May 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

OSP Interop w/ Trans Nexus
by Gregory Boehnlein 02 Dec '09

02 Dec '09

Anyone know if Freeswitch can work w/ OSP and use something like Transnexus for CDR/Rating?

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Inboud Call Queue
by Saeed Ahmed 23 Aug '09

23 Aug '09

Hi All, In an inbound call center scenario is it possible that customers calls in and calls are distributed between online (who are registered on FS and in idle state) agents. I saw some on going discussion on list where it looks that currently it's not possible but I am newbie so maybe I didn't understand it well. If it's possible then please give me a start point that how can I implement it. Many Thanks

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ESL Wrapper
by Rex Alex 08 Aug '09

08 Aug '09

Hi All, I am new to freeswitch but installed(freeswitch version 1.0.3), configured and tested successfully. Now I want to do the dialling funtions through a php script. Read about Event Socket Library(ESL). How to implement the same in freeswitch. Please assist. Thanks, Rex

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Compact, fanless appliance?
by Fred-145 07 Aug '09

07 Aug '09

Hello For those SOHO customers (ie. at most, a couple of POTS/ISDN connections and simultaneous SIP calls) who'd rather not use a big, noisy PC to run Freeswitch, I'd like to offer an alternative that has the following features: - not old hardware sold on eBay, ie. it must be up-to-date hardware sold by a company currently in business - compact, silent - has room for a 2.5" hard-disk, but if not, must provide a CompactFlash plug - ideally, room for a PCI card, possibly laid down with a riser to save space - total cost (shipping + VAT) < 200 euro If it's cheaper and not much work, I don't mind buying the parts and putting the box together myself, but otherwise, I'd rather order a complete box, ready-to-use. What are my options to provide customers with that kind of solution? Thank you for any hint. -- View this message in context: http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23193738.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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how to enable ESL for ruby?
by dujinfang 29 Jul '09

29 Jul '09

Hi, Following the wiki: http://wiki.freeswitch.org/wiki/Event_Socket_Library On MacOSX 10.5, I can't get ESL for ruby work. make throws error: sevens-mac-pro:~/workspace/test/freeswitch/trunk/libs/esl$ make rubymod make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/Users/ seven/workspace/test/freeswitch/trunk/libs/esl/src/include - DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast- math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- prototypes -Wmissing-prototypes" CXXFLAGS="-I/Users/seven/workspace/ test/freeswitch/trunk/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb - I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C ruby g++ -I/Users/seven/workspace/test/freeswitch/trunk/libs/esl/src/ include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -I/opt/local/lib/ruby/1.8/i686-darwin9 -c esl_wrap.cpp -o esl_wrap.o g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -lruby -o ESL.so -L. Undefined symbols: "_main", referenced from: start in crt1.10.5.o ld: symbol(s) not found collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 Thanks for any help.

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setInputCallback not working with Javascript?
by mszlazak＠aol.com 24 Jul '09

24 Jul '09

I'm getting in build 12653M: [ERR] notify.js:130 mod_spidermonkey()? TypeError: session.setInputCallback is not a function The wiki says this function should work in Javascript. http://wiki.freeswitch.org/wiki/CoreSession_Constructor#session:setInputCal… Also, has there been changes to session.collectInput with type="event"? I get dtmf type events with my callback function but can't seem to get type="event" with speech events. Mark.

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Inbound call routing help
by Lars Zeb 23 Jul '09

23 Jul '09

I want to setup a dialplan for a single DID. I would like it to go to a specific extension, and if not picked up in 15 seconds, go to voicemail. I have set this scenario up and it works. But I would also like this person to be able to call this DID from outside FS via a phone and be able to retrieve their voicemail. I've seen the example of how to pick up an extension's voicemail while inside FS by checking to see if the destination_number is the same as the caller_id_number, and if so, listen to voicemail, otherwise leave the message with voicemail. But I don't have a clue how to accomplish this from outside, other than dedicating another DID to solely retrieving voicemail from outside. Any ideas? Thanks, Lars

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PCMU fallback for T.38
by Gabriel Kuri 01 Jul '09

