Re: [Freeswitch-users] external call
On Sep 4, 2008, at 11:37 PM, Gayatri Kulkarni wrote: > How do I call a number that is not administered on the FS, through FS? You need to setup a dial plan that sends the call through a gateway... Marty > > Regards, > Gayatri Kulkarni > > - > Whenever you find yourself on the side of the majority, it is time > to pause and reflect. > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] call through service
Does Freeswitch implement the call through service? -- I couldnt find any relevant application. By call through service I mean: enables authorized corporate users outside CallThrough services are the most common form of service. They require no registration or pre-payment. On your existing telephone, you simply dial the service provider's number for the destination you require. You usually hear a introductory pre-recorded message and you then dial the number you want to talk to. You are charged by your existing telephone provider for the call to the initial number at the rate published by the CallThrough provider. Does Freeswitch implement the call through service? If it does, how can i invoke it through the console? Thanks a million for answering this, in advance, Regards, Gayatri Kulkarni ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Register cell phone (Gayatri Kulkarni)
I'm not sure about the Nokia 6085 but I have successfully used various mobile phones without a SIP client or wifi by using the Fring http://www.fring.com http://www.fring.com/downloado and GPRS/3G to connect to my FS box. If you have a good data plan this can be pretty economical. Hope this helps Steve Steven Brown [EMAIL PROTECTED] t. 08707706968 m. 07768755409 f. 07884636663 -- Message: 1 Date: Thu, 4 Sep 2008 22:17:06 +0530 From: "Gayatri Kulkarni" <[EMAIL PROTECTED]> Subject: Re: [Freeswitch-users] Register cell phone To: freeswitch-users@lists.freeswitch.org Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" Nokia 6085 On Thu, Sep 4, 2008 at 5:26 PM, hemant kumar <[EMAIL PROTECTED]> wrote: > any information on the model and make of cell phone? > > On Thu, Sep 4, 2008 at 5:18 PM, Gayatri Kulkarni <[EMAIL PROTECTED]>wrote: > >> Hi, >> >> Can I register my cell phone to FS? >> How can I do that? >> >> Regards, >> Gayatri Kulkarni >> >> - >> Whenever you find yourself on the side of the majority, it is time to >> pause and reflect. >> >> ___ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> ers >> http://www.freeswitch.org >> >> > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] call through service
Hello, You'll need to write a script for FS in your choice of language - I knocked together a quick demo for one in Lua the other week. Something like this is possibly a good place to start: http://pastebin.freeswitch.org/5479 - change the bits in square brackets for your particular configuration. Obviously no warranties provided, I'm not responsible if it makes your cat get jiggy with your dog and makes your bowl of cereal start mooing one morning :) Once you've written the script, tie it to an entry in the dial plan and you're ready to rock. Hope that helps, Alex On 5 Sep 2008, at 12:42, Gayatri Kulkarni wrote: Does Freeswitch implement the call through service? -- I couldnt find any relevant application. By call through service I mean: CallThrough services are the most common form of service. They require no registration or pre-payment. On your existing telephone, you simply dial the service provider's number for the destination you require. You usually hear a introductory pre-recorded message and you then dial the number you want to talk to. You are charged by your existing telephone provider for the call to the initial number at the rate published by the CallThrough provider. Does Freeswitch implement the call through service? If it does, how can i invoke it through the console? Thanks a million for answering this, in advance, Regards, Gayatri Kulkarni ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] simulate att_xfer , threeway
I have no idea what you are talking about?
What exact dialplan are you using to test att_xfer there is a working
example in the default config.
On Thu, Sep 4, 2008 at 10:24 PM, Lee JJ <[EMAIL PROTECTED]> wrote:
> Hello :
>
> While the att_xfer , I collect show the calls and channels info .
