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mod_managed loader loadfile
by Diego Toro 22 Apr '09

22 Apr '09

I have assembly that references others assembly in my managed AppFunction, when FS start it fail with error message: can't load reference assembly... I checked Loader code and i changed next line: System.Reflection.Assembly.LoadFile(f) in loadAssemblies(string managedDir) method by System.Reflection.Assembly.LoadFrom(f); (LoadFile by LoadFrom) and FS start with no problem. Is possible use LoadFrom instead of LoadFile in Loader.cs code ? Information about LoadFrom and LoadFile: http://blogs.msdn.com/suzcook/archive/2003/09/19/loadfile-vs-loadfrom.aspx

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DialPlan and Mod_Managed
by Robert Clayton 22 Apr '09

22 Apr '09

All, I am sure this is a simple error, but in using mod_managed to call the Demo.cs class AppDemo, what is improper with this dialplan entry? <extension name="managed_app"> <condition field="destination_number" expression="^9009$"> <action application="managed" data="AppDemo"/> </condition> </extension> This seems to match the documentation but has a routing problem. 2009-04-22 00:34:03 [INFO] switch_core_state_machine.c:136 switch_core_standard_on_routing() No Route, Aborting Are there any working contributions? Bob

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FIFO call park has about 2 seconds delay
by Scott Shen 21 Apr '09

21 Apr '09

<extension name="test1"> <condition field="destination_number" expression="^50000$"> <action application="fifo" data="myqueue in"/> </condition> </extension> <extension name="test2"> <condition field="destination_number" expression="^60000$"> <action application="fifo" data="myqueue out nowait"/> </condition> </extension> user 101 dials into 50000, and user 102 dials into 60000, no problem, they get connected. Now, log into Event Socket, and type "bgapi uuid_park whatever_the_user_101_uuid_get_from_show_channels" User 102 gets disconnected only after about 2 seconds, Why is this delay? How to fix it? BTW, if user 101 directly connects to 102 ( eg. 101 dials 102 directly ), then the same uuid_park happens immediately Thanks & Regards Scott

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monitor_exec functionality ?
by Frank ＠ Impact 17 Apr '09

17 Apr '09

Is there any equivalent functionality to Asterisk's monitor_exec that we can use with record_session in the xml dial plan? Some post call processing of the recorded call files after the call is down and the files are closed?

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how many simultaneous calls support in freeswitch
by Parveen Kumar Jain 15 Apr '09

15 Apr '09

Hi Developers, I need to develop an IVR application which makes an outbound calls and then plays the some audio file for the user. For this I was trying to evaluate Freeswitch under following criteria: - Does freeswitch have outbound calls support(is there any conf file file avilabel where I just can put some series of no. and freeswitch just calls those no. sequentially)? - If yes, how many simultaneous calls are possible on a simple pentium-4 using G711U as a codec(1 GB RAM, 2.2 GHz machine) machine?(I had checked the switch.conf file and it says that by default it can support upto 1000 calls , is it true for a small machine also)? In other terms is Freeswitch is scalable if I need to add more users to call from here? - Does freeswitch have the support of CDR(call data record) after succesfull calls ? It will be great help if any of you can comment on these question and it can save my several hours of testing before I can make my conclusion :) Best Regards, Parveen Jain

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Entire Wiki.FreeSwitch.org on Single PDF ?
by Mitul Limbani 15 Apr '09

15 Apr '09

BODY { font-family:Arial, Helvetica, sans-serif;font-size:12px; }Hello there, In my previous encounter with FreeSwitch, I had found that Bret had posted on the Mailing List somewhere about availability of the entire FreeSwitch Wiki Documentation on a single PDF, this is useful coz at the offset apart from Wiki there is no other offline media to learn it. Is the same PDF available looking at the growth of Wiki pages and the updation. I look forward to hear from you guys, Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions, The Enterprise Linux Company (TM), www.enterux.com +91-9820332422 ------------------------- Msg sent via Enterux Enterprise Email Server : http://www.enterux.com/

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ANNOUNCEMENT: FreeSWITCH 1.0.4pre4 Now Available
by Michael Collins 14 Apr '09

14 Apr '09

The FreeSWITCH team is pleased to announce the immediate availability of version 1.0.4pre4. Details are available here: http://www.freeswitch.org/node/173 All are encouraged to upgrade as soon as possible. Thanks to everyone for their feedback, ideas, and bug reports. Please keep them coming. -Michael http://www.cluecon.com

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Fwd: [Freeswitch-users] codecs initialization flags in endpoint modules
by Matteo 13 Apr '09

13 Apr '09

Hello, I'm forwarding the message from my friend since is better suited for -dev ML. In addition, we changed the codec_init to ask only for DECODE on read and for ENCODE on write in sofia and skypiax and everything seems to work... there's a reason for that? are we going to break up things? regards, Matteo. ----- Messaggio inoltrato ----- Da: "Lele Forzani" <lele(a)windmill.it> A: freeswitch-users(a)lists.freeswitch.org Inviato: Venerdì, 3 aprile 2009 15:20:13 GMT +01:00 Amsterdam/Berlino/Berna/Roma/Stoccolma/Vienna Oggetto: [Freeswitch-users] codecs initialization flags in endpoint modules Hello, I've been experimenting with the use of mod_dahdi_codec and other ways to perform external transcoding for codecs, and came up with noticing that transcoding resources seemed to be used up twice what I expected. That is and 2x the number of call legs, ending up to two encoder and two decoder instances per leg. So, I looked at the code and noticed almost every endpoint module does something like this (excerpt from mod_sofia, sofia_glue.c:~1800): if (switch_core_codec_init(&tech_pvt->read_codec, tech_pvt->iananame, tech_pvt->rm_fmtp, tech_pvt->rm_rate, tech_pvt->codec_ms, 1, SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE | tech_pvt->profile->codec_flags, NULL, switch_core_session_get_pool(tech_pvt->session)) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Can't load codec?\n"); switch_goto_status(SWITCH_STATUS_FALSE, end); } if (switch_core_codec_init(&tech_pvt->write_codec, tech_pvt->iananame, tech_pvt->rm_fmtp, tech_pvt->rm_rate, tech_pvt->codec_ms, 1, SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE | tech_pvt->profile->codec_flags, NULL, switch_core_session_get_pool(tech_pvt->session)) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Can't load codec?\n"); switch_goto_status(SWITCH_STATUS_FALSE, end); } The flags being SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE seems to be causing the apparent 'double' allocation of transcoding resources, and I fail to understand the need for both, in both cases. Could someone please spend a minute to explain? thanks lele _______________________________________________ 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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setting profile->caller_id_name
by Rupa Schomaker 11 Apr '09

11 Apr '09

One normally sets caller_id_name after the fact by setting effective_caller_id variable. This works great when bridging to legb but if I want applications like: show calls show channels conference list to show the updated caller_id_name I have to set the variable myself. Am I asking for trouble if I simply do a profile->caller_id_name = newname; ? It seems to work fine for me, but.... is it acceptable? -- -Rupa

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GDB doesn't catching breakpoints in module
by Ceino 08 Apr '09

08 Apr '09

Hi all, I try to use breakpoints in mod_dialplan_xml.c but it doesn't get catched (the program work). This did work in fs 1.0.3. I use fs svn thrunk 12914 and GDB version 6.6. Inside gdb I use the following command: break mod_dialplan_xml.c:dialplan_hunt Best Regars Lars Sivertsen

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