April 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Community Thank Yous
by Michael Collins 25 Apr '11

25 Apr '11

Happy Monday to you all! I just wanted to take a moment to say thank you to the many folks who have been stepping up to help out with the not-so-glamorous side of running an open source project: documentation and community support. We all know that hacking code is the fun part, but writing documentation and helping out with IRC and mailing list questions are often thankless tasks. I wanted to personally say thank you to all those who've been stepping up. Regarding documentation, I've seen dozens of new users on the wiki, each of whom is adding something. This is indicative of people "paying the wiki tax" when a question is answered via ML or IRC. So to those who have been adding knowledge to the wiki: thank you! It is much appreciated. Please keep up the good work. In the IRC channel we've been having a good number of community members come in and make themselves available for answering questions. This can be very time-consuming, so again we say thank you. The same thing goes for the mailing list. It takes a lot of time to read through all the posts and respond. The following users have been especially active in helping others of late and I feel it is only fair to acknowledge their fine efforts. In no particular order... Ken Rice Steven Ayre Jeff Link Avi Marcus Kristian Kielhofner Steve Underwood Peter Olsson Stefan Knobloch Giovanni Maruzzelli Christopher Rienzo Every time a community member helps another community member we grow stronger. Please keep up the good work! -Michael

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conference_set_auto_outcall No Audio
by Chris Monkey 25 Apr '11

25 Apr '11

Hi, I am am relatively new to FS and have setup a home PBX with about 5 Snom 320s with TLS. Everything works fine, except when I try to use the mad_boss example, or any auto_outcall conference, the phones ring and will auto answer, but there is no audio. However, if I dial in to the conference, everything works fine. I've tried adding <action application="set" data="sip_secure_media=true"/> to the conference extension, but to no avail. I don't see any errors in the log, and I just did a make clean this afternoon. What am I missing? Thanks

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mod_enum compilation error
by mazilo 25 Apr '11

25 Apr '11

I have a local FS git pulled to adffe07f7512b00f7093091778965a119ca81513 and failed to compile mod_enum for OpenWRT as shown below: making all mod_enum make[7]: Entering directory `/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/mod/applications/mod_enum' make[7]: *** No rule to make target `Makefile.am', needed by `Makefile.in'. Stop. make[7]: Leaving directory `/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/mod/applications/mod_enum' make[6]: *** [mod_enum-all] Error 1 make[6]: Leaving directory `/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/mod' make[5]: *** [all-recursive] Error 1 make[5]: Leaving directory `/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src' make[4]: *** [all-recursive] Error 1 make[4]: Leaving directory `/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git' make[3]: *** [all] Error 2 make[3]: Leaving directory `/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git' make[2]: *** [/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/.built] Error 2 make[2]: Leaving directory `/opt/tmp/OpenWRT/feeds/packages/net/freeswitch_git' make[1]: *** [package/feeds/local/freeswitch_git/compile] Error 2 make[1]: Leaving directory `/opt/tmp/openwrt-svn-trunk' make: *** [package/freeswitch_git/compile] Error 2 ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-enum-compilation-error-tp… Sent from the freeswitch-users mailing list archive at Nabble.com.

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mod_radius_cdr
by Camila Troncoso 25 Apr '11

25 Apr '11

Hi, I´m working with radius accounting using the mod_radius_cdr module. I´m having trouble with the date format that Radius server receive. An example of what the radius server receive is: “Freeswitch-Callstartdate = "2011-04-21T18:23:59.780945-0400"” This date format is very difficult to read and I want to change it to make the accounting easier. I search all around for some param or configuration that allows me to do so, but I only find that this date format is define in mod_radius_cdr.c module. Please some help with this matter. Regards, *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

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Эхо на Белоруссию
by Boris Kovalenko 25 Apr '11

25 Apr '11

Добрый день! 25.04.2011 20:26:06 +375292718781 - на удаленном конце (в Белоруссии) слышали собственное эхо. -- С уважением, Борис Коваленко ЗАО "Тагнет" тел. +7 (3435) 230001 факс +7 (3435) 230005

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Can FreeSWITCH play MoH from remote server?
by Иван Чистяков 25 Apr '11

25 Apr '11

Can FreeSWITCH play MoH from remote server?

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g729 annexa/annexb interoperability
by Javier Gallart 25 Apr '11

25 Apr '11

Hello all we're using a Sangoma D100 card for transcoding. Our configuration, as far as the codec policy is concerned, basically does this: -In the first offer to the b-leg we use the same codec list we receive (<action application="export" data="absolute_codec_string=${ep_codec_string}"/>) -If the call fails with status 488, we repeat the call using all the codecs available. This includes changing from annexb to annexa and viceversa. Let's suppose a-leg only supports g729 annexA, and B-leg only supports G729 annexB. My understanding is that those different codec flavours don't interoperate; in fact I've experienced many times audio problems when trying to set up a call between endpoints supporting different g729 variants. In the described case, FS sends initially annexb=no, and B-leg rejects it with cause 488. We rebuild the offer with annexb=yes, and then the offer is accepted by the B-leg. Our concern is that no transcoding resources are used in this case, and we might run into audio problems because of that. The other concern is that in the answer to the A-leg (in 183 and 200), annexb=yes is included. I'm not sure if all devices would support different fmtp parameters in the offer and the answer (the RFC, as usual, won't be explicit about this). Is there any way to force the transcoding in a situation like the one I described? Thanks in advance Javi

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No Dial Tone ... Zaptel + FreeSwitch + TDM400P
by Pablo Silva 25 Apr '11

25 Apr '11

Hi! I've tryed for severals hours to make a phone call from freeswitch server, using TDM400P. I don't know why i can't hear the dial tone, and it's impossible make a phone call.., i read freeswitch wiki page about TDM400P configuration, as you can see that's my configuration an log http://pastebin.com/Y94UqN5R if I focus on information from the log, you can see that loads openzap, the server recognizes the TDM400P card, the call is routed to the PSTN but no concrete achievements. When you call, I hear the tone, not the dialing, I guess I'm near the solution to my problem, I request your cooperation. In advance, thanks. -Pablo

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Build and install FS with new customized folder on CentOS 5.5
by fieldpeak 25 Apr '11

25 Apr '11

i added a folders(e.g. new_folder) inside the FS source code folder, inside the new_folder there are some files, like below, ./freeswitch/conf ./freeswitch/bin ./freeswitch/*new_folder*/files1, files2... i need this folder and the files under it will be installed when i run 'make install', however, it did not. the commands i run as below, i assume bootstrap.sh will run autoscan which will find my new_folder... ./bootstrap.sh make make install anyone can help how to realize it? thanks. Regards, Charles

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HOLD - signals from phones? How to play MoH?
by Иван Чистяков 25 Apr '11

25 Apr '11

In a bridge mode, when i push a button hold, FreeSWITCH don't playing MoH. But if I set hold through FS console it's works.

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