> After press "0" , the calls is Zero , and found the last channel using
> application "three_way"
>
> # during att_xfer , press "0"
> [EMAIL PROTECTED]> show channels
> API CALL [show(channels)] output:
>
> uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate
> 33c232f5-e0ff-446e-ad93-8c2f80d61656,2008-09-05
> 11:00:40,1220583640,sofia/inter2/[EMAIL PROTECTED],CS_EXECUTE,1529,1529,
> 172.16.93.231
> ,1526,att_xfer,sofia/inter2/1528,XML,inter2,PCMU,8000,PCMU,8000
> 0d51ddb6-cac2-428b-a310-d9c54f979efc,2008-09-05
> 11:00:40,1220583640,sofia/inter2/1526,CS_EXCHANGE_MEDIA,Extension
> 1529,1529,172.16.93.231
> ,1526,playback,local_stream://moh,XML,inter2,PCMU,8000,PCMU,8000
> 501e6351-6b9f-46a6-81bc-9a2c5a3306fa,2008-09-05
> 11:00:53,1220583653,sofia/inter2/1528,CS_EXECUTE,Extension
> 1529,1529,172.16.93.231
> ,1528,three_way,33c232f5-e0ff-446e-ad93-8c2f80d61656,XML,inter2,PCMU,8000,PCMU,8000
> 3 total.
> [EMAIL PROTECTED]>
> [EMAIL PROTECTED]> show calls
> API CALL [show(calls)] output:
> 0 total.
>
> [EMAIL PROTECTED]> show calls
> API CALL [show(calls)] output:
>
> function,created,created_epoch,caller_cid_name,caller_cid_num,caller_dest_num,caller_chan_name,caller_uuid,callee_cid_name,callee_cid_num,callee_dest_num,callee_chan_name,callee_uuid
> 2008-09-05
> 11:00:42,1220583642,switch_ivr_multi_threaded_bridge,1529,1529,1526,sofia/inter2/
> [EMAIL PROTECTED],33c232f5-e0ff-446e-ad93-8c2f80d61656,Extension
> 1529,1529,1526,sofia/inter2/1526,0d51ddb6-cac2-428b-a310-d9c54f979efc
> 2008-09-05
> 11:00:56,1220583656,switch_ivr_multi_threaded_bridge,1529,1529,1526,sofia/inter2/
> [EMAIL PROTECTED],33c232f5-e0ff-446e-ad93-8c2f80d61656,Extension
> 1529,1529,1528,sofia/inter2/1528,501e6351-6b9f-46a6-81bc-9a2c5a3306fa
> 2 total.
> [EMAIL PROTECTED]>
> [EMAIL PROTECTED]> show channels
> API CALL [show(channels)] output:
>
> uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate
> 33c232f5-e0ff-446e-ad93-8c2f80d61656,2008-09-05
> 11:00:40,1220583640,sofia/inter2/[EMAIL PROTECTED],CS_EXECUTE,1529,1529,
> 172.16.93.231
> ,1526,att_xfer,sofia/inter2/1528,XML,inter2,PCMU,8000,PCMU,8000
> 0d51ddb6-cac2-428b-a310-d9c54f979efc,2008-09-05
> 11:00:40,1220583640,sofia/inter2/1526,CS_EXCHANGE_MEDIA,Extension
> 1529,1529,172.16.93.231
> ,1526,playback,local_stream://moh,XML,inter2,PCMU,8000,PCMU,8000
> 501e6351-6b9f-46a6-81bc-9a2c5a3306fa,2008-09-05
> 11:00:53,1220583653,sofia/inter2/1528,CS_EXCHANGE_MEDIA,Extension
> 1529,1529,172.16.93.231,1528,,,XML,inter2,PCMU,8000,PCMU,8000
> 3 total.
>
> So , what's the no calls changing meaning ?
>
> And I simulate the three_way working using scripts, pitty thing is
> either hanup immediately or someone in hold music.
> PLS give some advice .
