October 2009 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] FreeSWITCH Weekly Conference Call Starting, Please Join Us!
by Michael Collins 09 Oct '09

09 Oct '09

Agenda and call-in info: http://bit.ly/lzEYy

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[Freeswitch-dev] Loading mod_sofia from Erlang -- good news and bad news
by Mark Sobkow 08 Oct '09

08 Oct '09

The good news is I was able to download the ppa-source from launchpad.net and do a local source build of the packages. I was then able to get a running install via: dpkg -i freeswitch_1.0.4*.deb freeswitch-asr*.deb freeswitch-codec-*.deb freeswitch-lang-en*.deb freeswitch-lua*.deb freeswitch-perl*.deb freeswitch-python*.deb freeswitch-sky*.deb freeswitch-spider*.deb (Installing all the .debs breaks the install as the Russian sound files aren't available, and there seems to be an issue with the .deb properly _blocking_ the install until those dependencies are resolved.) However, I'm still having the same problem where fs_cli locks up after trying to do a "load mod_sofia". Our Erlang logs show that it's been sent back in response to the query from FS, but FS freezes or does not get the file contents. Is there some sort of debug option I can set in fs_cli to see exactly what's happening? I'd like to confirm that it's _receiving_ the XML back from Erlang, especially as attempts by Andrew to use our Erlang source were successful on his installation. Clearly there is still some difference between our boxen. BTW, we're running Erlang ERTS 5.5.5. -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow(a)marketelsystems.com Web: http://www.marketelsystems.com

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[Freeswitch-dev] REMINDER: Weekly FreeSWITCH Conference Scheduled for October 9, 11AM CST (GMT -6)
by Michael Collins 08 Oct '09

08 Oct '09

Just a friendly reminder that we will be having the weekly FreeSWITCH conference call tomorrow, October 9th, at 11AM Central. The agenda is updated: http://bit.ly/lzEYy The conference lasts for six hours, so feel free to dial in at any time. We usually start going over the agenda about 15 minutes after the conference starts. The agenda itself only takes about an hour or two, after which we spend most of the time discussing whatever is on the minds of the community members. Remember, the core FreeSWITCH development team of Anthony Minessale (primary author, lead architect), Mike Jerris (build master), and Brian West (all around VoIP expert, FS configuration guru) are present for the whole conference. This is a great time to ask questions about the project. Please join us! Call in options include SIP, Skype, PSTN, and Jabber. Please see the agenda for more information. Looking forward to speaking with you all tomorrow, Michael

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[Freeswitch-dev] FreeSWITCH Weekly Conference Call Followup, Agenda For This Week
by Michael Collins 07 Oct '09

07 Oct '09

Ladies and Gentlemen, Thank you for calling in to the weekly FreeSWITCH conference call. Last week's agenda was rather light, so if you have things that you would like to have discussed please be sure to add them here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_09 Here are a few updates for everyone to keep in mind: * Starting with the upcoming meeting (Oct 9) the conference will support 48kHz CELT codec. * When you call in, please mute your phone if you are just listening or if you will not be speaking a lot. Background noise from one user isn't usually too bad, but when 10 or 15 people are not muted it can get a little distracting. :) * We are always looking for people to help out. If you are looking for ways to help out then please by all means call the conference. If you cannot call the conference for some reason but still want to help out, please email me off list. We have numerous documentation, janitorial, code review, etc. sub-projects that FreeSWITCH users can help with. * Be sure to update to latest SVN trunk and test test test! We are prepping for 1.0.5 and more people testing means a more stable release delivered more quickly. One topic that came up was the testing of Mike van Lammerman's Ultramonkey setup. (http://bit.ly/8zZUS) Several people agreed to try it out. We would love to see others try it out and report back their experiences. Lastly, we'd like to thank everyone who has been helping out with testing, documentation, and answering questions on the email list and IRC channels. A special thanks to Diego Viola for doing lots of wiki cleanup lately. Everyone's efforts are appreciated. -Michael

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[Freeswitch-dev] upnp
by vip killa 06 Oct '09

06 Oct '09

could someone tell me why freeswitch does not detect nat upnp? last week it was detecting it, but now it won't. what could possibly cause this to happen?

