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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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design of XML structure
by Seweryn Niemiec 24 Dec '08

24 Dec '08

Hi, Wy there is so many ugly constructions in FS configuration files? For example configuration of a gateway looks like this: <include> <gateway name="asterlink.com"> <param name="username" value="cluecon"/> <param name="realm" value="asterlink.com"/> <param name="from-user" value="cluecon"/> ... This is not good structure design. I know that it gives extreme flexibility for developers, but config files are for admins not software developers. IMHO it should look like this: <include> <gateway name="asterlink.com"> <username>cluecon</username/> <realm>asterlink.com</realm/> <from-user>cluecon</from-user/> ... Such structure + schema file would be a great help in configuration editing (autocompletion and syntax check). greetings, Seweryn

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another switch_ivr_set_user() can't find user
by John Wehle 23 Dec '08

23 Dec '08

> You don't have a default user in domain 192.168.14.10, in the default > config I used this so that you can set some vars on every call with Thanks for pointing it out and explaining the purpose. It looks like the domain is coming from set_domain in default.xml which gets it from sip_auth_realm. I guess the question is if force-register-domain is being used then: a) Should sip_auth_realm be set by FreeSWITCH to the value associated with force-register-domain b) or should set_domain in default.xml simply check for force-register-domain when setting domain? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Latest SVN trunk r10919 failed to build on OS X
by Chris Chen 23 Dec '08

23 Dec '08

Hi all, if you have the same problem as me failing to build the latest trunk on OS X 10.5.6 please see the error message below: making all in . Compiling src/switch_xml.c ... cc1: warnings being treated as errors src/switch_xml.c: In function 'switch_xml_find_child_multi': src/switch_xml.c:310: warning: 'value' may be used uninitialized in this functio n make[2]: *** [libfreeswitch_la-switch_xml.lo] Error 1 Making all in src Making all in mod making all mod_cdr_csv make[5]: *** No rule to make target `/Users/yunzhangchen/freeswitch/libfreeswitc h.la', needed by `mod_cdr_csv.so'. Stop. make[4]: *** [all] Error 1 make[3]: *** [mod_cdr_csv-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Thanks Chris

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Originate retry problem
by Alexandru Nedelcu 23 Dec '08

23 Dec '08

Hi, When I make a unsuccesfull call using session.originate, I'd like to have a 10 minutes pause and then try again. For our dialer we are using JS scripts, and setTimeout is not defined, session.execute("sleep",...) doesn't work because the session has to be originated first. And I don't really know what originate_retry_sleep_ms does. Basically I want a retry as described here, but with a delay between calls: http://wiki.freeswitch.org/wiki/Busy_Call_Retry

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