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Re: [Freeswitch-users] sessions and CS_INIT events
by Juan Pablo L. 07 Mar '15

07 Mar '15

Thank you very much for the advise. I accidentally found it to be useful as well. --- Original Message --- From: "Anthony Minessale" <anthony.minessale(a)gmail.com> Sent: March 6, 2015 2:23 PM To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] sessions and CS_INIT events You probably want to look at the dialplan module interface as well. And pay attention to the direction variable to distinguish inbound from out bound Especially at the routing state not init. On Thu, Mar 5, 2015 at 12:05 PM, Juan Pablo L. <jpablolorenzetti(a)hotmail.com > wrote: > Hi, yes i had a look at it, and yes the needs are similar, i used it at > the beginning > to get started, and i m using as a reference at this point but it seems > that the use cases > are different . thanks! > > > > ------------------------------ > From: mike(a)jerris.com > Date: Thu, 5 Mar 2015 12:46:28 -0500 > > To: freeswitch-users(a)lists.freeswitch.org > Subject: Re: [Freeswitch-users] sessions and CS_INIT events > > Given the similarity in purpose, I would look closely at how > mod_nibblebill interfaces with freeswitch. It sounds like your interface > needs are nearly identical. > > > On Mar 5, 2015, at 9:23 AM, Juan Pablo L. <jpablolorenzetti(a)hotmail.com> > wrote: > > Thank you very much guys for your contributions. > > I m doing it as a module for couple of reasons, being the > most important (i believe) performance, because the module > i m working on is to do real time charging of voice calls on a switch > that is already serving as a RBT service plus a bunch of IVR's to purchase > services, this is for a ~150K user base on a single machine (cold standby) > this switch is also scheduled to soon start providing hosted PBX services, > > so going the script direction > i personally dont see that as an option at all. I do use scripts for small > no so much used > much simpler stuff though, e.g: a lua script takes care of authenticating > users > when doing international calls from company extensions in the hosted PBX > solution. > > The other reason i chose to do > this as a module because C is the language i feel more comfortable with. > i hope this clarifies i little bit this. > > Moving on, right now i m developing on a test freeswitch that we have and > yes i noticed > that subscribing to the CS_INIT event does represent a big problem > because i get notified for every single of those events that is generated > on > freeswitch which would be very inconvenient because as i mentioned, the > same > switch does many other things that i m not interested in, so i m going to > try the advise > provided and try to do it in the dial plan, i will explore this option. > > thank you very much all! > > > > > > > Date: Thu, 5 Mar 2015 14:19:51 +0100 > > From: ssinyagin(a)gmail.com > > To: freeswitch-users(a)lists.freeswitch.org > > Subject: Re: [Freeswitch-users] sessions and CS_INIT events > > > > but for the task that OP has described, writing (and maintaining it in > > the long term) a module is really an overkill. Plus, he would also > > need to take care of multithreading within FreeSWITCH, as well as > > memory management, etc. > > > > Also, a module makes sense if it's some common task which can be > > re-used by others and published as open source. If it's some > > closed-source module for a specific enterprise task that Juan has, it > > just doesn't make sense and too much risk for a long-term solution. > > > > > > > > > > > > On Thu, Mar 5, 2015 at 1:56 PM, Vik Killa <vipkilla(a)gmail.com> wrote: > > > It all depends on what you are trying to do with your module. > > > You can use a dialplan handler in your module (see mod_enum for > example) to > > > route inbound calls using your custom dialplan. > > > You can use a state handler in your module and bind to channel states > (much > > > like binding to events). > > > You can create a dialplan app in your module to execute code when the > app is > > > called in dialplan > > > (Example: <application action="mycustom_module" data="${uuid} arg1 > arg2"/>) > > > You can use an endpoint in your module to originate calls outbound (see > > > mod_lcr or mod_callcenter for an example) > > > Also, you can create an API for your module > > > > > > IMO creating a module is much more powerful than using a script with > ESL. > > > But if you are going to create a module, you really don't need to mess > with > > > events (unless they are very specific events like CUSTOM::) because > your > > > module has access to much of the freeswitch core. > > > > > > Thanks. > > > > > > > > > > > > On Thu, Mar 5, 2015 at 5:07 AM, Stanislav Sinyagin < > ssinyagin(a)gmail.com> > > > wrote: > > >> > > >> why at all do you need it to be a C module inside FreeSWITCH? > > >> > > >> Why not writing an ESL program which would subscribe to events and > > >> perform the needed actions? > > >> > > >> How about the following scenario: > > >> > > >> 1. In the XML dialplan, you execute "park" application on the incoming > > >> call. > > >> > > >> 2. Your program is listening to events via ESL, and it recognizes that > > >> a channel has been parked > > >> > > >> 3. Your program starts to playback the ringback tone into that channel > > >> > > >> 4. Your program performs all the needed lookups and sets needed > > >> variables on the channel > > >> > > >> 5. Your program transfers or bridges the call where needed. > > >> > > >> This is quite easy to implement in any programming language of your > > >> choice, easy to debug, and it's easily scalable. It can be done in a > > >> multi-threading fashion, like Go or Erlang, or even Java, and perform > > >> as many parallel calls as required. > > >> > > >> quite easy, and you don't have to mess with FreeSWITCH internals :) > > >> > > >> > > >> > > >> > > >> > > >> > > >> On Thu, Mar 5, 2015 at 4:07 AM, Juan Pablo L. > > >> <jpablolorenzetti(a)hotmail.com> wrote: > > >> > Hi, i m writing a module in C that needs to check for certain > > >> > information in > > >> > a > > >> > database for the caller and the destination number, > > >> > for this the module is subscribing to the CS_INIT channel events, so > > >> > everytime a channel is created > > >> > the module callback is called and it checks the numbers, > > >> > the problem is that the callback gets called twice, > > >> > for the creation of the a-leg of the call and the creation of the > b-leg. > > >> > Is there any way to accomplish what i m trying to do ? > > >> > Am i doing it the wrong way? > > >> > I have already try getting testing for the flags in the channel but > it > > >> > did > > >> > not work, > > >> > testing of originator or originating does not yield anything .... > > >> > > > >> > i might be doing it wrong maybe ? > > >> > > > >> > Thanks! > > >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org > http://www.freeswitchsolutions.com Official FreeSWITCH Sites > http://www.freeswitch.org http://confluence.freeswitch.org > http://www.cluecon.com 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > 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 > -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com 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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sessions and CS_INIT events
by Juan Pablo L. 06 Mar '15

