[Freeswitch-users] need to get some basics right

2009-06-19 Thread xbipin

im just confused about certains tuff in freeswitch, firstly if we use TLS for
SIP then FS can read such packets but if it were to act as gateway can it
communicate in plain SIP, so its like phone connected to FS using TLS and
then from FS to gateway in plain SIP?
secondly, if phone uses SRTP, can FS take that media stream and convert to
plain RTP so it can forward it to remote Gateway, i know it cant do for
G723, G729 due to license issues but can it do for the other codecs?

my main concern is can it actually act as a encryption and decryption server
by remaining between the gateway and phone?
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Re: [Freeswitch-users] is there any available gui for freeswitch using cake php?

2009-06-16 Thread xbipin

hi,

if u need any help, i can always provide that.

Regards,
Bipin



Diego Viola wrote:
 
 Sure, I will let you know when it's done.
 
 On Tue, Jun 16, 2009 at 5:26 AM, Edmar Cruzdarklio...@yahoo.com wrote:

 Thanks for that info... Can you send me this project if and only if it is
 already finished on this email darkl...@yahoo.com? Thanks a lot...


 Diego Viola wrote:

 I'm currently rewriting the entire thing, it was a commercial app
 first, but I'm re-writing it in order to make it open source. It's not
 ready yet, as soon as I finish it, I will release it to the public.

 Diego

 On Mon, Jun 15, 2009 at 11:06 PM, Edmar Cruzdarklio...@yahoo.com
 wrote:

 Can you share me the link of it so i can try... Please

 Diego Viola wrote:

 I'm currently writing a rails app that uses mod_nibblebill for
 billing,
 it's
 a calling card app.

 Diego

 On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz darklio...@yahoo.com
 wrote:


 Yup tcapi is a great cake php GUI for freeswitch but it is not yet
 fully
 developed...
 Is there any GUI with billing options?


 seven-8 wrote:
 
  http://www.tcapi.org/index.php?title=Main_Page
 
 
 
  On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote:
 
 
  is there any available gui for freeswitch using cake php complete
  instead of
  wikipbx, spice softphone or pfsense?
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Re: [Freeswitch-users] FXO cards for freeswitch on windows

2009-04-23 Thread xbipin

voxilla, linksys etc all forums i have already tried, firstly very few ppl r
using it here and those who r, non of them have figured out how to get
caller id running, as a matter of fact i was one of them to discover a good
disconnect tone to work for the spa3102 and also have posted on voxilla. the
issue is the caller id is sent after first ring and FSK type  but the caller
id method listed in it doesnt work any of it in UAE, what i know is its
supposed to be ETSI FSK but setting that also doesnt work, some1 in voxilla
posted it works for it setting it at bellcore only but once in 5 times only
it works fine for him. the other issue is the telephone company wont release
any technical details also, mayb some in the telecom industry might be able
to help only. SPA400 works for sure but i told linksys about it also but
they wont produce a similar firmware update for spa3102 at all.

the sangoma card, does it work for sure and secondly can any1 ship it to
UAE?



Fred-145 wrote:
 
 
 xbipin wrote:
 i was just thinking of throwing away my spa3102 and setting my windows
 machine with FS on it to conenct to the telephone line for incomming
 calls, something like a calling card gateway using a FXO card. the reason
 being from the time i bought the spa3102, i haven't been able to get the
 caller id on it to work with the specs of my country
 
 Before dumping the 3102, you could ask in the Voxilla forum, and see if
 someone knows how to get it to receive CID in the UAE.
 
 If it still doesn't work, as far as I know, Sangoma is the only solution
 for Windows, using the OpenZAP interface (wanpipe):
 www.sangoma.com/products_and_solutions/hardware/analog_telephony/B600_Analog_Voice_Card.html
 
 I think the USB device doesn't work with Windows yet:
 www.sangoma.com/products_and_solutions/hardware/analog_telephony/usb_fxo.html
 
 Otherwise, with single-FXO PCI cards from OpenVox selling for $80, it's
 too bad there's still no way to use those in Windows :-/
 

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Re: [Freeswitch-users] FXO cards for freeswitch on windows

2009-04-23 Thread xbipin

some docs i found for the telecom company and this is whats written about
caller id

Products which offer CLI, should recognize DTMF signalling in accordance
with the following standard: 

ETSI ES 201 235-3   
Access and terminals (AT) specification of Dual –Tone Multi Frequency (DTMF)
Transmitters and receivers; Part 3 Receivers. 




Michael Jerris wrote:
 
 That all being said, I am sure we will need to make modifications to  
 make your caller id work as well, are there technical details available?
 
 Mike
 
 On Apr 23, 2009, at 6:58 AM, Fred-145 wrote:
 


 xbipin wrote:
 i was just thinking of throwing away my spa3102 and setting my  
 windows
 machine with FS on it to conenct to the telephone line for incomming
 calls, something like a calling card gateway using a FXO card. the  
 reason
 being from the time i bought the spa3102, i haven't been able to  
 get the
 caller id on it to work with the specs of my country

 Before dumping the 3102, you could ask in the Voxilla forum, and see  
 if
 someone knows how to get it to receive CID in the UAE.

