quite poor. In searching previous threads, I could only find audio problems related to using loopback, but nothing for the bridge dialplan tool. Also note that the ptime is 20 on both legs. Am I missing something here? Is there another way I should be transferring a call to another freeswitch instance? Here is what the bridge looks like: EXECUTE sofia/profile_1_external/userA(a)hostA.com bridge(sofia/internal/userB@hostB) Both instances are running 1.0.head-git-72bb196 2012-03-12 16-07-56 -0700 Thanks, Adam

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user dial-string getting variables
by Antonio 13 Mar '12

13 Mar '12

Hi, Is it possible to access a variable that is passed per channel, in some script called in the directory.xml, parameter dial-string? I need to be able to set the var1 dynamically from the dialplan, so i can parse it from my lua script. Using the set command directly it will set the variables for both channels. For example, I dialed two users: <extension name="test" continue="false"> <condition field="destination_number" expression=".*"> <action application="bridge" data="[leg_timeout=60,var1=555]user/100@default.com|[leg_timeout=60,var1=777]user/101@default.com"/> </condition> </extension> then in my directory.xml, i have: <param name="dial-string" value="${lua(dial.lua ${dialed_user} ${dialed_domain} ${var1})}"/> So when the return dial-string for the user 100 has it's own var1, and when dial the user 102 has another value. I already try to uuid_dump in dial.lua, but i can't see the variable var1... Using this method, [var1=value], it will set the variable var1 for the B channel, but can i access it from the A channel? Or is there another way to set this variables before processing the dial-string... Thanks, António -- Un cordial saludo / Best regards, _________________________ António Silva E-mail:asilva@wirelessmundi.com

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