06 Mar '15

Hi, i m writing a module in C that needs to check for certain information in a database for the caller and the destination number, for this the module is subscribing to the CS_INIT channel events, so everytime a channel is created the module callback is called and it checks the numbers, the problem is that the callback gets called twice, for the creation of the a-leg of the call and the creation of the b-leg. Is there any way to accomplish what i m trying to do ? Am i doing it the wrong way? I have already try getting testing for the flags in the channel but it did not work, testing of originator or originating does not yield anything .... i might be doing it wrong maybe ? Thanks!

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Freeswitch RTP handling payload RFC 4040 / clearmode
by Regis M 06 Mar '15

06 Mar '15

Hi, Does FS support RFC 4040 (https://tools.ietf.org/html/rfc4040) clearmode ? "This document describes how to carry 64 kbit/s channel data transparently in RTP packets, using a pseudo-codec called "Clearmode". It also serves as registration for a related MIME type called "audio/clearmode"." Not found on wiki or confluence.. Regards,

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Flooded with Stun Errors
by Naveen Khanna 06 Mar '15

06 Mar '15

Hi, I am getting flood of following messages. 2015-03-05 13:49:25.766474 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE 2015-03-05 13:49:25.766474 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID 2015-03-05 13:49:25.926473 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2002@df7jal23ls0d.invalid audio stat 98.00 2012/2042 flaws: 30 mos: 4.48 v: 23.30 9.52/400.00 2015-03-05 13:49:25.946476 [ALERT] switch_rtp.c:5440 sofia/internal/sip:2002@df7jal23ls0d.invalid syncing 1 audio packet(s) 2015-03-05 13:49:25.986475 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2003@df7jal23ls0d.invalid audio stat 99.00 241/242 flaws: 1 mos: 4.49 v: 55.49 10.00/400.00 2015-03-05 13:49:26.146475 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2002@df7jal23ls0d.invalid audio stat 98.00 2012/2043 flaws: 31 mos: 4.48 v: 23.29 9.52/400.00 2015-03-05 13:49:26.166475 [ALERT] switch_rtp.c:5440 sofia/internal/sip:2002@df7jal23ls0d.invalid syncing 1 audio packet(s) 2015-03-05 13:49:26.286473 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE 2015-03-05 13:49:26.286473 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID 2015-03-05 13:49:26.366474 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2002@df7jal23ls0d.invalid audio stat 98.00 2012/2044 flaws: 32 mos: 4.48 v: 23.28 9.52/400.00 2015-03-05 13:49:26.426473 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE 2015-03-05 13:49:26.426473 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID 2015-03-05 13:49:26.486474 [ALERT] switch_rtp.c:5440 sofia/internal/sip:2002@df7jal23ls0d.invalid syncing 1 audio packet(s) 2015-03-05 13:49:26.666475 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE 2015-03-05 13:49:26.666475 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID Requesting an insight what could have been causing this. Regards, Naveen Khanna