 If it still doesn't work, as far as I know, Sangoma is the only  
 solution for
 Windows, using the OpenZAP interface (wanpipe):
 www.sangoma.com/products_and_solutions/hardware/analog_telephony/B600_Analog_Voice_Card.html

 I think the USB device doesn't work with Windows yet:
 www.sangoma.com/products_and_solutions/hardware/analog_telephony/usb_fxo.html

 Otherwise, with single-FXO PCI cards from OpenVox selling for $80,  
 it's too
 bad there's still no way to use those in Windows :-/
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[Freeswitch-users] FXO cards for freeswitch on windows

2009-04-22 Thread xbipin

i was just thinking of throwing away my spa3102 and setting my windows
machine with FS on it to conenct to the telephone line for incomming calls,
something like a calling card gateway using a FXO card.
the reason being from the time i bought the spa3102, i haven't been able to
get the caller id on it to work with the specs of my country and the linksys
ppl also have no clue about it nor do they provide any help on it so im just
sick and tired of it so i wanted to know if any single port FXO cards work
with FS or no, if so which ones and that too on a windows machine.

and secondly if any1 of the sellers on this mailing list could ship it to my
location coz its not available in my country.

i read somewhere that a normal 56k dialup modem can be sued like a FXO card,
any ideas on it?
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Re: [Freeswitch-users] FXO cards for freeswitch on windows

2009-04-22 Thread xbipin

the issue was, its banned in the country i live, UAE as voip is considered
illegal here so i had bought it from india and i never got any file or
documentation from the company directly where i bought it from and i tried
like more than 200 different regional settings and i also set it up how UAEs
settings would be but for some reason it wouldn't receive caller id inspite
of the line having callerid functionality.
secondly the issue was the caller id type used in UAE is a bit different and
i guess that can also be one reason as its FSK, sent after first ring
between the first and second ring and the bellcore or some other technial
detail is a bit different.
end of the day it doesnt work, mayb some1 good at it can make it work but i
havent got any help from the company as well as their forum or anything like
that.


Mike Fedyk wrote:
 
 Most lkely would be sangoma products since they have a windows driver.
 
 As for the callerid you will only be able to receive it with analog
 ports. Was that not working with the line in your country?
 
 You should be able to get the provisioning file documentation from the
 reseller you purchased the ata from.
 
 Did you try changing the regional settings to match the analog
 signalling used in your country?
 
 On 4/22/09, xbipin bi...@xbipin.com wrote:

 i was just thinking of throwing away my spa3102 and setting my windows
 machine with FS on it to conenct to the telephone line for incomming
 calls,
 something like a calling card gateway using a FXO card.
 the reason being from the time i bought the spa3102, i haven't been able
 to
 get the caller id on it to work with the specs of my country and the
 linksys
 ppl also have no clue about it nor do they provide any help on it so im
 just
 sick and tired of it so i wanted to know if any single port FXO cards
 work
 with FS or no, if so which ones and that too on a windows machine.

 and secondly if any1 of the sellers on this mailing list could ship it to
 my
 location coz its not available in my country.

 i read somewhere that a normal 56k dialup modem can be sued like a FXO
 card,
 any ideas on it?
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Re: [Freeswitch-users] upper registration in FS?

2009-04-17 Thread xbipin

hi,

i was trying to do the stuff i had written above but seems a bit too tough
so on giving it a second though, the reason i need to do the above was each
client id is basically billed on my voipswitch so a way around it could be
was the client id and pass on voipswitch for all clients, i can create them
on FS so users can register to FS using the same id and pass, now in the FS
provider list, i can create provider accounts for each client account, now
the issue is, how can i have a dialplan such that to route all calls from
id1/pass1 to provider1  and id2/pass2 to provider2.

any help on it?

regards,
Bipin



Jason White-14 wrote:
 
 xbipin bi...@xbipin.com wrote:
 
 any1 have any idea how what to sue in dialplan such that calls from a
 single
 id go to a specific gateway only with blind registration enabled, this is
 the only major issue im having.
 
 Perhaps you could match the source address in the dial plan and then
 bridge or
 redirect the call to the desired gateway.
 
 condition field=network_addr expression=^192.168.140.5$
 for example.
 
 I tested a similar example once and it did work.
 
 
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Re: [Freeswitch-users] upper registration in FS?

2009-04-17 Thread xbipin

does it only work for registrations only or does it limit the user to calling
through that gateway only?



Anthony Minessale-2 wrote:
 
 It has not been used much but there was a way to add gateways to a user
 tag
 in the directory
 so if you register locally to FS to that user then it will register that
 gateway and unreg it when he unregisters.
 I don't have time to dig it up but the params are in there somewhere.
 
 
 On Fri, Apr 17, 2009 at 5:08 AM, xbipin bi...@xbipin.com wrote:
 

 hi,

 i was trying to do the stuff i had written above but seems a bit too
 tough
 so on giving it a second though, the reason i need to do the above was
 each
 client id is basically billed on my voipswitch so a way around it could
 be
 was the client id and pass on voipswitch for all clients, i can create
 them
 on FS so users can register to FS using the same id and pass, now in the
 FS
 provider list, i can create provider accounts for each client account,
 now
 the issue is, how can i have a dialplan such that to route all calls from
 id1/pass1 to provider1  and id2/pass2 to provider2.

 any help on it?

 regards,
 Bipin



 Jason White-14 wrote:
 
  xbipin bi...@xbipin.com wrote:
 
  any1 have any idea how what to sue in dialplan such that calls from a
  single
  id go to a specific gateway only with blind registration enabled, this
 is
  the only major issue im having.
 
  Perhaps you could match the source address in the dial plan and then
  bridge or
  redirect the call to the desired gateway.
 
  condition field=network_addr expression=^192.168.140.5$
  for example.
 
  I tested a similar example once and it did work.
 
 
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 -- 
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 FreeSWITCH http://www.freeswitch.org/
 ClueCon http://www.cluecon.com/
 
 AIM: anthm
 MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
 IRC: irc.freenode.net #freeswitch
 
 FreeSWITCH Developer Conference
 sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
 iax:gu...@conference.freeswitch.org/888
 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
 pstn:213-799-1400
 
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Re: [Freeswitch-users] upper registration in FS?

