January 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 67, Issue 264
by clive engelberg 31 Jan '12

31 Jan '12

Hi I understand Josue's question to be about RTP multiplexing, where on a packet level the media streams of multiple calls are bundled within a single large packet with one header. This method saves a lot of bandwidth as the packet headers are fairly large, and reducing these headers can easily halve bandwidth usage. This is quite useful for low bandwdith links. I chatted to Anthm long ago about this, but it is not a priority issue due to the fact that most Freeswitch users have more than enough bandwidth that they do not have to be concerned about how to squeeze 30 channels of voice down a 256k line :) regards Clive On Tue, 31 Jan 2012 06:22:48 +0300 freeswitch-users-request(a)lists.freeswitch.org wrote > Josue, > > > Is there any way to have a SIP Trunk. I mean to have for example 32 > channels merged in one? or something like this? > >When i try to find SIP trunk on internet i just see options for TDM > gateway or similar but not really a multiplexed trunk. > > Let's define a SIP trunk for this discussion as "virtual > internet connection" between your switch or gateway's IP address and > your SIP provider's IP address. Over this connection your SIP provider > can send you any number of simultaneous conversations (basically > equivalent to "channels" or "time slots" in the TDM world). A SIP > "trunk" can have as many simultaneous conversations ("Channels") as you > wish, limited only by your internet bandwidth, how many "channels" you > wish to pay for, and any limits set by your SIP provider. Also the SIP > trunk can have as many different DID numbers as you wish or as limited > by your provider. In this scenario, your provider is multiplexing the > TDM sources into your SIP trunk. In its most basic form, a SIP provider > can "point" the SIP traffic directly at your Freeswitch's external IP > address and there you are!! Set up Freeswitch to be compatible with > your Provider's requirements and you are there. > > As far as TDM gateways are concerned, if you are using legacy > equipment that requires TDM service, there are several SIP-TDM gateways > available that can receive the SIP traffic from your Provider and > convert it back-and-forth between SIP and TDM. As far as the > suitability and hardware requirements for Freeswitch to perform that > function, since I am newly acquainted with Freeswitch, I leave that > discussion to those who are experienced. > > Win N. > > > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of > Sergey Okhapkin > Sent: Saturday, January 28, 2012 12:14 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SIP Multiplexed Trunk as IAX2 or TDM > > SIP (and RTP) have no concept of trunking/audio frames multiplexing > unlike > IAX2 and TDM. > > On Saturday 28 January 2012 21:06:50 Josue Diaz Cruz wrote: > > Is there any way to have a SIP Trunk. I mean to have for example 32 > channels > > merged in one? or something like this? When i try to find SIP trunk on > > internet i just see options for TDM gateway or similar but not really > a > > multiplexed trunk. > > > > Can we do something with freeswitch? > > > > Josue Diaz Cruz > > > > Departamento Tecnico y Soporte > > > > <mailto:jdiaz@coinfru.com> jdiaz(a)coinfru.com > > > > > > > > C/ Balsicas 3 > > > > Alquerias | 30580 | Murcia > > > > <http://www.coinfru.com/> www.coinfru.com > > ________________________________________________________________________ > _ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.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 ____________________________________________________________ South Africas premier free email service - www.webmail.co.za For super low premiums, click here. http://www.dialdirect.co.za/?vdn=15828

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Update on Brian's Condition
by Michael Collins 31 Jan '12

31 Jan '12

Hey all, Brian says that the latest is as follows: The lump in his chest is actually a lymph node. They are running further tests to see if they can track down what's going with it. The mass in his left lung appears just to be an infection, but again they are running more tests to make sure. Please everyone keep Brian in your thoughts and prayers. Thanks! -Michael

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Inbound ANI
by John Macleod 31 Jan '12