>
> ### call test scripts staring
> n_sess = new Session() ;
> res = n_sess.originate(n_sess, "sofia/inter2/1528%10.20.1.233" );
>
> // wait for it ... answer
> n_sess.waitForAnswer(1);
>
> if ( n_sess.ready()) { //never go into here
>console_log("info", "n_sess uuid: "+ n_sess.uuid + "\n");
>n_sess.execute("three_way", session.uuid ) ;
> }
>
> ___
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> http://www.freeswitch.org
>
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:[EMAIL PROTECTED] <[EMAIL PROTECTED]>
GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]>
iax:[EMAIL PROTECTED]/888
googletalk:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
pstn:213-799-1400
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Re: [Freeswitch-users] Register cell phone
talking riddles On Fri, Sep 5, 2008 at 12:22 AM, Gayatri Kulkarni <[EMAIL PROTECTED]>wrote: > Can I register it via ENUM then? > > > *From:* Brian West <[EMAIL PROTECTED]> > *Sent:* Thursday, September 04, 2008 9:53 AM > *To:* freeswitch-users@lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Register cell phone > > And by register? You mean via sip? If so that phone doesn't have a SIP > client it from what I have googled. > /b > > On Sep 4, 2008, at 11:47 AM, Gayatri Kulkarni wrote: > > Nokia 6085 > > On Thu, Sep 4, 2008 at 5:26 PM, hemant kumar <[EMAIL PROTECTED]> > wrote: > >> any information on the model and make of cell phone? >> >> On Thu, Sep 4, 2008 at 5:18 PM, Gayatri Kulkarni < >> [EMAIL PROTECTED]> wrote: >> >>> Hi, >>> >>> Can I register my cell phone to FS? >>> How can I do that? >>> >>> Regards, >>> Gayatri Kulkarni >>> >>> - >>> Whenever you find yourself on the side of the majority, it is time to >>> pause and reflect. >>> >> > Brian West > sip:[EMAIL PROTECTED] > > > > > > > -- > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ > Freeswitch-users mailing list > Freeswitch-users@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 FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] <[EMAIL PROTECTED]> GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]<[EMAIL PROTECTED]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED]<[EMAIL PROTECTED]> pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] call through service
Thanks Alex! I'll try this one Regards, Gayatri Kulkarni On Fri, Sep 5, 2008 at 6:15 PM, Alex Kinch <[EMAIL PROTECTED]> wrote: > Hello, > You'll need to write a script for FS in your choice of language - I knocked > together a quick demo for one in Lua the other week. Something like this is > possibly a good place to start: http://pastebin.freeswitch.org/5479 - > change the bits in square brackets for your particular configuration. > Obviously no warranties provided, I'm not responsible if it makes your cat > get jiggy with your dog and makes your bowl of cereal start mooing one > morning :) > > Once you've written the script, tie it to an entry in the dial plan and > you're ready to rock. > > Hope that helps, > > Alex > > > On 5 Sep 2008, at 12:42, Gayatri Kulkarni wrote: > > Does Freeswitch implement the call through service? -- I couldnt find any > relevant application. > > By call through service I mean: > > CallThrough services are the most common form of service. They require no > registration or pre-payment. On your existing telephone, you simply dial the > service provider's number for the destination you require. You usually hear > a introductory pre-recorded message and you then dial the number you want to > talk to. You are charged by your existing telephone provider for the call to > the initial number at the rate published by the CallThrough provider. > > Does Freeswitch implement the call through service? > If it does, how can i invoke it through the console? > > Thanks a million for answering this, in advance, > Regards, > Gayatri Kulkarni > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] How to divert a virtual PSTN line to another server ?