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 06 Mar '15

06 Mar '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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Does Sangoma Analog card support callerid in dtmf format?
by james.zhu 06 Mar '15

06 Mar '15

Hello: My customer installed Sangoma A200 with FreeTDM. A200 can not get the callerid in dtmf format, But if the callerid send by FSK format, the callerid can be received. I also take a loot at the code for callerid process in freetdm, it seems lack of supporting the callerid in dtmf format. Mr.Moises also mentioned that three year ago: http://freeswitch-users.2379917.n2.nabble.com/Caller-id-on-incoming-FXO-wit… I am not very sure the problem is still there. I post this issue to with logs, please help me check that: https://freeswitch.org/jira/browse/OPENZAP-235?jql=project%20%3D%20OPENZAP%… I think callerid in dtmf/fsk format should be a standard feature, hope it can be resolved soon.

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freeswitch support
by Antony Aguirre Morales 06 Mar '15

06 Mar '15

I have configured my fs with xml_curl module in the directory part, may serve 100,000 extensions from a single FS or is there any limitation as to the software? IN the hardware I have a server with the following specifications: Brand: DELL RAM: 4GB CPU: Intel (R) Xeon (R) CPU E5405 @ 2.00GHz cores: 4 DD: 50GB regards.

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Stale Channels build up
by Naveen Khanna 06 Mar '15

06 Mar '15

Hi, I am getting Stale Channels build up in my Freeswitch installation. We have a deployed a Dialer solution which is based on mod_callcenter, and a continuous stream of call are generated using Originate api command. As time progress the hung channels build increase and moment it reaches a threshold of 420+ of stale channels the entire site goes down. The Sip connections / extensions get disconnect and there is no means that they can get connected again. However, switch continues receive incoming calls, but extensions / agents sitting behind mod_callcenter are unable to answer calls. The only solution that I have is to restart the switch. I am using : Freeswitch 1.4 enabled with WebRTC support, SipmML5 based Sip Client and postgres as database. The solution is deployed on CentOS 6.6 64 bit. Regards, Naveen Khanna

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Conference Announce Count Inline - Not working
by T Fred Farmington 05 Mar '15

05 Mar '15

I am following the instructions on: "This example is a very quick and dirty dialplan and conference config that lets you hear how many callers are in conference." https://wiki.freeswitch.org/wiki/Conference_Announce_Count_Inline Within my existing conf\autoload_configs\conference.conf.xml I added the new caller controls: <group name="plain"> as shown in the referenced page. Additionally within the same file I edited the <profile name="default"> to use the new: <param name="caller-controls" value="plain"/> I then created a new XML file: conf/dialplan/default/01_Announce_Conf_Count.xml containing the code shown in the referenced page. Lastly I edited this file to change the last 'application' line beginning with: application="say" to the following <action application="phrase" data="callers_in_conf,${conf_count}"/> so as to play the existing macro phrase which exists in: conf\lang\en\ivr\sounds.xml It should be playing existing wav files and, therefore not need, TTS But when I enter the conference.. 1. The first caller in gets a 'voice' message played indicating: "You are currently the only person in this conference" 2. But subsequent callers get nothing but a tone upon entry into the conference. The intended macro phrase is not playing. What needs to change to make the 'Announce Conference Participant Count" work. Any assistance would be greatly appreciated. Thanks

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Re: [Freeswitch-users] Real-time billing application for the, FreeSWITCH (mod_lua, mod_perl or ESL)
by DanB 05 Mar '15

05 Mar '15

One more thing to consider when you build your real-time billing application is call authorize before connect. If you only treat answer and hangup you can end up with the call without balance going through and eating some seconds at each connect (which in case of DoS requests can result in quite serious amounts for you). In CGRateS we use park application in dialplan to put the call on hold before being authorized by the same CHANNEL_PARK event and loop the call through dialplan back when we are done checking it. Just my two cents, DanB On 05.03.2015 18:36, freeswitch-users-request(a)lists.freeswitch.org wrote: > 015-03-04 17:17 GMT-05:00 jorgemariodlc<jorgemariodlc(a)gmail.com>: > >> >I actually working in it, I found it yesterday you need to add those lines >> >(/autoload_configs/lua.conf.xml): >> > >> ><configuration name="lua.conf" description="LUA Configuration"> >> > <settings> >> > >> > <hook event="CHANNEL_ANSWER" script="answer.lua"/> >> > <hook event="CHANNEL_HANGUP_COMPLETE" script="hangup.lua"/> >> > >> > </settings> >> ></configuration> >> > >> >Check this link to know what information is in each event, because it's >> >important to handle the direction-call (Outbound, Inbound) >> >https://wiki.freeswitch.org/wiki/Event_List

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