01 Jul '09

hey folks, I'm trying to configure PCMU fallback for T.38. The originating endpoint (Linksys SPA-2102) sends an INVITE to FS with G729 and PCMU in the sdp. the INVITE to the provider includes G729 and PCMU as part of the sdp as well (absolute_codec_string=G729,PCMU) ... m=audio 16458 RTP/AVP 18 0 100 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 once the FAX tone is detected on the PSTN side, FS receives a T.38 re-INVITE from the provider and FS sends back a 488/Not Acceptable (proxy_media=false). at that point the provider than attempts fallback to PCMU with another reINVITE ... m=audio 16816 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 immediately after the PCMU reINVITE, FS closes the channel and the text below is in the FS logs. given the SPA-2102 included PCMU in the original INVITE, even though it was the second preferred codec, shouldn't FS fallback to using PCMU if it was re-INVITEd with PCMU by the provider? It seems like it's not passing the PCMU Re-INVITE back to the endpoint (SPA-2102), since it originally negotiated G729 with the SPA2102 as that was the 1st codec in the sdp, but trying to transcode between the two (G729 and PCMU)? 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[G729:18:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2371 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [telephone-event:101:8000]/[G729:18:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1550 sofia_glue_tech_set_codec() Changing Codec from G729 to PCMU 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1601 sofia_glue_tech_set_codec() Set Codec sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1 PCMU/8000 20 ms 160 samples 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1811 sofia_glue_activate_rtp() Audio params are unchanged for sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1. 2009-03-20 01:19:58 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() Processing Reinvite 2009-03-20 01:19:58 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1 entering state [completed] 2009-03-20 01:19:58 [DEBUG] switch_core_io.c:655 switch_core_session_write_frame() sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1 receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] 2009-03-20 01:19:58 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! 2009-03-20 01:19:58 [ERR] switch_core_io.c:723 switch_core_session_write_frame() Codec G.729 decoder error! Thanks, Gabe

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Linking problems with mod_portaudio.so module
by Jason White 27 Jun '09

27 Jun '09

I've already discussed this with a few members of the community, but I would like to raise it with a wider FreeSWITCH audience. Since upgrading to libtool 2.2.6a (now the default in debian testing), I can't successfully link mod_portaudio.so. The system is Debian Testing, x86_64 architecture. ldd -d mod_portaudio.so returns the following (after listing the library dependencies): undefined symbol: snd_config (./mod_portaudio.so) snd_config is from the Alsa library, libasound. mod_portaudio.la shows a reference to /usr/lib/libasound.la, all of which looks correct. I have also examined the build logs, without finding the problem, as there are no errors reported during linking and I didn't find any linker invocations with library parameters apparently missing. The ldd -d output: undefined symbol: snd_config (./mod_portaudio.so) linux-vdso.so.1 => (0x00007fffd31ff000) libm.so.6 => /lib/libm.so.6 (0x00007f0ccacf7000) libfreeswitch.so.1 => /opt/freeswitch/lib/libfreeswitch.so.1 (0x00007f0c ca8c6000) libpthread.so.0 => /lib/libpthread.so.0 (0x00007f0cca6aa000) libc.so.6 => /lib/libc.so.6 (0x00007f0cca357000) libncurses.so.5 => /lib/libncurses.so.5 (0x00007f0cca118000) libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f0cc9e08000) libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f0cc9bed000) libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f0cc998d000) /lib64/ld-linux-x86-64.so.2 (0x00007f0ccb1d1000) libdl.so.2 => /lib/libdl.so.2 (0x00007f0cc9788000) libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f0cc957f000) I spent several hours yesterday trying to debug this one. Any suggestions would be welcome. At this stage, I'm not sure whether it's a FreeSWITCH problem or a bug in libtool, or elsewhere.

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Re: [Freeswitch-users] FS & Wikipbx Help SOLVED
by FERNANDO VILLARROEL 09 Jun '09

09 Jun '09

Dear. Excuse some off topic of Wikipbx, but today i solved my problem; wikipbx and FS working now. Best regard. --- On Sun, 5/24/09, Jason White <jason(a)jasonjgw.net> wrote: > From: Jason White <jason(a)jasonjgw.net> > Subject: Re: [Freeswitch-users] FS & Wikipbx Help > To: freeswitch-users(a)lists.freeswitch.org > Date: Sunday, May 24, 2009, 11:17 PM > FERNANDO VILLARROEL <fvillarroel(a)yahoo.com> > wrote: > > > My install of wikipbx was succesfully, but i have > problem for registering a > > Softphone Xlite for testing; i look the next warning > in FS_CLI: > > > > freeswitch@internal> 2009-04-28 06:13:58 [WARNING] > sofia_reg.c:1701 > > sofia_reg_parse_auth() Can't find user > [223(a)192.168.1.150] You must define a > > domain called '192.168.1.150' in your directory and > add a user with the > > id="223" attribute and you must configure your device > to use the proper > > domain in it's authentication credentials. > > > > When i solve this problem i want make a gateway > between FS and Asterisk, for > > sending traffic from FS to Asterisk. > > > > If any could help me to resolve register ATA or > Softphones with wikipbx. > > If WikiPBX has its own mailing list, I think you would > receive better help > there. > > After installing FreeSWITCH, try registering to one of the > extensions (1000 to > 1019) provided in the default configuration. If that works, > which it should, > you should then try creating your own extension, or > modifying an existing one > as required. > > The error message you quote says very explicitly what is > wrong, but without > access to your FreeSWITCH configuration, it is impossible > to help any further, > except to suggest that you start with the default > configuration and work from > there. > > > _______________________________________________ > 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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