December 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

NAT Help needed
by Simon Leck 26 Dec '08

26 Dec '08

Hi Everybody I am a newbie in this. I have managed to setup freeSwitch but I am unable to resolve the NAT issue. Hope somebody out there can furnish me with guidance. So far I have manage to use "Use Stun server to resolve registration problem regarding client behind NAT device but behind two layers, NAT devices still have problem in registration. Thanks Everybody for your kind assistance. Simon Email: simon0922(a)gmail.com

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Limiting port for OpenMRCP
by mark morreny 26 Dec '08

26 Dec '08

Hi, I have a dev version of a MRCP-supported TTS, but it can only allow 1 port to connect. Does OpenMRCP has a way to limit the number of port to connect and queue the rest of the request somehow? Thanks for your suggestion and help. Mark

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Setting up port audio for incoming/outgoing calls
by Jason White 26 Dec '08

26 Dec '08

On the wiki, an example of a port audio configuration is given that involves creating a Sip gateway on localhost. As I couldn't get this to work (apparently due to the external profile's detection of NAT), I thought I would try an alternative approach. I am modifying the default dial plan here. At some point I'll probably just rewrite it anyway. I have created a user in the directory for extension 1020. For outbound calls, in default.xml, I have the following: <extension name="portaudio" continue="true"> <condition field="source" expression="mod_portaudio"> <action application="set_user" data="1020@$${domain}"/> <action application="set" data="effective_caller_id_number=1020"/> </condition> </extension> The log shows that the set_user is executed, as is the set effective_caller_id_number (the latter shouldn't be necessary, unless I'm misunderstanding). However, running show channels after making a call from the portaudio device still shows the user name and caller id as FreeSWITCH,0000000000 Also, when I try to call a local extension from the audio device, I get the following in the logs, and the call is terminated. I've checked the code, and clearly the failure to open the file is the cause of the termination. The Sip phone on the extension rings once and then it receives the cancellation from FreeSWITCH. 2008-12-25 10:29:59 [DEBUG] switch_ivr_originate.c:1313 switch_ivr_originate() P lay Ringback File [local_stream://moh] 2008-12-25 10:29:59 [ERR] mod_local_stream.c:308 local_stream_file_open() Unknow n source moh 2008-12-25 10:29:59 [ERR] switch_ivr_originate.c:1322 switch_ivr_originate() Err or Playing File 2008-12-25 10:29:59 [DEBUG] switch_core_codec.c:122 switch_core_session_set_read _codec() Restore original codec. 2008-12-25 10:29:59 [NOTICE] switch_ivr_originate.c:1560 switch_ivr_originate() Hangup sofia/internal/sip:1000@192.168.0.4:2048;line=mxyv04us [CS_CONSUME_MEDIA] [NO_ANSWER] 2008-12-25 10:29:59 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup( ) Send signal sofia/internal/sip:1000@192.168.0.4:2048;line=mxyv04us [KILL] Any hints would be welcome. There is no urgency, of course, as I'm doing this for fun and out of interest. Happy holidays to all on the FreeSWITCH list.

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Behind a router
by kriko 25 Dec '08

25 Dec '08

Hello! I installed freeswitch on different configuration, but can't get registration working. So my phone (192.168.10.2) is now behind a wireless router (192.168.10.1), ethernet port (192.168.0.5) is attached to eth1 (192.168.0.1). Fs is running on that machine (it has eth0 - net and eth1 - inside lan), sofia shows: http://pastebin.com/m6f349c32 and registration fails: http://pastebin.com/m2916e20d How could I fix that? So when I would call 1000(a)212.235.180.41 would ring my phone at 192.168.10.2? I've setup port forwarding on port 5060 between 192.168.10.2 and 192.168.0.1 both ways. --

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what is going on with openmrcp?
by mark morreny 25 Dec '08

25 Dec '08

Hi, I checked out Unimrcp and it seems like it is down. Does anyone know what is happening to Unimrcp? It seems like mod_unimrcp is down. I am looking for a way to integrate Freeswitch to MRCP TTS server. Is there anywhere to do it? Thanks, Mark

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How PBXs works on different platforms
by Evgeniy Zolotov 25 Dec '08