I use a 'virtual' PSTN line (voip trunk) from (http://www.voxbone.com) as incoming external line to my Asterisk server (192.168.1.100) In my router I have have : Application Start End ProtocolIP Address - SIP 5004 to5082 Both(UDP&TCP) 192.168.1.100 RTP 5090 to 5100UDP 192.168.1.100 That works OK. Now I want to divert that PSTN line from Asterisk to my Freeswitch server (192.168.1.101) So I changed in my router the ip addreese from 192.168.1.100 to 192.168.1.101 Application Start End ProtocolIP Address - SIP 5004 to5082 Both(UDP&TCP) 192.168.1.101 RTP 5090 to 5100UDP 192.168.1.101 But.when an external call comes in, it still goes to Asterisk. Am I on the wrong track or ... (?) Rgds Henk ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] any VOIP service provider support the freeswitch now?
I like to try out the freeswitch in real life call, but besides the connection through the asterisk, it seems that it has no way to directly support by VOIP service provider. I am sure some of the VOIP service provider do see the potential business in this area, but not sure why we still didn't see the crowded. Can someone share with me the list of the VOIP service provider who can support the freeswitch connection, or any good method to connect to VOIP service throguh the SIP. Thanks. David ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] any VOIP service provider support the freeswitch now?
Many VOIP services supporting SIP can work with freeswitch now, for me, like www.voipstunt.com, www.voipraider.com, www.vbuzzer.com they are all working fine on my freeswitch via SIP. Thanks, Chris On Fri, Sep 5, 2008 at 2:45 PM, Dave <[EMAIL PROTECTED]> wrote: > >I like to try out the freeswitch in real life call, but besides the > connection through the asterisk, it seems that it has no way to directly > support by VOIP service provider. >I am sure some of the VOIP service provider do see the potential > business in this area, but not sure why we still didn't see the crowded. >Can someone share with me the list of the VOIP service provider who can > support the freeswitch connection, or any good method to connect to VOIP > service throguh the SIP. > Thanks. > > > > David > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] any VOIP service provider support the freeswitch now?
On Sep 5, 2008, at 11:45 AM, Dave wrote: > >I like to try out the freeswitch in real life call, but besides > the connection through the asterisk, it seems that it has no way to > directly support by VOIP service provider. Look Here: http://wiki.freeswitch.org/wiki/SIP_Provider_Examples Marty >I am sure some of the VOIP service provider do see the potential > business in this area, but not sure why we still didn't see the > crowded. >Can someone share with me the list of the VOIP service provider > who can support the freeswitch connection, or any good method to > connect to VOIP service throguh the SIP. > Thanks. > > > > David > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] any VOIP service provider support the freeswitch now?
http://wiki.freeswitch.org/wiki/Tested_Phone_Providers_Listing Tested_Phone_Providers_Listing -- View this message in context: http://www.nabble.com/any-VOIP-service-provider-support-the-freeswitch-now--tp19337344p19337894.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] any VOIP service provider support the freeswitch now?
On 9/5/2008 1:45 PM, Dave wrote: [snip] >Can someone share with me the list of the VOIP service provider who > can support the freeswitch connection, or any good method to connect to > VOIP service throguh the SIP. > Thanks. I have it working with voicepulse and vitelity. -Rupa ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] any VOIP service provider support the freeswitch now?