31 Jan '12

I have a toll-free number coming into FreeSwitch via SIP. In the log, I see the ANI of the caller (I have sanitised this to protect the innocent): 2012-01-29 10:23:04.968117 [INFO] mod_dialplan_xml.c:481 Processing unknown <0000000000>->18001231234 in context public 2012-01-29 10:23:04.968117 [INFO] switch_core_session.c:1332 sofia/external/614xxxxxxx(a)x.x.x.x setting session heartbeat to 60 second(s). 2012-01-29 10:23:05.008105 [INFO] switch_core_session.c:2135 Sending early media 2012-01-29 10:23:05.008105 [NOTICE] mod_sofia.c:2576 Pre-Answer sofia/external/614xxxxxxx(a)x.x.x.x! 2012-01-29 10:23:07.328098 [NOTICE] mod_dptools.c:1135 Channel [sofia/external/614xxxxxxx(a)x.x.x.x] has been answered 2012-01-29 10:23:12.108110 [NOTICE] mod_dptools.c:1121 Hangup sofia/external/614xxxxxxx(a)x.x.x.x [CS_RESET] [NORMAL_CLEARING] 2012-01-29 10:23:12.108110 [NOTICE] switch_core_session.c:1398 Session 7 (sofia/external/614xxxxxxx(a)x.x.x.x) Ended 2012-01-29 10:23:12.108110 [NOTICE] switch_core_session.c:1400 Close Channel sofia/external/614xxxxxxx(a)x.x.x.x [CS_DESTROY] This is just a Plivo app that plays a greeting and disconnects. However, in the CDRs, I do not get the ANI, I get 0000000000 and the DNIS, I would like the ANI and DNIS. "unknown","0000000000","18001231234","public","2012-01-29 10:23:04","2012-01-29 10:23:07","2012-01-29 10:23:12","8","5","NORMAL_CLEARING","49209654-4aa6-11e1-ac59-9949a61da724","","","PCMU","PCMU" Is there a parameter I am missing to pass this variable from sofia? Thanks, John

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Re: [Freeswitch-users] freeswitch billing
by David Villasmil 31 Jan '12

31 Jan '12

Hello Guys, Just commited a new version in github. I have NOT included the retail part yet. This includes a basic install script which you shuld execute like this: The script WILL REPLACE some xml config, so PLEASE CHECK THE FILES IN gateway/ to make sure it doesn't messes up your existing configuration!! you MUST set some variables in "variables.conf", you execute the install script like so: perl install.pl For a full explanation read the README file... Let me know your experience with it. David On Sun, Jan 29, 2012 at 11:44 AM, Sherif Omran <sherif.omran(a)gmx.de> wrote: > Hi David, > > to which path should i copy the PERL gateway files ? > https://github.com/davidcsi/FreeSWITCH-Billing/tree/master/gateway_scripts > > I have a usr calle FreeSwitch and FS is located at > /usr/local/freeswitch > > > There are some modules that were not installed by default and i skipped > them. Please take a look the attached file. This is what i made so far. > > Waiting your reply > > kind regards, > Sherif > > > > > > On Jan 28, 2012, at 8:29 PM, David Villasmil wrote: > > No, I don't. > > On Sat, Jan 28, 2012 at 1:09 AM, Sherif Omran <sherif.omran(a)gmx.de> wrote: > >> Hi David, >> >> do you use mod_nibble? >> >> kind regards, >> Sherif >> >> >> On Jan 27, 2012, at 2:47 PM, David Villasmil wrote: >> >> hello Sherif, >> >> Help it is always welcome, even more because i'm not an expert on web >> pages. >> I think it is better if we setup a chat conference to talk about this and >> any other questions you may have. >> >> The CDRs are posted via web with xml_curl. >> The scripts in the gateway directoy are to be copied into the freeswitch >> server. "gateway" is just a name i gave to the server. >> >> Hope this helps. there are some changes that you need to make before >> making it work, but i can walk you through it. >> >> David >> >> On Thu, Jan 26, 2012 at 10:13 PM, Sherif Omran <sherif.omran(a)gmx.de>wrote: >> >>> Dear David, >>> >>> there is no folder called gateway in my freeswitch folder. the normal >>> vars.xml file is located in usr/local/freeswitch/conf/var.xml >>> >>> I am a bit confused, do you mean one should create a new folder called >>> gateway inside the conf folder? or should it be inside the freeswitch >>> folder? >>> >>> Also can it work on 1 server? >>> >>> waiting your reply >>> >>> best regards, >>> Sherif >>> >>> >>> On Jan 9, 2012, at 6:35 PM, David Villasmil wrote: >>> >>> Sorry, >>> >>> It is on https://github.com/davidcsi/FreeSWITCH-Billing >>> >>> David >>> >>> On Mon, Jan 9, 2012 at 6:34 PM, David Villasmil < >>> david.villasmil.work(a)gmail.com> wrote: >>> >>>> Hello Sheriff, >>>> >>>> I have been using it for more than a year on production. >>>> >>>> Enjoy >>>> >>>> David >>>> >>>> >>>> On Sat, Jan 7, 2012 at 11:27 PM, Sherif Omran <sherif.omran(a)gmx.de>wrote: >>>> >>>>> Hi David >>>>> >>>>> Is it a full application ready to be used or still under development? >>>>> Could you please give me the link to the download >>>>> >>>>> also please add your program here >>>>> >>>>> http://wiki.freeswitch.org/wiki/Billing >>>>> >>>>> thank you >>>>> >>>>> regards, >>>>> Sherif >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On Jan 1, 2012, at 10:48 PM, David Villasmil wrote: >>>>> >>>>> Hello Sherif, >>>>> >>>>> Unfortunately, I don't have a script yet. But the Readme explains how >>>>> to install it. >>>>> I don't have a online demo either as of now, but I will set it up >>>>> shortly. >>>>> >>>>> Thanks for trying it >>>>> >>>>> David >>>>> >>>>> On Sun, Jan 1, 2012 at 2:27 PM, Sherif Omran <sherif.omran(a)gmx.de>wrote: >>>>> >>>>>> hi David >>>>>> >>>>>> I am interested in your module, do you have an installation script >>>>>> for a running FS? >>>>>> >>>>>> Is there a demo online? >>>>>> >>>>>> best regards, >>>>>> Sherif Omran >>>>>> >>>>>> >>>>> >>>>> >>>> >>> >>> >> >> > > >

