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Few question regarding move from Asterisk to FS - resend
by Ivica Samija 23 Jan '09

23 Jan '09

Last message was incomplete, sorry for that, resending. Hi all, our company have implemented two Asterisk servers to: - connect two company sites - transition to IP telephony - cut down TCO regarding telephony Our interconnection schema: --T1/E1 provider1--< > --T1/E1 provider2--< Asterisk1 >--T1/E1/ trunk--< propriety PBX1 > ---SIP provider3---< > | SIP trunk | < Asterisk2 >--T1/E1/ trunk--< propriety PBX2 > On each site we have number of IP phones connected to Asterisk and analog phones connected to propriety PBX. Features implemented on Asterisk boxes are: - IVR - queue - conference - DISA - BLF - transfer calls All was more or less good until we had to implement call rating (we have to keep track cost made on each extension for statistic). Company policy is that implementation has to be in house. We hit brick wall because Asterisk have inaccurate CDRs (transfers, forwards,...) I am looking in FS for last few weeks and it seams to me that it can replace our Asterisk boxes, and more :). I am a little confused with XML config but it seams to me that it is worth of learning. For most of my question I have found answers in documentation wiki and on list archive, but I have just a few question still without answer. I am sorry if the answers are out there but I was to clumsy to find them. In that case some info or link would be great. So here we go: 1) OpenZap is stable enough that it can be used in production? As you can see we depend on 4 zap trunks. We use OpenVox T1/E1 cards (D210P and D410P) with wct4xxp module. We use ccs, hdb3, crc4, loadzone=it in zaptel.conf We use switchtype=euroisdn, pridialplan=international, pridialplan=unknown, pri_cpe and pri_net signaling. Any suggestions are welcome. 2) Is it possible do bridge to Zap group instead of channel something like Dial(Zap/g4/${EXTEN:1},60,t)? 3) If I understood correct I can talk with SQLite (or other DB) only through Lua, Javascript,... there is nothing similar to Set(mobile=${DB(mob/${EXTEN:1})}) 4) mod_nibblebill, is it possible to get resulting cost per channel in some channel variable and than put value in custom cdr field Sorry on bad english, not a native speaker. Best regards, Ivica

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Few question regarding move from Asterisk to FS
by Ivica Samija 23 Jan '09

23 Jan '09

Hi all, our company have implemented two Asterisk servers to: - connect two company sites - transition to IP telephony - cut down TCO regarding telephony Our interconnection schema: --T1/E1 provider1--< > --T1/E1 provider2--< Asterisk1 >--T1/E1/ trunk--< propriety PBX1 > ---SIP provider3---< > | SIP trunk | < Asterisk2 >--T1/E1/ trunk--< propriety PBX2 > On each site we have number of IP phones connected to Asterisk and analog phones connected to propriety PBX. Features implemented on Asterisk boxes are: - IVR - queue - conference - DISA - BLF - transfer calls All was more or less good until we had to implement call rating (we have to keep track cost made on each extension for statistic). Company policy is that implementation has to be in house. We hit brick wall because Asterisk have inaccurate CDRs (transfers, forwards,...) I am looking in FS for last few weeks and it seams to me that it can replace our Asterisk boxes, and more :). I am a little confused with XML config but it seams to me that it is worth of learning. For most of my question I have found answers in documentation wiki and on list archive, but I have just a few question still without answer. I am sorry if the answers are out there but I was to clumsy to find them. In that case some info or link woud be great. So here we go: 1) OpenZap is stable enough that it can be used in production? As you can see we depend on 4 zap trunks.