And Asterlink :P /b On Sep 5, 2008, at 2:36 PM, Rupa Schomaker (lists) wrote: > On 9/5/2008 1:45 PM, Dave wrote: > [snip] > >> Can someone share with me the list of the VOIP service provider who >> can support the freeswitch connection, or any good method to >> connect to >> VOIP service throguh the SIP. >>Thanks. > > I have it working with voicepulse and vitelity. > > -Rupa > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West sip:[EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] compile error (on windows with VS 2005)
I got the compile error on windows XP pro (with VS 2005). ..\..\src\switch_xml.c(2234) : error C2220: warning treated as error - no 'object' file generated ..\..\src\switch_xml.c(2234) : warning C4267: '=' : conversion from 'size_t' to 'int', possible loss of data ..\..\src\switch_xml.c(2266) : warning C4267: '=' : conversion from 'size_t' to 'unsigned int', possible loss of data ..\..\src\switch_xml.c(2507) : warning C4267: '=' : conversion from 'size_t' to 'unsigned int', possible loss of data ..\..\src\switch_xml.c(2522) : warning C4267: '=' : conversion from 'size_t' to 'int', possible loss of data ..\..\src\switch_xml.c(2529) : warning C4267: '=' : conversion from 'size_t' to 'unsigned int', possible loss of data ..\..\src\switch_xml.c(2600) : warning C4267: '=' : conversion from 'size_t' to 'int', possible loss of data Generating Code... Results Build log was saved at "file://g:\opensource\tele\freeswitch\freeswitch-1.0.1\w32\Library\Debug\BuildLog FreeSwitchCoreLib.htm" FreeSwitchCoreLib - 1 error(s), 6 warning(s) Please help me out. ___ Sent by ePrompter, the premier email notification software. Free download at http://www.ePrompter.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] post paid account creation
the TLS issue on windows seems to be unsolvable by me and the interest got over when i came to know my linksys device doesnt do the TLS which freeswitch supports. wouldnt it be better if FS could be backward compatible with the TLS dont by other devices as i cant seem to find a update for the firmware also and linksys not developing also so i started searching for other stuff and came across openSBC which does a form of encryption called XOR, very basic but the good news is it bypasses most blocks as it seem to function similar to voipswitch voip tunnel. neways, i think im gonna use that as a SBC as it bypasses my block so i can now access my voip server, bytheway they also have opensipstack which supports almost all the codecs, lets me not go into it. basically what im looking for is a way directly by modifying the xml files in FS to support account balances for all configured accounts on it. i mean, that user accounts i create, can they be configured to work on credit basis or lets just make it even simpler, can i make changes to xml files so that a specific user account can make calls of for eg: 200mins to any number? after each call that number should decrease based on call duration. not using a billing software but for temporary means is it possible and if so then how? -- View this message in context: http://www.nabble.com/post-paid-account-creation-tp19339543p19339543.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] any VOIP service provider support the freeswitch now?
Works fine with PennyTel, too. Gonz On Fri, Sep 5, 2008 at 4:43 PM, Brian West <[EMAIL PROTECTED]> wrote: > And Asterlink :P > > /b > > On Sep 5, 2008, at 2:36 PM, Rupa Schomaker (lists) wrote: > > > On 9/5/2008 1:45 PM, Dave wrote: > > [snip] > > > >> Can someone share with me the list of the VOIP service provider who > >> can support the freeswitch connection, or any good method to > >> connect to > >> VOIP service throguh the SIP. > >>Thanks. > > > > I have it working with voicepulse and vitelity. > > > > -Rupa > > > > ___ > > Freeswitch-users mailing list > > Freeswitch-users@lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > Brian West > sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> > > > > > > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] any VOIP service provider support the freeswitch now?
Lets just put this thread to rest and say It will pretty much work with ANY voip/itsp out there. ;) /b On Sep 5, 2008, at 4:54 PM, Gonzalo Servat wrote: > Works fine with PennyTel, too. > > Gonz Brian West sip:[EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] IVR Timeout/Invalid Option Actions
Is there a way to have an IVR menu take an action besides disconnecting if the digit timeout is reached or if there is an invalid option? An IVR that looks like this would be ideal, with two additional actions menu-timeout and menu-invalid. This would give the IVR a great deal more flexibility. -- Marc Lewis Avvatel Corporation ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] IVR Timeout/Invalid Option Actions
You would put something in the dialplan after the application="ivr" and transfer it elsewhere or execute another ivr. Thats what I would do. On Sep 5, 2008, at 7:01 PM, Marc Lewis wrote: > > An IVR that looks like this would be ideal, with two additional > actions > menu-timeout and menu-invalid. This would give the IVR a great deal > more flexibility. Brian West sip:[EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] SIP, NAT and Amazon EC2
Hello All,
I have installed freeswitch on a computing cloud (Amazon EC2). My main
network configurations are:
(FreeSwitch ARI LAN ) <---> (WAN) <--{INTERNET} ---> (WAN)
<> (SIP Soft or Hard Phone LAN)
I have pointed all of my clients to the external (5080) sip port
No matter what I try STUN, Port Forwarding or the "Use Asterisk Method"
settings in the directory, I get no audio. My guess would be the RTP
side, but at this point I feel like I have done everything I can and
still be a FreeSwitch Newb. Can anyone provide any in-site?