2009-04-03 Thread xbipin

hi,

any1 have any idea how what to sue in dialplan such that calls from a single
id go to a specific gateway only with blind registration enabled, this is
the only major issue im having.


Regards,
Bipin
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Re: [Freeswitch-users] upper registration in FS?

2009-04-02 Thread xbipin

hi,

i wanted to know if there was any way to actually accept all registrations
coming towards freeswitch, the normal function is to have all the suerid and
passwords configured, but is there a way to accept all registrations coming
towards a single ip or domain?


Regards,
Bipin
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Re: [Freeswitch-users] FS and Skypiax on Windows Video How To

2009-03-31 Thread xbipin

hi,

is it just audio or is it that im having broken codecs so cant view any
video?

Regards,
Bipin



Giovanni Maruzzelli-3 wrote:
 
 Kulwinder Singh contributed this HOW TO: Freeswitch  Skype- OS
 Microsoft Windows
 Download 118MB HD: http://www.celliax.org/final.avi
 
 Sincerely,
 
 Giovanni Maruzzelli
 =
 www.celliax.org
 via Pierlombardo 9, 20135 Milano
 Italy
 gmaruzz at celliax dot org
 Cell : +39-347-2665618
 Fax : +39-02-87390039
 
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Re: [Freeswitch-users] how to do upper registration

2009-03-30 Thread xbipin

so what i make of it is that FS cant do upper registration or lets say it
wont proxy everything and wont use the registration details it receives from
clients registering to it and use it to register clients to a different
server for which its proxying, or mayb its not meant to do so but then i ask
can FS be configured to accept all registrations coming to it without each
userid and password being defined in the profiles or can FS be configured to
simply accept all incomming calls without the need for clients to register
to it in which case it will fullfill my needs partially.

or one more thing is use the id and pass sent by client to be forwarded to
my actual server and then its up to the server to authenticate it and
connect the call or to reply with incorrect id and pass. this can be done
only if FS could accept all registrations without it being defined in its
profile and some how the dialplan was crafted to use the registration id and
pass sent by client to be used as the id and pass for the SIP TRUNK or the
server in general.
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[Freeswitch-users] live iso image with freeswitch

2009-03-30 Thread xbipin

can any1 tell me where can i find a live cd image with the basic stuff to run
FS and FS with all it tools installed and WITH A GUI, something like a pbx
in a flash iso image so windows users like me find it easier to get testing
with FS as the support for windows SIP proxy or any SIP related tool for
windows platform is just about nil so i realized FS on windows also wont
make much sense coz the rest of the developers etc use linux for FS and if i
simply keep waiting for FS to actually do something productive on windows
platform then it might take long or forever.

if any i can provide me a live CD image with just enough tools to run FS to
its fullest coz till date i have been only using Voipswitch and its time i
need to implement TLS or any such type of encryption to reach new markets.
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Re: [Freeswitch-users] live iso image with freeswitch

2009-03-30 Thread xbipin

hi,

im really sorry if it sounded as if i was demanding, its a place where
every1 helps so i did the same thing, asked if any1 knew of any such live cd
image coz its been quiet some time and i havent reached anywhere with the
testing of FS coz the reason being, i can test it offline at my home but
that will be like a single client connecting to FS and again the issue
being, i live in a country where VoIP is blocked so cant even get FS to
register to a provider also. then how do i ever test it, so in reply i would
say i have a dedicated server and a proper VoIP setup with loads of clients
but i have been using Voipswitch on windows till date and they got a very
simple form of encryption which helps me bypass blocks and do the selling as
well as testing, so now if i install FS on my server then how can i still
test it in my existing setup coz the TLS, as discussed, is currently not
functional on the windows platform but works on linux which is like learning
a new thing from scratch, not that i cant, but will take up a lot of my time
to reach to a level where a normal VoIP user on linux is currently compared
to a windows user like me.

in this part of the world, there r many guys selling Voip etc but all they
know is of voipswitch as windows is what ppl know in this region. mayb i
might be the only one from the whole region mostly using and promoting open
source softwares like pfsense, opensbc and many such stuff and also might be
the only one who knows there r things like asterisk and freeswitch. i
searched like 75 outlets to get asterisk cards and all i got for an answer
is, what is asterisk.

living in this region and being able to test and try open source softwares
is not easy coz never 2 guys meet and end up talking something about linux,
pbx, softswitches etc coz no1 has ever heard of such things other than the
stuff whats running in the economy.



Brian West-3 wrote:
 
 
 On Mar 30, 2009, at 9:33 AM, xbipin wrote:
 can any1 tell me where can i find a live cd image with the basic  
 stuff to run
 FS and FS with all it tools installed and WITH A GUI, something like  
 a pbx
 in a flash iso image so windows users like me find it easier to get  
 testing
 with FS as the support for windows SIP proxy or any SIP related tool  
 for
 windows platform is just about nil so i realized FS on windows also  
 wont
 make much sense coz the rest of the developers etc use linux for FS  
 and if i
 simply keep waiting for FS to actually do something productive on  
 windows
 platform then it might take long or forever.
 
 Well this is a tall order... You know people ask for it.. or shall I  
 say demand it... but nobody really steps up to help out at all on the  
 GUI requests.
 
 FreeSWITCH on windows is equally capable minus a couple of things like  
 TLS since nobody will actually DO the work required to make it  
 happen.  This isn't a buffet where you pull up and demand things be  
 one way or the other... this is a community where you start helping.   
 I would love to see more helping and less demanding!
 
 if any i can provide me a live CD image with just enough tools to  
 run FS to
 its fullest coz till date i have been only using Voipswitch and its  
 time i
 need to implement TLS or any such type of encryption to reach new  
 markets.
 