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Conference and socket outbound
by Dennis 23 Jan '09

23 Jan '09

hi, now, that everything is working, what i wanted to do with freeswitch, i want to look a little bit more left and right. one interesting thing to "play" arround with, is conferencing. is it possible to do conference calls, when using socket outbound and php? where could i start? ok, of course with "mod conference" in the wiki. but i wonder, how i can put a caller in a conference room and how to set the properties (what dtmf does what and which soundfiles are played when) dynamically. let's say i get a socket inbound call and i answer it. then i play a soundfile, which asks to enter the pin... how can i move the inbound call or originated calls into a conference room? it seems, as if one can do a lot with api in the conference room. but i wonder what i have to do in the beginning, to put people (specific uuid's) into a conference room. thanks, dennis

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voicemail web interface
by Helmut Kuper 23 Jan '09

23 Jan '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I tried to get voicemail web interface up and running. I found that we need xml_rpc to open the 8080 port of FS. Then USer-profile needs "http-allowed-api=voicemail" parameter. After that I left a message for my account and tried to access my voicemails via web (I replaced fs.ip with my IP): http://fs.ip:8080/api/voicemail/web This led to a web page generating out of web-vm.tpl with 0 (zero) voicemails. If I call my voicebox to listen to the messages, there is one new message available. My phone gets a mwi saying there is 1 new message for me. I wonder how the page knows for whom it shall display the voicemails. After that I tried "http://fs.ip:8080/api/domains/this/api/voicemail/web" This delivered: Error 404 Not Found ABYSS Web Server for XML-RPC For C/C++ version Xmlrpc-c 1.14.99 Hm, any ideas ? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkl4m2gACgkQ4tZeNddg3dzIogCgnZybNeIkfsFbyKzSmusgUP5b hc8AoLYYO8Fi5RHKLEk7eXxUfzj+udIW =ofO3 -----END PGP SIGNATURE-----

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freeswitch memory leak issue
by Simon Leck 23 Jan '09

23 Jan '09

Hi All, I am experiencing memory leak issue in FreeSwitch? Anybody knows how this can be resolved, please email me. Thanks in advance to everybody for your kind assistance. Thanks Simon

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mod_radius_cdr questions and thoughts
by Apostolos Pantsiopoulos 23 Jan '09

23 Jan '09

I am trying to implement a radius based solution using FS. I have seen that the mod_radius_cdr module is actively maintained. so I have a few questions/remarks : 1) When I place a call and my radius server is down, the call blocks forever instead of just radius_timeout * radius_retries seconds (I have declared only one server). I would expect that FS would stop trying to send an Acc Start packet after some time and get on with the call. 2) I have also noticed that FS sends only 1 packet (I waited for a minute) instead of 3 (default in the config) since the first (and second) attempt failed. If my server was up (the port was responding) but it returned a req. failed answer would the above time-out be valid? 3) When I tried to load the dictionary.freeswitch to my freeradius server, it complained : "Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary.freeswitch[33]: dict_addattr: Duplicate attribute name NAS-Port-Id" How can I overcome this? When I "instructed" my freeradius to only load the freesitch dictionary I got this : "Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary.freeswitch[257]: unknown option "Freeswitch"" How can I use the dictionary with the freeradius? (or other radius servers?) 4) The radius attributes included in the current requests are a) hard-coded, b) limited in number. I think many of us would like to use more attributes. Or even better define what to include (and what to put in them) using a config file (the same maybe?) 5) Does the module send accounting packets only for the a-leg of a call or for both legs? (Maybe that could be configurable too). If anyone is interested in the above questions/remarks please post a reply. I would really like to know how many of the mailing list users are also interested in FS radius support and your opinions on the matter. Regards, -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs(a)kinetix.gr -------------------------------------------

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Alternate config sets.
by Brian West 22 Jan '09

22 Jan '09

FreeSWITCHers, I have started to put alternate config examples up for people to check out. Right now it only has the Softphone example... if you have ideas or examples to add to this please let me know: http://svn.freeswitch.org/svn/configs/softphone/ Thanks, Brian West FreeSWITCH.org

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Re: [Freeswitch-users] Recording ULAW files
by freeswitch-users＠digitaldan.com 22 Jan '09