Seems like a great system ... more directly the idea of no hardware
dependency!
Thanks in advance
Damon
begin:vcard
fn:Damon Brown
n:Brown;Damon
org:Technicate Solutions;Development and Solutions
adr:Suite K;;7529 Sunset Avenue;Fair Oaks;CA;95628;USA
email;internet:[EMAIL PROTECTED]
title:Owner
tel;work:1.916.880.1128
tel;cell:1.916.601.8190
url:www.technicate.com
version:2.1
end:vcard
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Re: [Freeswitch-users] SIP, NAT and Amazon EC2
Did you happen to
ec2-authorize default -P udp -p 16384-32768
/b
On Sep 5, 2008, at 7:29 PM, Damon Brown wrote:
> Hello All,
>
> I have installed freeswitch on a computing cloud (Amazon EC2). My
> main network configurations are:
>
> (FreeSwitch ARI LAN ) <---> (WAN) <--{INTERNET} ---> (WAN)
> <> (SIP Soft or Hard Phone LAN)
>
> I have pointed all of my clients to the external (5080) sip port
>
> No matter what I try STUN, Port Forwarding or the "Use Asterisk
> Method" settings in the directory, I get no audio. My guess would
> be the RTP side, but at this point I feel like I have done
> everything I can and still be a FreeSwitch Newb. Can anyone provide
> any in-site?
>
> Seems like a great system ... more directly the idea of no hardware
> dependency!
> Thanks in advance
>
> Damon
> ___
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> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
Brian West
sip:[EMAIL PROTECTED]
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Re: [Freeswitch-users] SIP, NAT and Amazon EC2
Yes, I have all of the valid posts open on my security group
-d
-Original Message-
From: "Brian West" <[EMAIL PROTECTED]>
Sent: Friday, September 5, 2008 6:14pm
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2
Did you happen to
ec2-authorize default -P udp -p 16384-32768
/b
On Sep 5, 2008, at 7:29 PM, Damon Brown wrote:
> Hello All,
>
> I have installed freeswitch on a computing cloud (Amazon EC2). My
> main network configurations are:
>
> (FreeSwitch ARI LAN ) <---> (WAN) <--{INTERNET} ---> (WAN)
> <> (SIP Soft or Hard Phone LAN)
>
> I have pointed all of my clients to the external (5080) sip port
>
> No matter what I try STUN, Port Forwarding or the "Use Asterisk
> Method" settings in the directory, I get no audio. My guess would
> be the RTP side, but at this point I feel like I have done
> everything I can and still be a FreeSwitch Newb. Can anyone provide
> any in-site?
>
> Seems like a great system ... more directly the idea of no hardware
> dependency!