 
 You could follow the linux how to and install CentOS and be done  
 already.
 
 Brian West
 br...@freeswitch.org
 
 -- Meet us a ClueCon!  http://www.cluecon.com
 
 
 
 
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Re: [Freeswitch-users] live iso image with freeswitch

2009-03-30 Thread xbipin

hi,

i think this thread has got a bit too much of attention but in all
directions, bytheway sorry for that double posting but i think its something
to do with nabble as i get a mail saying ur post is rejected so i resend and
then i find both the posts get accepted.
UAE is very strict in terms of allowing VoIP, they block skype and any sort
of voip available but come to see it, they use all the latest gadgets to
block advanced stuff but its basic stuff that bypasses it thats y guys like
me end up selling voip and any1 dealing in UAE will only tell u one thing,
its a million dollar market for voip.
simple technic to bypass is, use any port other than the standard SIP ports
and in the SIP packet, if the header says SIP or something like that, change
it to SI_P or S_IP or whatever u like as the header and ur through the
block. known methods to bypass r openvpn but they find out and simply block
the port, XOR or CBCOM as said by grandstream, voipswitch's voiptunnel and
just about anything that doesnt say anything about SIP or RTP in the packet
header or description or whatever it may be.

bytheway that astlinux is for embedded systems etc right? i have a ALIX
board already running pfsense, im looking for a normal linux image for a
workstation but just enough tools to run freeswitch, have the gnome or K
desktop environment and later on when the web GUI is developed then simply
the web GUI.
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[Freeswitch-users] upper registration in FS?

2009-03-29 Thread xbipin

i have been trying to do this but seem to have been lost, actually i want to
use freeswitch as a session border controller so basically all the clients
that try to register to FS will actually be authenticated by voipswitch but
i want FS to be in between and proxy everything and with any of its advanced
options turned off so as to simply work as topology hiding, proxy, SBC only.
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[Freeswitch-users] how to do upper registration

2009-03-29 Thread xbipin

i have been trying to do this but seem to have been lost, actually i want to
use freeswitch as a session border controller so basically all the clients
that try to register to FS will actually be authenticated voipswitch but i
want FS to be in between and proxy everything and with any of its advanced
options turned off.
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Re: [Freeswitch-users] freeswitch as a session border controller

2009-03-28 Thread xbipin

hi,

im trying to do the same, use FS as a plain and simple SBC but cant figure
out how to do so
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[Freeswitch-users] cant access mailbox if userid is non numeric

2008-09-29 Thread xbipin

under windows, i faced the following problem with FS. if i change the user_id
to a non numeric value then the user can register to the FS but can access
his mail box by calling 1005.
as soon as i change the user_id to 1005 then dialing that i cana ccess my
mailbox, y is this so? y cant the user have his name as user_id rather a
number?

include
  user id=demo mailbox=1005
params
  
  
/params
variables
  variable name=accountcode value=1005/ 
  variable name=user_context value=default/
  variable name=effective_caller_id_name value=Ankur/
  variable name=effective_caller_id_number value=1005/
/variables
  /user
/include

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Re: [Freeswitch-users] cant access mailbox if userid is non numeric

2008-09-29 Thread xbipin

the dialplan is the standard config that comes with FS, no changes to it



Yossi Neiman wrote:
 
 I believe the problem might lie in the dialplan that you are using to 
 get the user into the mailbox.  If you can resubmit this section from 
 the user directory plus the snippet from the dialplan that pertains to 
 this, that would make it easier to help you.  Alternatively, you can 
 stop by the #freeswitch channel on freenode and we can help in real time 
 then.
 
 -- 
 Yossi Neiman
 Cartis Solutions, Inc.
 http://www.cartissolutions.com
 
 
 
 xbipin wrote:
 under windows, i faced the following problem with FS. if i change the
 user_id
 to a non numeric value then the user can register to the FS but can
 access
 his mail box by calling 1005.
 as soon as i change the user_id to 1005 then dialing that i cana ccess my
 mailbox, y is this so? y cant the user have his name as user_id rather a
 number?

 include
   user id=demo mailbox=1005
 params
   
   
 /params
 variables
   variable name=accountcode value=1005/ 
   variable name=user_context value=default/
   variable name=effective_caller_id_name value=Ankur/
   variable name=effective_caller_id_number value=1005/
 /variables
   /user
 /include

   
 
 -- 
 Yossi Neiman
 President
 Cartis Solutions, Inc.
 P) 630-259-8100
 http://www.cartissolutions.com
 
 
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Re: [Freeswitch-users] cant access mailbox if userid is non numeric

2008-09-29 Thread xbipin

what needs to be changed here, im a bit confused

extension name=Local_Extension
  condition field=destination_number expression=^(10[01][0-9])$
action application=set data=dialed_ext=$1/
action application=export data=dialed_ext=$1/
  /condition
  condition field=destination_number
expression=^${caller_id_number}$
action application=set 
data=voicemail_authorized=${sip_authorized}/
action application=answer/
action application=sleep data=1000/
action application=voicemail data=check default $${domain}
${dialed_ext}/
!-- bind_meta_app can have these args key [a|b|ab] [a|b|o|s] app 
--
anti-action application=bind_meta_app data=1 b s 
execute_extension::dx
XML features/
anti-action application=bind_meta_app data=2 b s
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav/
anti-action application=bind_meta_app data=3 b s 
execute_extension::cf
XML features/
anti-action application=set data=transfer_ringback=${us-ring}/
anti-action application=set data=call_timeout=30/
!-- anti-action application=set
data=sip_exclude_contact=${network_addr}/ --
anti-action application=set data=hangup_after_bridge=true/
!--anti-action application=set
data=continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION/
--
anti-action application=set data=continue_on_fail=true/
anti-action application=db
data=insert/call_return/${dialed_ext}/${caller_id_number}/
anti-action application=db
data=insert/last_dial_ext/${dialed_ext}/${uuid}/
anti-action application=bridge data=user/[EMAIL PROTECTED]/
anti-action application=answer/
!--anti-action application=send_display data=Voicemail for
${dialed_ext}/--
anti-action application=sleep data=1000/
anti-action application=voicemail data=default $${domain}
${dialed_ext}/
  /condition
/extension




Brian West-3 wrote:
 
 You can also override this via the ${sip_mailbox} variable instead of  
 the dialed_ext that is a variable that is magically set for you based  
 on the mailbox = attr.
 