22 Jan '09

Thanks for the quick reply. I'm new to this project so I'm not familiar with the inner workings just yet but at looking at mod_native_file.c it seems this is a thin wrapper around the switch's own file input and output routines? Would it be best to change this class or register a new file type, like .ul? If so, where would be a good starting point. ----- Original Message ----- From: "Anthony Minessale" <anthony.minessale(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, January 21, 2009 10:56:45 AM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Recording ULAW files no, there is currently no way to do that. It would be feasible to add an option to mod_native_file to write wav headers around the raw data but it has not been attempted. On Wed, Jan 21, 2009 at 10:48 AM, < freeswitch-users(a)digitaldan.com > wrote: Hi, I'm recording files using the pcmu extension in order to save them in the g.711 ulaw format, which is what everything in my network uses. It appears that the recorded file is just raw data without a header. Is there any way to save this as a wav type with a header (keeping the ulaw format)? for example, running the unix command 'file' on the recording prints: /tmp/185065_f7bb8e0c-e641-11dd-800d-5ffe41c540dd.PCMU: data if I run it through sox which just outputs the same data (ulaw,8000,mono) but specifying the type as wav /usr/bin/sox -t .ul -r 8000 -c 1 -b -U file.pcmu -t wav -r 8000 -c 1 -b -U file.wav it produces a file that shows: /tmp/185065_f7bb8e0c-e641-11dd-800d-5ffe41c540dd.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law, mono 8000 Hz if I try and save it within freeswitch using the wav extension, it trans-codes it to a pcm format RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz Is there any way to have freeswitch record the file as ulaw with the RIFF wav header? Thanks! _______________________________________________ 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 FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ 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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Error: eval_some_python() Error importing module
by Brian Deacon 22 Jan '09

22 Jan '09

Greetings, Couldn't find anything on the wiki or in the mail archives. (Let me know where you think a good home for this info might be on the wiki and I'd be more than happy to write something up in there.) I'm guessing I haven't done everything necessary to enable python on my machine. I have python-2.4.3 and python-devel 2.4.3-21 installed on the FS machine. Per the instructions, I uncommented the mod_python line from modules.conf and rebuilt my freeswitch instance. Vanilla functionality is working. I set PYTHONPATH to /usr/local/freeswitch/python before restarting the mod_python-enabled freeswitch (via modules.conf.xml) # echo $PYTHONPATH /usr/local/freeswitch/python # grep mod_python \ > /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml <load module="mod_python"/> # ls $PYTHONPATH bar.py bar.pyc # cat $PYTHONPATH/bar.py stream.write("baz") The relevant entry in conf/dialplan/default.xml: <extension name="Test_Simple_Python"> <condition field="destination_number" expression="^1235$"> <action application="set" data="foo=${python(bar)}" /> <action application="bridge" data="{group_confirm_file=vm-hello,group_confirm_key=4,call_timeout=60} sofia/internal/1000,sofia/internal/1002" /> </condition> </extension> (The bridge action is only there because it seemed unhappy unless it was going to actually try to do something.) The following output shows up in fs_cli when I dial 1235: > 2009-01-21 15:20:14 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/1000(a)10.48.5.207 Execute set(foo=${python(bar)}) > 2009-01-21 15:20:14 [NOTICE] mod_python.c:107 eval_some_python() Invoking py module: bar > 2009-01-21 15:20:14 [ERR] mod_python.c:121 eval_some_python() Error importing module > 2009-01-21 15:20:14 [DEBUG] switch_core_session.c:1254 switch_core_session_execute_application() sofia/internal/1000(a)10.48.5.207 Expanded String set(foo=) > 2009-01-21 15:20:14 [DEBUG] mod_dptools.c:699 set_function() sofia/internal/1000(a)10.48.5.207 SET [foo]=[UNDEF] So I'm guessing something dumb on my part. But there are so many dumb things I'm capable of doing... :) TIA, Brian

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Dialplan Question: Is extension local?
by Helmut Kuper 22 Jan '09

22 Jan '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, is there a dialan or dptool function, which allows to check whether a user/extension is local or not ? What I try to do is this: Local extension? -> bridge -> bridge fails (on_busy, no answere, timeout, offline)->voicemail Every other call should go to PSTN. But unfortunately calls which should go to PSTN goes to voicemail ... I use default dialplan for local extensions which ships with FS regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkl4RNsACgkQ4tZeNddg3dz26gCdHbAbrp0Z+YzIuVTewo+5tE7K GJcAniMGvoIE6yLCJJ3auVSJtKW/fma4 =5eQI -----END PGP SIGNATURE-----

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