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[Freeswitch-dev] call is dropped after answer
by Francisco Scaramanga 05 Oct '09

05 Oct '09

I wrote my own new FreeSwitch module "mod_skel" and successfully created an application interface with: SWITCH_ADD_APP(app_interface, "my_mod_skel", "Modul skel", "test", mod_skel_function, "no Syntax", SAF_NONE); The dialplan is <extension name="sip_test"> <condition field="destination_number" expression="^54322$"> <action application="my_mod_skel" data=""/> </condition> </extension> In the function mod_skel_function I do the following to answer the call. SWITCH_STANDARD_APP(mod_skel_function) { switch_channel_t *channel = switch_core_session_get_channel(session); switch_channel_clear_flag(channel, CF_PROXY_MEDIA); switch_channel_clear_flag(channel, CF_PROXY_MODE); switch_channel_answer(channel); } The problem is that sofia hungs up immediatley (see logfile below). How can I keep the call alive? How can I pick up a new incoming call and start a conference? 2009-10-04 21:13:01.827000 [NOTICE] mod_sofia.c:1503 Pre-Answer sofia/internal/1 002(a)192.168.1.33! 2009-10-04 21:13:06.795000 [NOTICE] mod_skel.c:121 Channel [sofia/internal/1002@ 192.168.1.33] has been answered 2009-10-04 21:13:06.803000 [NOTICE] switch_core_state_machine.c:179 Hangup sofia /internal/1002(a)192.168.1.33 [CS_EXECUTE] [NORMAL_CLEARING] 2009-10-04 21:13:08.229000 [NOTICE] switch_core_session.c:1087 Session 1 (sofia/ internal/1002(a)192.168.1.33) Ended 2009-10-04 21:13:08.229000 [NOTICE] switch_core_session.c:1089 Close Channel sof ia/internal/1002(a)192.168.1.33 [CS_DESTROY] _________________________________________________________________ http://redirect.gimas.net/?n=M0910xHotmail2 Neu, Besser, Hotmail - Die neuen Features sind da!

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[Freeswitch-dev] how to connect a dialplan with my own module
by Francisco Scaramanga 05 Oct '09

05 Oct '09

I want to create my own freeswitch module which should simply answer a call and write the audiostream into a file. Here is the dialplan: <extension name="my new module"> <condition field="destination_number" expression="^54321$"> <action application="mod_myModul" data=""/> </condition> </extension> The question is how the dialplan entry (application="mod_myModule") gets associated with mod_myModul so that the dialplan actually executes mod_myModul? I need to understand the mechanism in general but I can't find useful documentation for that. _________________________________________________________________ http://redirect.gimas.net/?n=M0910xWin72 Schneller, einfacher und noch mehr Fun – mit Windows 7

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[Freeswitch-dev] Eclipse Project
by Jerry Richards 02 Oct '09

02 Oct '09

Hello All, I am running FS on a Centos 5.3 computer. Is there an Eclipse project to build FS and step through the code? Best Regards, Jerry

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[Freeswitch-dev] FreeSWITCH Weekly Conference Starting, Please Call In!
by Michael Collins 02 Oct '09

02 Oct '09

Hey folks, the weekly conference call is starting. Please see the agenda for instructions on dialing: http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_02 Looking forward to speaking with you all! -Michael

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[Freeswitch-dev] switch_core_media_bug_add and some related questions
by Leon de Rooij 01 Oct '09

01 Oct '09

Hi all, I'm trying to make a legal intercept module. My first attempt was to get the raw RTP packets from switch_frame_t, but they don't seem to be available. My second attempt is to get the raw pcm data from the frames, which seems to work. I still have a few questions though: - On setting the mediabug (switch_core_media_bug_add), I can pass an enum. Is there a difference between SMBF_BOTH (0) or SMBF_READ_STREAM | SMBF_WRITE_STREAM (1+2=3) ? - In the callback function passed to the bug, what does each of the following switch_abc_type_t mean ? At first I assumed that READ and WRITE state the direction of the media, but some recording functions only use the READ so that can't be true. SWITCH_ABC_TYPE_INIT SWITCH_ABC_TYPE_READ_PING SWITCH_ABC_TYPE_CLOSE SWITCH_ABC_TYPE_READ SWITCH_ABC_TYPE_WRITE SWITCH_ABC_TYPE_READ_REPLACE SWITCH_ABC_TYPE_WRITE_REPLACE - When the callback function is called with SWITCH_ABC_TYPE_READ, I'm doing a switch_core_media_bug_read to copy all media to a switch_frame_t. To then get the raw data, I see in many examples that the size of the frame is retrieved by "(switch_size_t) frame.datalen / 2". Why the division by two ? - I thought that frame.data contained a un-encoded version of the RTP packet, but the datalen seems to differ when I'm calling in over g711 or g722. What format is frame.data ? Thanks a lot ! regards, Leon

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