 /b
 
 On Sep 29, 2008, at 9:55 AM, xbipin wrote:
 

 the dialplan is the standard config that comes with FS, no changes  
 to it



 Yossi Neiman wrote:

 I believe the problem might lie in the dialplan that you are using to
 get the user into the mailbox.  If you can resubmit this section from
 the user directory plus the snippet from the dialplan that pertains  
 to
 this, that would make it easier to help you.  Alternatively, you can
 stop by the #freeswitch channel on freenode and we can help in real  
 time
 then.

 -- 
 Yossi Neiman
 Cartis Solutions, Inc.
 http://www.cartissolutions.com



 xbipin wrote:
 under windows, i faced the following problem with FS. if i change  
 the
 user_id
 to a non numeric value then the user can register to the FS but can
 access
 his mail box by calling 1005.
 as soon as i change the user_id to 1005 then dialing that i cana  
 ccess my
 mailbox, y is this so? y cant the user have his name as user_id  
 rather a
 number?

 include
  user id=demo mailbox=1005
params


/params
variables
  variable name=accountcode value=1005/
  variable name=user_context value=default/
  variable name=effective_caller_id_name value=Ankur/
  variable name=effective_caller_id_number value=1005/
/variables
  /user
 /include



 -- 
 Yossi Neiman
 President
 Cartis Solutions, Inc.
 P) 630-259-8100
 http://www.cartissolutions.com


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Re: [Freeswitch-users] cant access mailbox if userid is non numeric

2008-09-29 Thread xbipin

i tried it but still doesnt work, the below is the log

the suerid is bipin and the mailbox is 1001


2008-09-29 21:07:16 [NOTICE] switch_channel.c:538 switch_channel_set_name()
New
Channel sofia/internal/[EMAIL PROTECTED]
[52b5650d-3d2f-1f44-b36b-05bcd1f9c47
d]
2008-09-29 21:07:16 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing
bip
in-1001 in context default
2008-09-29 21:07:17 [INFO] switch_ivr_async.c:1481
switch_ivr_bind_dtmf_meta_ses
sion() Bound B-Leg: 1 execute_extension::dx XML features
2008-09-29 21:07:17 [INFO] switch_ivr_async.c:1481
switch_ivr_bind_dtmf_meta_ses
sion() Bound B-Leg: 2 record_session::C:Program Files
(x86)FreeSWITCH/recordings
/bipin.2008-09-29-21-07-17.wav
2008-09-29 21:07:17 [INFO] switch_ivr_async.c:1481
switch_ivr_bind_dtmf_meta_ses
sion() Bound B-Leg: 3 execute_extension::cf XML features
2008-09-29 21:07:18 [WARNING] mod_dptools.c:1972 user_outgoing_channel()
can't f
ind user [EMAIL PROTECTED]
2008-09-29 21:07:18 [ERR] switch_ivr_originate.c:958 switch_ivr_originate()
Cann
ot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
2008-09-29 21:07:18 [INFO] mod_dptools.c:1814 audio_bridge_function()
Originate
Failed.  Cause: SUBSCRIBER_ABSENT
2008-09-29 21:07:18 [NOTICE] switch_channel.c:1426
switch_channel_perform_mark_p
re_answered() Ring-Ready sofia/internal/[EMAIL PROTECTED]
2008-09-29 21:07:18 [NOTICE] sofia_glue.c:2032 sofia_glue_tech_media()
Pre-Answe
r sofia/internal/[EMAIL PROTECTED]
2008-09-29 21:07:18 [NOTICE] mod_dptools.c:606 answer_function() Channel
[sofia/
internal/[EMAIL PROTECTED] has been answered
2008-09-29 21:07:21 [NOTICE] sofia.c:2588 sofia_handle_sip_i_state() Hangup
sofi
a/internal/[EMAIL PROTECTED] [CS_EXECUTE] [NORMAL_CLEARING]
2008-09-29 21:07:21 [ERR] mod_sndfile.c:175 sndfile_file_open() Error
Opening Fi
le [C:\Program Files
(x86)\FreeSWITCH\storage\voicemail\default\sip.xbipin.com\(
null)\msg_52b5650d-3d2f-1f44-b36b-05bcd1f9c47d.wav] [System error : The
system c
annot find the file specified.
]
2008-09-29 21:07:21 [NOTICE] switch_core_session.c:812
switch_core_session_threa
d() Session 3 (sofia/internal/[EMAIL PROTECTED]) Ended
2008-09-29 21:07:21 [NOTICE] switch_core_session.c:814
switch_core_session_threa
d() Close Channel sofia/internal/[EMAIL PROTECTED] [CS_HANGUP]







Brian West-3 wrote:
 
 You have some options here.  I highly recommend you check out the  
 changes I have done to the default configs btw.
 
 anti-action application=voicemail data=default $${domain} $ 
 {user_data([EMAIL PROTECTED] param real_mailbox)}/
 
 And just set the param of the real_mailbox for that user and it'll go  
 snag it out of the directory for that user.
 
