June 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Weekly Conference Call -- June 04 2008
by Raymond Chandler 03 Jun '08

03 Jun '08

Here is what we have for the agenda for the weekly conference call. Please feel free to add to what's there http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_04 We look forward to hearing from everyone tomorrow at 1PM Central Standard Time (US ) 6PM GMT (i think) * 1 we still need to finish adopting the orphaned pages <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_04#we_still_need_to_finis…> * 2 there are still quite a few dead-end pages that need linkbacks <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_04#there_are_still_quite_…> * 3 the default config page is still lacking some info <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_04#the_default_config_pag…> o 3.1 I think we should have a list of mods and files that are included in the default config and link them to pages that explain them. <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_04#I_think_we_should_have….> o 3.2 the mod pages should have the list of apps that they include and what they do also. <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_04#the_mod_pages_should_h….> o 3.3 the mod pages should probably also have a list of params that affect the way they behave and links to the files in which to add them <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_04#the_mod_pages_should_p…> * 4 we need volunteers that will take ownership of certain areas of the wiki and not take on more than they can actually handle <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_04#we_need_volunteers_tha…> * 5 preparing for 1.0.1 <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_04#preparing_for_1.0.1>

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freeswitch compilation and external libraries
by Vieri 03 Jun '08

03 Jun '08

Hi, I'd like to know if there's a reason for packaging so many external libraries in the FS tarball (eg. nspr, sofia...). I'm asking because: 1) the FS tarball could be a lot smaller than the current 20MB tgz and 17MB bz2. 2) I found that manual compilation fails on my system with the FS-provided nspr but succeeds if I use an external library. Please have a look at http://jira.freeswitch.org/browse/FSBUILD-50 Thanks! Vieri

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FS Newbie: Question regarding subscriber info
by Aadilkhan Maniyar 03 Jun '08

03 Jun '08

Hi All, I would like to know how can I maintain the subscribers information in a database. From what I see after playing around with a few things is that the users have to be configured in the $PREFIX/freeswitch/conf/directory by creating a xml file for each user. I would like to know is there any database wherein I can store/retrieve all the user information at will. TiA Regards, Aadil

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OZ question: disconnected number handling
by Michael Collins 03 Jun '08

03 Jun '08

Guys, I don't know if this is "normal" or not so I'm hoping you can help me figure out what's up. I've got a disconnected number that I call on a PRI and the sequence goes like this: Dial number, hear ring back, see PROGRESS from telco, hear SIT and "you've reached a DC'd number...", then hear fast busy for 30 seconds (or so), then receive from telco that the call state is up, and then 3 seconds later call is terminated. I've PB'd the complete log here: http://pastebin.freeswitch.org/4549 FYI, I tried it with both with and without ignore_early_media. When ignoring early media I just get 60 seconds of FS internal ringback tone then I hear that the call failed. When not ignoring early media is when I get the SIT tone, etc. and that's also what is in pb 4549. Any thoughts on how to handle a call like this? FS doesn't consider the call "connected" because telco sends us PROGRESS_MEDIA but doesn't actually say that the call is "up" - at least as far as I can tell. Thanks, Michael

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beginner's question
by Dave 03 Jun '08

03 Jun '08

I used to install the asterisk on centos 5 box, and has little experience with sipX. In this weekend, I installed the freeswitch on the same box, and have a test drive. Here it is couple of my questions. 1. it seems that both asterisk and freeswitch will use the same port (5060, I guess), so if I like to start the freeswitch, I must kill the asterisk, can I change the default port to another port. 2. follow the first question, can I use freeswitch along with asterisk, if I can, what may need to be taken care. 3. I saw the integration guide of "sipX and freeswitch", is this one work with the latest FS release. 4. If I like to use the FS library as the telephony layer for some other application, do you have the demo or samples, the closet doc I can found from your web site is the "embedded application guide". 5. currently, I am using the vbuzzer as the voip service provider, like to know if there is anyway to keep the same VSP, but switch from asterisk to freeswitch, if can, is there a "how to" guide, if can not with vbuzzer, is there any VSP who claim the support of the FS. Thanks in advance. Dave ___________________________________________________________ Sent by ePrompter, the premier email notification software. Free download at http://www.ePrompter.com.

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Reregister gateway after ping failed
by Sune Kristensen 02 Jun '08

02 Jun '08

Hi I have set up a external sip gateway with the ping option to check if it is available and it works fine. Have been trying to find out how to reregister the gateway if it comes back up, but can't find any information about it, so is that possible? /Sune

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Freeswitch Ldap Integration
by Faraz R. Khan 02 Jun '08

02 Jun '08

First of all- Amazing project. Tired of asterisk deadlocking all the time we have been deploying asterisk with OpenSER as the registrar. Freeswitch is a huge relief! This is an extremely important feature we have been looking for. Asterisk realtime ldap integration is very flaky. I found this page: http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_Directory But there are no clues as to the schema, the queries generated and what exactly it can hold in Ldap. I am also curious to know whether sofia's sip registrations, gateways etc can be kept in LDAP. We are basically developing an extensive plugin based control panel and a Asterisk module is already ready. However, we are writing asterisk .conf files for managing asterisk. We would be quite pleased to develop a FreeSwitch Ldap plugin to manage users,sip gateways, groups, features, etc. Though the XML configuration file is extremely easy to parse and write, pure LDAP integration would be amazing. Any pointers on this would be appreciated. -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.529.0381 x200 www.emergen.biz

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error sip 500
by Pieter Eduard 02 Jun '08

02 Jun '08

Hi all, Im trying to bridge 2 softswitch (sysmaster and mera mvts 2) using fs, i already setup the dialplan and external gateway in the sip profile but i'm having this trouble when trying to send call to the mvts 2, sometime the call could establish fine and the next minute the call to the same number failed with this error : variable_proto_specific_hangup_cause: [sip:500] variable_originate_disposition: [NORMAL_TEMPORARY_FAILURE] variable_hangup_cause: [NORMAL_TEMPORARY_FAILURE] what can cause this error? every failed call is never logged at the mvts2 box as like fs never sends the call but at the fs log i see that it already send the call to the right gateway (mvts2). -Pieter-

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RTCP
by Peder ＠ NetworkOblivion 02 Jun '08

02 Jun '08

Does FS support RTCP? I am interested in getting per call quality stats from it.

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Media Bugs
by Michael Collins 02 Jun '08

02 Jun '08

Anthm/MikeJ/bkw, Any chance you could write a few paragraphs about media bugs? Nothing super in depth, just a basic definition of what they are, why one might use them, etc. I'd like to update the wiki with at least a basic write-up on media bugs. Thanks! -MC

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