- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Outbound Caller ID on anoymous calls
by bennygeorge 31 Jan '12

31 Jan '12

Hello all we have a situation where we receive the caller id in from as per below: INVITE sip:123456789@1.1.1.1 SIP/2.0. From: "anonymous" <sip:1.2.3.4>; when this happens, freeswitch sends the call out and inserts the caller id number of 0000000000 I have changed the vars.xml to the following: <X-PRE-PROCESS cmd="set" data="outbound_caller_name=Anonymous"/> <X-PRE-PROCESS cmd="set" data="outbound_caller_id=anonymous(a)anonymous.invalid"/> however this doesn't work. What I'm trying to achieve is that if FS doesn't receive a good caller id, then it will format it with anonymous(a)anonymous.invalid This particular carrier is not using PAI. It would be fine if FS left the from number empty - i did try changing the vars.xml to the following: <X-PRE-PROCESS cmd="set" data="outbound_caller_id=_undef_"/> but that didn't work either any help greatly appreciated as always brgds -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/RE-Outbound-Caller-ID-on-anoy… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Installing with support for Digium card ... no freetdm in modules.conf?
by Kevin White 31 Jan '12

31 Jan '12

Hi all, I apologize for such a novice question, but I'm trying to get FreeSwitch installed correctly to support my Digium analog card, and then have Plivo run on top of it. According to FreeTDM site, it says to simply uncomment the mod_freetdm line in modules.conf of freeswitch. But I don't have that line in my modules.conf. I know I could just add it, but I'm not sure if the rest of the system actually has the required stuff in it to work that way. I look in the FreeSwitch 1.0.6 book and it has little to no actual information on doing anything with FreeTDM other than that it exists and replaces OpenZAP. No actual instructions to make it work. What's the "proper" way to get freeswitch installed and working with FreeTDM for digium analog card? Thanks! Kevin ________________________________ Kevin White Senior Software Engineer (801) 727-2362 office CentralLogic www.centrallogic.com<http://www.centrallogic.com>

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Music on hold
by Darcy 31 Jan '12

31 Jan '12

When I receive a call and place it on hold, the calling party receives MOH as specified in <X-PRE-PROCESS cmd="set" data= hold_music=/usr/local/freeswitch/sounds/music/8000/moh.WAV"/>. Also, I transfer this call to another extension, the caller receives MOH during the transfer, however, If I place an outbound call then place the party on hold, they just get silence. Am I missing a basic setting here. I also cannot transfer this call or park it. Using snom 3xx, 7xx and 8xx phones. Thanks, Darcy Primrose

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CAMA Stack
by Michael Gende 31 Jan '12

31 Jan '12

I'm just starting a search on this subject, but does a stack exist for FS implementing CAMA, or at least a subset of that protocol? Thanks in Advance for any info, an email address of someone working on this now, or anyone who has considered it. Regards, Mike G.

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Call FW, redirection
by Miha Zoubek 31 Jan '12

31 Jan '12

Hi, I need a little help about understanding 302 redirect. I have set on my phone (registered on FS) redirection on number, that is not registered on FS. I have made a condition in my external dialplan. IF the number is on FS that dial to this number, if the number is not on FS, than FS should redirect to default dialplan and call outside number. I have problems with redirecting (302) to default dialplan. I have put in sip_profiles/external this <param name="manual-redirect" value="true"/> and set in my public dialplan <anti-action application="set" data="sip_redirect_contex=default" />. How to put this variables for redirect BEFORE the redirect condition is hit. You can see in mypastebin what happens. http://pastebin.freeswitch.org/18257 Tahnks! -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o.

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unable to sending and receiving fax between two peer
by amit nakum 31 Jan '12

31 Jan '12

Dear Friends Pls Help me...... I am unable to send and receiving fax directly between two zoiper peer i.e 1001(zoiper peer) --->FS---->1002(zoiper peer) using spandsp module. My actual test scenario is,I dial 1002 from 1001 with selecting some.tiff file,and fax should be send to 1002.but this scenario is not working in zoiper. I also try another scenario.By dialing 555 and try to set some T.38 variable and then bridge the call to 1002 for sending fax. <extension name="555" continue=true> <condition field="destination_number" expression="^(555)$"> <action application="answer"/> <action application="set" data="fax_enable_t38=true"/> <action application="export" data="fax_enable_t38_request=true"/> <action application="set" data="execute_on_answer=t38_gateway self nocng"/> <action application="bridge" data="user/1001"/> </condition> </extension> but this scenario is also not working. pls help me resolved my problem soon. Thanks Amit Nakum

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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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