 Same for the check version:
 
 action application=voicemail data=check default $${domain} $ 
 {user_data([EMAIL PROTECTED] param real_mailbox)}/
 
 Now if you really want to get crazy you can do something like this:
 
 action application=set data=real_mailbox=${user_data(${dialed_ext}@ 
 $${domain} param real_mailbox)}/
 action application=voicemail data=default $${domain} ${cond($ 
 {real_mailbox} ==  ? ${real_mailbox} : ${dialed_ext})}/
 
 That would use the real_mailbox param on the user or the dialed_ext...  
 I didn't test that but it looks correct.
 
 Chances are I have done nothing but confuse you.  :(
 
 /b
 
 
 
 On Sep 29, 2008, at 11:19 AM, xbipin wrote:
 

 what needs to be changed here, im a bit confused

extension name=Local_Extension
  condition field=destination_number expression=^(10[01][0-9]) 
 $
  action application=set data=dialed_ext=$1/
  action application=export data=dialed_ext=$1/
  /condition
  condition field=destination_number
 expression=^${caller_id_number}$
  action application=set data=voicemail_authorized=$ 
 {sip_authorized}/
  action application=answer/
  action application=sleep data=1000/
  action application=voicemail data=check default $${domain}
 ${dialed_ext}/
  !-- bind_meta_app can have these args key [a|b|ab] [a|b|o|s]  
 app --
  anti-action application=bind_meta_app data=1 b s  
 execute_extension::dx
 XML features/
  anti-action application=bind_meta_app data=2 b s
 record_session::$${base_dir}/recordings/${caller_id_number}.$ 
 {strftime(%Y-%m-%d-%H-%M-%S)}.wav/
  anti-action application=bind_meta_app data=3 b s  
 execute_extension::cf
 XML features/
  anti-action application=set data=transfer_ringback=${us-ring}/
  anti-action application=set data=call_timeout=30/
  !-- anti-action application=set
 data=sip_exclude_contact=${network_addr}/ --
  anti-action application=set data=hangup_after_bridge=true/
  !--anti-action application=set
 data 
 = 
 continue_on_fail 
 = 
 NORMAL_TEMPORARY_FAILURE 
 ,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION/
 --
  anti-action application=set data=continue_on_fail=true/
  anti-action application=db
 data=insert/call_return/${dialed_ext}/${caller_id_number}/
  anti-action application=db
 data=insert

Re: [Freeswitch-users] cant access mailbox if userid is non numeric

2008-09-29 Thread xbipin

im not using it by myself but actually my softphone calls my server whicha
ctually has openSBC as my session border controller which is in upper
registration mode so its proxys all the calls and then passes it over to FS
so that might be doing it. from my dialer i simply call 1001.
if i dont use opensbc then i cant use voip at all coz my isp blocks voip
calls and opensbc does the needed encryption and decryption.

is there anyway round it?



Brian West-3 wrote:
 
 Why are you calling with the @sign?  REMOVE THAT.
 
 
 /b
 
 On Sep 29, 2008, at 12:09 PM, xbipin wrote:
 
 Channel sofia/internal/[EMAIL PROTECTED]
 
 
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Re: [Freeswitch-users] Freeswitch billing

2008-09-24 Thread xbipin

cant seem to be able to register that that site at all, tired multiple times
but i never get the registration details in my mailbox, the id gets
registered though



Darren Wiebe wrote:
 
 Can we move this to www.astpp.org?  I'm working on it and it is being 
 used.  The Windows platform should be not that hard.  I'd really like to 
 move this to www.astpp.org though to keep from bugging the entire 
 Freeswitch group. 
 
 Darren Wiebe
 [EMAIL PROTECTED]
 
 xbipin wrote:
 u can try astpp, i havent been able to make it work at all with
 freeswitch on
 windows platform as little support but the author says he can make it
 work,
 im still trying.



 gineta wrote:
   
 Hi Patrick
 Is possible you post all the configurations you make from freeswitch to
 jbilling

 THanks



 Patrick-139 wrote:
 
 Cliconnect wrote:
   
  
 Sure, I'll do that.
  
 At this point I am investigating the OS ones.
  
 http://jbilling.com  (I am having trouble with java since I have 64
 bits 
 OS and it is not working)
 
 I just downloaded jbilling on a laptop with Fedora 9 x86_64 which has 
 the Open Source java implementation called icedtea. I followed the 
 installation instructions in jbilling_installation_instructions.pdf and 
 fired up jbilling with sudo sh bin/run.sh. Had no problems starting it 
 and logging in.

 Did the same on a CentOS 5.2 x86 box with Sun Java JDK version 1.6.0
 and 
 also did not have a problem.

 Hope this helps.

