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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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Answering Machine Detection
by David Brazier 30 May '08

30 May '08

Hi All Just wanted to clarify: * Is there any means of answering machine detection in FreeSwitch (along the lines of Asterisk app_amd)? * If not, could something like that be implemented in a custom app (I'm not sure if there is sufficient access to the media stream?), or does it need to be in the core? * If in the core, switch_ivr_record_file seems like a good starting point, as it already does speech detection. I would hope I would be able to implement something starting from switch_ivr_record_file, but I didn't want to charge in at that level if it would be better done elsewhere. Any thoughts appreciated. David PS Great project - a real credit to the open source community! 360CRM Limited, registered address, Caledonia House, 89 Seaward Street, Glasgow, G41 1HJ, Registered in Scotland, company number SC261952. This email and its attachments may be confidential and are intended solely for the use of the individual to whom it is addressed. Any views or opinions expressed are solely those of the author and do not necessarily represent those of 360CRM Ltd. This email does not form any part of a contract. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error.

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Max of 170 channels in the conference room.
by Johny Kadarisman 30 May '08

30 May '08

Hi all, I build a new test boxes for freeswitch. and trying to simulate high load condition. I have one freeswitch box that host conference apps, and another box with simple js to originate call to the conference room. To monitor the sound quality, I have one phones that dial into the conference room, and then start streaming/playing moh music in the conference. In many scenario that i tried (mute, unmute, one or multiple rooms), the max total number of channels is about 170 channels before FS start to slowing down on creating new channels. The Cpu's only tops around 40-50%, and the voice/rtp start to jitter or delays on my phone. These test boxes is running on 2 processor 2.8GHz Dual Core xeon cpu's and about 1Gb memory. I think the number already pretty impressive, at least for me :) But is this limit sounds right from others experience? Can I still do more tuning? Thanks, Johny K.

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RESTful Bounty ..... Putting the competition to REST with RestFul VoIP2.x services
by EdPimentl 29 May '08

29 May '08

I am putting a bounty to add RESTful WebServices in FreeSwitch. The bounty ask that TONIC is use as the RESTful Framework. TONIC should then be use to create simple CALLback application and establish ad-hoc VoiceConference. To those not familiar with TONIC here is write up. Tonic <http://tonic.sourceforge.net/> A *REST*ful Web App Development Framework Tonic is an open source less is more, RESTful Web application development and Web site management PHP script designed to do things "the right way", where resources are king and the framework gets out of the way and leaves the developer to get on with it. Tonic helps you develop Web applications that embrace the way the Web really works, enabling your applications to scale, extend and work with other systems easily. Everything is a resourceTonic works exclusively with resources, a resource is simply a piece of data with some metadata accessable by a URL using HTTP methods. Everything within a Tonic system is a resource and so can be handled via the standard HTTP interface.Domain modelling through resourcesWhen modelling your problem domain within a Tonic system, every domain object is a resource or a collection of resources.Keep it simple, stupid!Following the Unix philosophy of many small parts working together via a common interface, Tonic uses resources as it's "programs" and HTTP as it's "streams" to enable you to build complexity out of simplicity. Main features Tonic embraces a number of standard Web features and best practices, as well as features, principles and ideas taken from PHP, Unix, and other software design patterns: - Resource editing via HTTP PUT and DELETE methods (or via the Resource::save() and Resource::remove() PHP methods) - Easy to extend via new resource classes - HTTP basic and digest authentication - Resource method permissions to provide authorisation to individual resource methods - Output templating via Smarty - Output caching via HTTP 304 response codes and modified headers - Extendable resource storage types via adapters - Content negotiation - Match multiple URLs to a single resource - Follows Unix modular design principles And here is why everything one does or will do, should be done via RESTFul messaging.. Think about it, Google, AMAZON, YAHOO have standardized on REST. http://www.infoq.com/articles/rest-introduction http://wiki.opengarden.org/REST/REST_for_the_Rest_of_Us http://en.wikipedia.org/wiki/Representational_State_Transfer http://rest.blueoxen.net/cgi-bin/wiki.pl?FrontPage http://www.jibbering.com/2002/4/httprequest.html http://java.sun.com/developer/technicalArticles/WebServices/restful/ http://www.ics.uci.edu/~fielding/pubs/dissertation/software_arch.htm#sec_1_2 ****** -E http://mobiquity.ws

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