> Thanks in advance
>
> Damon
> ___
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> Freeswitch-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
Brian West
sip:[EMAIL PROTECTED]
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Re: [Freeswitch-users] SIP, NAT and Amazon EC2
You'll have to set an ext-sip-ip and ext-rtp-ip on the internal.xml profile on ec2 duplicate them from the external profile. /b On Sep 5, 2008, at 8:47 PM, Damon Brown wrote: > Yes, I have all of the valid posts open on my security group > -d Brian West sip:[EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP, NAT and Amazon EC2
Ive Tried the following with no success: internal.xml Ive also tried just changing the vars.xml file. I also tried forwarding the incoming rtp connections to my test pc. all with no audio success. I am sure im doing something wrong I jsut cant find it. I found another suggestion on the wiki of creating a "double nat" that listens on 5090. That didnt change anything either, here is that information:
Re: [Freeswitch-users] SIP, NAT and Amazon EC2
in vars.xml On Sat, Sep 6, 2008 at 1:21 AM, Diego Viola <[EMAIL PROTECTED]> wrote: > Try changing these lines and put your ip instead, that worked for me. > > > > > Diego > > On Sat, Sep 6, 2008 at 1:09 AM, Damon Brown <[EMAIL PROTECTED]> wrote: >> Ive Tried the following with no success: >> >> internal.xml >> >> >> >> >> Ive also tried just changing the vars.xml file. I also tried forwarding the >> incoming rtp connections to my test pc. all with no audio success. I am >> sure im doing something wrong I jsut cant find it. >> >> I found another suggestion on the wiki of creating a "double nat" that >> listens on 5090. That didnt change anything either, here is that >> information: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >>> From: "Brian West" <[EMAIL PROTECTED]> >> Sent: Friday, September 5, 2008 6:57pm >> To: freeswitch-users@lists.freeswitch.org >> Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2 >> >> You'll have to set an ext-sip-ip and ext-rtp-ip on the internal.xml >> profile on ec2 duplicate them from the external profile. >> >> /b >> >> On Sep 5, 2008, at 8:47 PM, Damon Brown wrote: >> >>> Yes, I have all of the valid posts open on my security group >>> -d >> >> Brian West >> sip:[EMAIL PROTECTED] >> >> >> >> >> >> >> >> ___ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> ___ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP, NAT and Amazon EC2
Try changing these lines and put your ip instead, that worked for me. Diego On Sat, Sep 6, 2008 at 1:09 AM, Damon Brown <[EMAIL PROTECTED]> wrote: > Ive Tried the following with no success: > > internal.xml > > > > > Ive also tried just changing the vars.xml file. I also tried forwarding the > incoming rtp connections to my test pc. all with no audio success. I am > sure im doing something wrong I jsut cant find it. > > I found another suggestion on the wiki of creating a "double nat" that > listens on 5090. That didnt change anything either, here is that information: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > From: "Brian West" <[EMAIL PROTECTED]> > Sent: Friday, September 5, 2008 6:57pm > To: freeswitch-users@lists.freeswitch.org > Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2 > > You'll have to set an ext-sip-ip and ext-rtp-ip on the internal.xml > profile on ec2 duplicate them from the external profile. > > /b > > On Sep 5, 2008, at 8:47 PM, Damon Brown wrote: > >> Yes, I have all of the valid posts open on my security group >> -d > > Brian West > sip:[EMAIL PROTECTED] > > > > > > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP, NAT and Amazon EC2
Let me launch mine in 32bit and see what I can do over the weekend... I had this working without a problem. ;/ /b On Sep 6, 2008, at 12:09 AM, Damon Brown wrote: > Ive Tried the following with no success: > > internal.xml > > > > > Ive also tried just changing the vars.xml file. I also tried > forwarding the incoming rtp connections to my test pc. all with no > audio success. I am sure im doing something wrong I jsut cant find > it. > > I found another suggestion on the wiki of creating a "double nat" > that listens on 5090. That didnt change anything either, here is > that information: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > From: "Brian West" <[EMAIL PROTECTED]> > Sent: Friday, September 5, 2008 6:57pm > To: freeswitch-users@lists.freeswitch.org > Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2 > > You'll have to set an ext-sip-ip and ext-rtp-ip on the internal.xml > profile on ec2 duplicate them from the external profile. > > /b > > On Sep 5, 2008, at 8:47 PM, Damon Brown wrote: > >> Yes, I have all of the valid posts open on my security group >> -d > > Brian West > sip:[EMAIL PROTECTED] > > > > > > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West sip:[EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP, NAT and Amazon EC2