 Regards,
 Patrick

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[Freeswitch-users] call bridge not sending hangup or call rejected to priginating party when destinaion cancels call

2008-09-24 Thread xbipin

basically the whole scene is such


my pc running googletalk with gtalk2voip id added and configured to my
freeswitch server  freeswitch server --- gateway server


basically i originate calls from my googletalk using the gtalk2voip service,
now what happens is im able to make calls and talk also but there is a
problem when the destinaion party is called and he rejects the call which is
reported by freeswitch as user busy but this isnt passed onto the gtalk2voip
server so me using googletalk i can hear the dummy ring music till it
timesout. what actually is happening is on user busy, the external bridge is
closed but the internal isnt as user busy int passed to gtalk2voip server
but this isnt the case if i connect to freeswitch directly using a softphone
or ata.

when i connect directly using softphone then the log shows as below, its
from start of call to call rejected by destination party to softphone being
passed message of call rejected so both the ends of the bridge are cleared

2008-09-24 12:46:28 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing
bip
in-971559270058 in context default
2008-09-24 12:46:29 [NOTICE] switch_channel.c:538 switch_channel_set_name()
New
Channel sofia/external/971559270058 [7df269f7-cbfc-7248-8688-299fc3d0b04b]
2008-09-24 12:46:37 [NOTICE] switch_channel.c:1426
switch_channel_perform_mark_p
re_answered() Ring-Ready sofia/external/971559270058!
2008-09-24 12:46:37 [NOTICE] sofia.c:2226 sofia_handle_sip_i_state()
Pre-Answer
sofia/external/971559270058!
2008-09-24 12:46:37 [INFO] mod_sofia.c:1085 sofia_receive_message() Asked to
sen
d early media by sofia/internal/[EMAIL PROTECTED]
2008-09-24 12:46:37 [NOTICE] switch_channel.c:1426
switch_channel_perform_mark_p
re_answered() Ring-Ready sofia/internal/[EMAIL PROTECTED]
2008-09-24 12:46:37 [NOTICE] mod_sofia.c:1129 sofia_receive_message()
Pre-Answer
 sofia/internal/[EMAIL PROTECTED]
2008-09-24 12:46:55 [NOTICE] sofia.c:2588 sofia_handle_sip_i_state() Hangup
sofi
a/external/971559270058 [CS_EXCHANGE_MEDIA] [USER_BUSY]
2008-09-24 12:46:55 [NOTICE] switch_ivr_bridge.c:379 audio_bridge_thread()
Hangu
p sofia/internal/[EMAIL PROTECTED] [CS_EXECUTE] [USER_BUSY]
2008-09-24 12:46:55 [NOTICE] switch_core_session.c:812
switch_core_session_threa
d() Session 20 (sofia/external/971559270058) Ended
2008-09-24 12:46:55 [NOTICE] switch_core_session.c:814
switch_core_session_threa
d() Close Channel sofia/external/971559270058 [CS_HANGUP]
2008-09-24 12:46:55 [NOTICE] switch_core_session.c:812
switch_core_session_threa
d() Session 19 (sofia/internal/[EMAIL PROTECTED]) Ended
2008-09-24 12:46:55 [NOTICE] switch_core_session.c:814
switch_core_session_threa
d() Close Channel sofia/internal/[EMAIL PROTECTED] [CS_HANGUP]




when calling from gtalk2voip id in googletalk, below is what happens

2008-09-24 12:44:22 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing
919
825967120-971559270058 in context default
2008-09-24 12:44:23 [NOTICE] switch_channel.c:538 switch_channel_set_name()
New
Channel sofia/external/971559270058 [358bb7a0-fcda-af41-b62d-3d6b91fbe212]
2008-09-24 12:44:30 [NOTICE] switch_channel.c:1426
switch_channel_perform_mark_p
re_answered() Ring-Ready sofia/external/971559270058!
2008-09-24 12:44:30 [NOTICE] sofia.c:2226 sofia_handle_sip_i_state()
Pre-Answer
sofia/external/971559270058!
2008-09-24 12:44:30 [INFO] mod_sofia.c:1085 sofia_receive_message() Asked to
sen
d early media by sofia/internal/[EMAIL PROTECTED]
2008-09-24 12:44:31 [NOTICE] switch_channel.c:1426
switch_channel_perform_mark_p
re_answered() Ring-Ready sofia/internal/[EMAIL PROTECTED]
2008-09-24 12:44:31 [NOTICE] mod_sofia.c:1129 sofia_receive_message()
Pre-Answer
 sofia/internal/[EMAIL PROTECTED]
2008-09-24 12:44:35 [NOTICE] sofia.c:2588 sofia_handle_sip_i_state() Hangup
sofi
a/external/971559270058 [CS_EXCHANGE_MEDIA] [USER_BUSY]

THEN THE USER BUSY ISNT PASSED TO GTALK2VOIP SERVER SO FOR ME IN GOOGLETALK,
THE CALL NEVER ENDS TILL IT TIMEOUTS AND THEN FREESWITCH REPORTS THE BELOW
LOG

2008-09-24 12:45:02 [NOTICE] sofia.c:2588 sofia_handle_sip_i_state() Hangup
sofi
a/internal/[EMAIL PROTECTED] [CS_EXECUTE] [ORIGINATOR_CANCEL]




and also is it possible to open the audio channel to originator as soon as
rtp is received from destination either it being call processing or whatever
so the originator can hear any provider messages etc that are passed by the
destination?

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Re: [Freeswitch-users] Freeswitch billing

2008-09-23 Thread xbipin

u can try astpp, i havent been able to make it work at all with freeswitch on
windows platform as little support but the author says he can make it work,
im still trying.



gineta wrote:
 
 Hi Patrick
 Is possible you post all the configurations you make from freeswitch to
 jbilling
 
 THanks
 
 
 
 Patrick-139 wrote:
 
 Cliconnect wrote:
  
 Sure, I'll do that.
  
 At this point I am investigating the OS ones.
  
 http://jbilling.com  (I am having trouble with java since I have 64 bits 
 OS and it is not working)
 
 I just downloaded jbilling on a laptop with Fedora 9 x86_64 which has 
 the Open Source java implementation called icedtea. I followed the 
 installation instructions in jbilling_installation_instructions.pdf and 
 fired up jbilling with sudo sh bin/run.sh. Had no problems starting it 
 and logging in.
 
 Did the same on a CentOS 5.2 x86 box with Sun Java JDK version 1.6.0 and 
 also did not have a problem.
 