25 Dec '08

Hello! Our experience (over 1 month) of FreeSWITCH testing under SunSolaris 10 shows us such things : 1. FreeSWITCH can be built with gcc compiler, which consists at SunSolaris 10 by default : $ /usr/sfw/bin/gcc --version gcc (GCC) 3.4.3 (csl-sol210-3_4-branch+sol_rpath) Copyright (C) 2004 Free Software Foundation, Inc. This is free software; see the source for copying conditions. There is NO warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. With this compiler is possible built FreeSWITCH except mod_openzap (this module needs ISO C99 compatible compiler), but, it'll be shown below, there are no any reasons to make mod_openzap under Solaris 10. 2. Newlyversion of gcc can be taken from CSW – repositories, for example here : http://mirrors.usc.edu/pub/csw/unstable/i386/5.10/ There are consists all CSW — packages, which gcc needs for install. Full list of CSW — packages in a world can be found here : http://www.opencsw.org/userguide/ So we have : $ /opt/csw/gcc4/bin/gcc --version gcc (GCC) 4.0.2 Copyright (C) 2005 Free Software Foundation, Inc. This is free software; see the source for copying conditions. There is NO warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. >From there we take packages aclocal & automake (they are absent into Sun Solaris 10 distributive ), if we want build FreeSWITCH — trunk starting from ./bootstrap.sh . 3. One more variant of FreeSWITCH installation, which described in a few sources in the Internet (http://www.voiceworks.pl/cypromis/category/solaris-opensolaris/) - is installation of Sun Studio 12 (package SunStudio12ml-solaris-x86-200709-pkg.tar.bz2 , from Sun Microsystems official site): /opt/SUNWspro/bin/cc Build can be done in a command mode, redefining parameters CC, CCOPT, LDOPT, or even in Sun Studio GUI IDE, launching it from remote host : # ssh -nfX <sun-host> sunstudio 4. We built & installed FreeSWITCH + Zaptel + Wanpipe on 2 64-bit SunSolaris servers $ isainfo -k amd64 $ isainfo -v 64-bit amd64 applications tscp cx16 sse3 sse2 sse fxsr amd_3dnowx amd_3dnow amd_mmx mmx cmov amd_sysc cx8 tsc fpu 32-bit i386 applications tscp cx16 sse3 sse2 sse fxsr amd_3dnowx amd_3dnow amd_mmx mmx cmov amd_sysc cx8 tsc fpu + on a 64-bit server SuperMicro — SYS-5025B; proc. Intel Xeon 3210 : # isainfo -v 64-bit amd64 applications ssse3 cx16 mon sse3 sse2 sse fxsr mmx cmov amd_sysc cx8 tsc fpu 32-bit i386 applications ssse3 cx16 mon sse3 sse2 sse fxsr mmx cmov sep cx8 tsc fpu # isainfo -k amd64 - on a last one we didn't even succeed even to start Zaptel + Wanpipe, having exactly the same configuration like on previous two. We didn't succeed to build some of FreeSWITCH libraries with 64-bit support, but that is a question of time and technology. So we built 32-bit version of FreeSWITCH. 5. This variant was successfully tested for SIP calls and execution of all extensions, whitch correctly works under Linux. 6. Then we set up support for E1/T1 Sangoma A-104. We found just this : - ftp://ftp.sangoma.com/Solaris/Beta/SVwanpipe-i386-5.10.pkg - ftp://ftp.sangoma.com/Solaris/Packages/SVzaptel-i386-5.10.pkg 7. We managed to configure and start that variant (zaptel + TDM Voice API). Complete equivalent of it successfully worked under Linux CentOS 5 — including configuration FreeSwitch mod_openzap. # ./wanrouter hwprobe ------------------------------- | Wanpipe Hardware Probe Info | ------------------------------- 1. AFT-A104-SH : SLOT=2 : BUS=2 : IRQ=5 : CPU=A : PORT=1 : HWEC=0 : V=33 2. AFT-A104-SH : SLOT=2 : BUS=2 : IRQ=5 : CPU=A : PORT=2 : HWEC=0 : V=33 3. AFT-A104-SH : SLOT=2 : BUS=2 : IRQ=5 : CPU=A : PORT=3 : HWEC=0 : V=33 4. AFT-A104-SH : SLOT=2 : BUS=2 : IRQ=5 : CPU=A : PORT=4 : HWEC=0 : V=33 Card Cnt: S508= 0 S514X= 0 S518= 0 A101-2= 0 A104= 1 A300= 0 A200= 0 A108= 0 # ./wanrouter status Devices currently active: wanpipe1 wanpipe2 wanpipe3 wanpipe4 Device name | Protocol | Station | Status | wanpipe4 | AFT TE1 | N/A | Connected | wanpipe3 | AFT TE1 | N/A | Connected | wanpipe2 | AFT TE1 | N/A | Connected | wanpipe1 | AFT TE1 | N/A | Connecting | I.e. E1 spans 4, 3, 2 connected with cables are perceived by Wanpipe in a synchronisation mode that is visible also on GREEN LEDs on the card. 8. But we do not manage to receive any reception PRI events (under FreeSwitch) that it is possible to receive in precisely same configuration in Linux. 