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how to find out the error?
by David Villasmil 30 Jan '12

30 Jan '12

Hello Guys, When I try to start FS I get the following: 2012-01-30 22:41:36.184340 [INFO] switch_event.c:631 Activate Eventing Engine. 2012-01-30 22:41:36.203503 [DEBUG] switch_event.c:610 Create event dispatch thread 0 Cannot Initialize [[error near line 46]: unclosed <!--] How do I find out the offending file??? there's no log containing the startup routines... Thanks David

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How to reload the gateway from the mod_managed
by Srini K 30 Jan '12

30 Jan '12

Hi, I like to add a gateway on the fly. FS applcation which is in mod_managed will read the gateway configuration from the db and will add/update the conf\sip_profiles\external\MygateWay.xml config file. How to execute this command "sofia profile external rescan reloadxml" to reload the gateway using the mod_managed application without using cli. RegardsSrini

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Extending the field size of core's db_data table?
by Yehavi Bourvine 30 Jan '12

30 Jan '12

Hello, We use the DB api for storing and caching various data. We need to store data that is longer than 255 characters, which is the current limit on the field size there. The backend for this API is MySQL (via ODBC). Can I just increase the field size in MySQL, or is there some dependency in FreeSwitch on this size? Thanks, __Yehavi:

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Equivalent of "s" extension in Asterisk
by Brian 30 Jan '12

30 Jan '12

Hi List, I'm quite new to FreeSwitch and I'm trying to replace an old asterisk installation with a new FreeSwitch. There is however one feature that is available with Asterisk and I haven't managed to figure it out how to do it with FreeSwitch. That is about the "s" extension in Asterisk dial plan. In the old Asterisk installation, when I pick up an analogic phone (plugged into an FXS slot of a Digium card) the "s" extension gets run and I can play early media to the phone. Does anyone know if it is possible to achieve the same thing with FreeSwitch/FreeTDM? Best regards, Brian

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Whats the best way to insert audio into a call from event socket
by Fraser Redmond 30 Jan '12

30 Jan '12

I want to insert a queue of voice recordings into a call, and be able to stop them all. Doing both via the event socket. Currently I'm doing: api uuid_broadcast $uuid fileName1.wav both api uuid_broadcast $uuid fileName2.wav both And stopping the audio with: api uuid_break $uuid all That works quite well, except for two problems: 1) I can't use it because when the audio is playing the parties can't hear each other talk. 2) There is a slight delay of about half a second before the audio starts (though I'm guessing this can't be helped.) Previously I'd used: api uuid_displace $uuid start fileName1.wav 0 mux But I don't think I'd been able to queue up more than one file with that. There's a few other possibilities that I've thought of, including going into a js/lua script and looping inside that, or changing the call into a conference, and using the conference commands. Rather than mess around trying lots of ideas, I'd rather focus on the one best approach - any ideas? Cheers, Fraser

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ZRTP
by Anto 30 Jan '12

30 Jan '12

Hi I tried to install new machines in a ZRTP with FreeSWITCH, but I have not found the ZRTP library. I haven´t found FreeSWITCH repositories, nor on the website of Zfone (http://zfoneproject.com) I do not like much the licensing (or mode of use) of this library, so searching I found these two projects: http://code.google.com/p/open-zrtp/ and http://www.gnutelephony.org/index.php/GNU_ZRTP. Can be used in FreeSWITCH? Thanks Best regards

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