Thanks Diego, I have : and im still having no luck my guess would be some issue with the elastic ip translation? then again im drawing at newbie straws im good with the others and nat/rtp have never been an issue ... is here. -Original Message- From: "Diego Viola" <[EMAIL PROTECTED]> Sent: Friday, September 5, 2008 10:21pm To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2 Try changing these lines and put your ip instead, that worked for me. Diego On Sat, Sep 6, 2008 at 1:09 AM, Damon Brown <[EMAIL PROTECTED]> wrote: > Ive Tried the following with no success: > > internal.xml > > > > > Ive also tried just changing the vars.xml file. I also tried forwarding the > incoming rtp connections to my test pc. all with no audio success. I am > sure im doing something wrong I jsut cant find it. > > I found another suggestion on the wiki of creating a "double nat" that > listens on 5090. That didnt change anything either, here is that information: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > From: "Brian West" <[EMAIL PROTECTED]> > Sent: Friday, September 5, 2008 6:57pm > To: freeswitch-users@lists.freeswitch.org > Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2 > > You'll have to set an ext-sip-ip and ext-rtp-ip on the internal.xml > profile on ec2 duplicate them from the external profile. > > /b > > On Sep 5, 2008, at 8:47 PM, Damon Brown wrote: > >> Yes, I have all of the valid posts open on my security group >> -d > > Brian West > sip:[EMAIL PROTECTED] > > > > > > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP, NAT and Amazon EC2
Great thanks ... i look forward to your results. I installed on a deb ARI -Original Message- From: "Brian West" <[EMAIL PROTECTED]> Sent: Friday, September 5, 2008 10:23pm To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2 Let me launch mine in 32bit and see what I can do over the weekend... I had this working without a problem. ;/ /b On Sep 6, 2008, at 12:09 AM, Damon Brown wrote: > Ive Tried the following with no success: > > internal.xml > > > > > Ive also tried just changing the vars.xml file. I also tried > forwarding the incoming rtp connections to my test pc. all with no > audio success. I am sure im doing something wrong I jsut cant find > it. > > I found another suggestion on the wiki of creating a "double nat" > that listens on 5090. That didnt change anything either, here is > that information: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > From: "Brian West" <[EMAIL PROTECTED]> > Sent: Friday, September 5, 2008 6:57pm > To: freeswitch-users@lists.freeswitch.org > Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2 > > You'll have to set an ext-sip-ip and ext-rtp-ip on the internal.xml > profile on ec2 duplicate them from the external profile. > > /b > > On Sep 5, 2008, at 8:47 PM, Damon Brown wrote: > >> Yes, I have all of the valid posts open on my security group >> -d > > Brian West > sip:[EMAIL PROTECTED] > > > > > > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West sip:[EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Scheduled hangup from javascript
Hi All,
How to use scheduled hangup from within a java script.
Here is the flow.
User dials 1212 from softphone call goes to default dialplan and calls
test.js
in test.js i am doing like this
session.answer();
//...
//Do some stuff like ivr play etc ...
//..
var dialstr = "sofia/external/[EMAIL PROTECTED]";
// now originate a new session
new_session = new Session();
new_session.setCallerData("caller_id_name", "Vidhya");
new_session.setCallerData("caller_id_number", "91114050");
new_session.originate(session, dialstr,60);
bridge(session, new_session);
new_session.hangup();
session.hangup();
===
Now i want to hangup new_session after lets say 2 minutes.
I can not use scheduled hangup in dialplan as this hangup time may vary
and will come from database.
Any help will be appreciated.
--
Thanks and Regards
Vidhya Sagar Dixit
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[Freeswitch-users] openbts
Hi, Have you seen this? http://openbts.sourceforge.net/background.html I guess, FreeSWITCH would be better for this ;) Regards, Tamas ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