9.After that we tested just transmition of raw B-channel data with our own tests, which has been created when we wrote Zaptel drivers for nonstandart E1 equipment. This tests pass test data sequences into B-channel and receives them from another B-channel, which connected to the first B-channel with crosscable; cheks up identity of the received information and measures delay of data passing on a loop. This tests works good under Linux (we use them for a year), but under Solaris they shows us that Zaptel + Wanpipe doesn't receives B-data from B-chans. We have source code for SVzaptel-i386-5.10.pkg , developed by SunLabs https://svn.sunlabs.com/svn/solaris-asterisk/zaptel-solaris/trunk/ Studying of this package and site has shown us such things : * SVzaptel-i386-5.10.pkg was developed by little community http://www.solarisvoip.com/; * development of this package moves very slow, last updates — 2006 year; * package has limited functionality (unlike original Zaptel ); * SVzaptel-i386-5.10.pkg was tested on a limited amount of E1/t1 cards (only one !) - Digium Wildcard TE110P T1/PRI). We doesn't have source code for SVwanpipe- i386-5.10.pkg , so we cann't say anything about it. Resume: Declared, for example: http://en.wikipedia.org/wiki/FreeSWITCH and other URLs, platform support is realy only for SIP-protocol features but : * this can be done without using of any PBX, for example with the help of SER/OpenSER or sofia library; * is grounded that into OS is present : a) C — compiler; b) NetBSD compatible IP-stack (in this case QNX 6.4 and many others OS could be included in the list of supported OS) We suppose same things under OS *BSD family. For objectivity it is necessary to underline that the same state of affairs takes place and in all others free PBX, realising support E1 through the Zaptel interface: Asterisk, YATE etc. And all declarations about «supported planforms» is only promotional declarations. Though it is tested and affirms at : http://www.thrallingpenguin.com/articles/asterisk-solaris.htm - that Solaris releases of PBX should have almost in 2 times the big productivity, than for Linux — that has it sense only for narrow case SIP connections that hold back testers and enter the others into error. Any info would be appreciated, Evgeniy.

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Happy Holidays
by Ken Rice 25 Dec '08

25 Dec '08

Merry Christmas and Chag orim same'ach Ken

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Lua script directory
by Woody Dickson 24 Dec '08

24 Dec '08

Hi, Is it possible to change the directory where freeswitch looks for .lua scripts? I would like to place the lua scripts in the shared drive so multiple freeswitch can refer to it. Thanks, Woody

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api_hangup_hook and bash
by Frank ＠ Impact 24 Dec '08

24 Dec '08

Can this command be used to run a bash script? I wanted to do some sox processing on some recordings after the bridge ends and thought I should use this command. But would like to do it in bash. Is there a better way? If this is the right way, what is the syntax for calling the bash script with some arguments?

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FS Dev Appreciation
by Michael Collins 24 Dec '08

24 Dec '08

FYI, If anyone would like to show their appreciation for the FS Dev team please email me off list so we can talk about options. I think we can all agree that Tony & Co. have worked very hard to make FreeSWITCH a success for all of us. Let's see if we can give a little love back to the core team! Thanks, Michael

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