 Hope this helps.
 
 Regards,
 Patrick
 
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[Freeswitch-users] cant record any session

2008-09-21 Thread xbipin

i just copied the dialplan given in wiki for record session but cant record
or lets say first i cant initiate the call only, sofia says codec not found
as soon as i hit dial on the dialer

can some1 help in the xml as im trying to record calls made from a specific
user whose account is created only and not in general for every1
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Re: [Freeswitch-users] cant record any session

2008-09-21 Thread xbipin

ok i marked bypass media as well as proxy media to default which was like
commented, means marked as comment. With proxy media to enabled i used to
get the error of cant find codec but after making it all to default config,
im now getting this

2008-09-22 00:07:46 [ERR] mod_sndfile.c:175 sndfile_file_open() Error
Opening Fi
le [C:Program Files
(x86)FreeSWITCH/recordings/2008-09-22-00-07-46_15194344406_b
ipin.wav] [System error : The system cannot find the path specified.
]
2008-09-22 00:07:46 [ERR] switch_ivr_async.c:794 switch_ivr_record_session()
Err
or opening C:Program Files
(x86)FreeSWITCH/recordings/2008-09-22-00-07-46_151943
44406_bipin.wav



Brian West-3 wrote:
 
 Are you using bypass media?  What exactly is the error?
 
 /b
 
 On Sep 21, 2008, at 8:29 AM, xbipin wrote:
 

 i just copied the dialplan given in wiki for record session but cant  
 record
 or lets say first i cant initiate the call only, sofia says codec  
 not found
 as soon as i hit dial on the dialer

 can some1 help in the xml as im trying to record calls made from a  
 specific
 user whose account is created only and not in general for every1
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Re: [Freeswitch-users] cant record any session

2008-09-21 Thread xbipin

file doesnt exist but doesnt it need to create it by itself as u never know
whose gonna call when and where.
basically im looking for it to create the file and record in it so i can get
a different file for each user, whats the point creating the file and then
letting it record to it



Michael Jerris wrote:
 
 
 On Sep 21, 2008, at 4:11 PM, xbipin wrote:
 

 ok i marked bypass media as well as proxy media to default which was  
 like
 commented, means marked as comment. With proxy media to enabled i  
 used to
 get the error of cant find codec but after making it all to default  
 config,
 im now getting this
 
 Correct, you can't do anything that interacts with the media if you  
 are doing bypass or proxy modes.
 
 2008-09-22 00:07:46 [ERR] mod_sndfile.c:175 sndfile_file_open() Error
 Opening Fi
 le [C:Program Files
 (x86)FreeSWITCH/recordings/2008-09-22-00-07-46_15194344406_b
 ipin.wav] [System error : The system cannot find the path specified.
 ]
 
 Does it exist?
 

 2008-09-22 00:07:46 [ERR] switch_ivr_async.c:794  
 switch_ivr_record_session()
 Err
 or opening C:Program Files
 (x86)FreeSWITCH/recordings/2008-09-22-00-07-46_151943
 44406_bipin.wav

 
 Mike
 
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Re: [Freeswitch-users] cant record any session

2008-09-21 Thread xbipin

yes it does


Michael Jerris wrote:
 
 
 On Sep 21, 2008, at 5:00 PM, xbipin wrote:
 

 file doesnt exist but doesnt it need to create it by itself as u  
 never know
 whose gonna call when and where.
 basically im looking for it to create the file and record in it so i  
 can get
 a different file for each user, whats the point creating the file  
 and then
 letting it record to it

 
 It said Path (not file) does not exist.  Does the directory its trying  
 to create the file in exist?
 
 
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Re: [Freeswitch-users] billing platform

2008-09-20 Thread xbipin

i have a win2k3 webserver with plesk panel installed so it has all the perl
support, now the issue is the freeswitch folder has the first perl file to
run but it just wont run as all the paths in it are defined as of linux and
not windows so the basic step of getting the install to work is only the
issue, then later mayb the integration but i cant reach that stage only at
all.



Wasim Baig wrote:
 
 On Sat, Sep 20, 2008 at 2:07 AM, xbipin [EMAIL PROTECTED] wrote:
 

 whats the amount ur looking for this bounty, mayb guys like me who wanna
 use
 
 
 dwiebe can let you know more about that
 
 it as a proper softswitch can pitch in a little and can also help in
 testing
 and coding.
 
 
 what is sorely missing is thorough testing and documentation and requests
 for features
 
 for example, one cool thing that I couldn't do with asterisk and now can
 with fs and astpp is charge for codec conversion :)
 (not coded yet, but simple to add)
 
 svn co
 https://astpp.svn.sourceforge.net/svnroot/astpp/trunk[path/to/where/you/want/it]
 
 look in the freeswitch subdir and start firing away
 
 perhaps we can track this on the astpp website on the forum there OR on
 the
 astpp tracker mailing list
 so that the rest of the freeswitch people here aren't disturbed, looking
 forward to seeing you there
 
 as Mike said www.astpp.org is the place for this
 
 - wasim
 

 Darren Wiebe wrote:
 
  As the author of ASTPP I'd be very interested in negotiating a bounty
 to
  speed things up.  User authentication is working with Freeswitch as
 well
  as lcr and call rating.  It's largely a matter of documenting and
 testing.
 
  Darren Wiebe
  [EMAIL PROTECTED]
 
  xbipin wrote:
  are u interested in a paid implementation as im willing to post a
 bounty
  for
  it if there r others like us willing to use it with freeswitch.
 wikipbx
  is
  also good for a start but one thing i dont understand is if freeswitch
 is
  made for cross platform then y r other projects related to it limiting
 it
  to
  just linux?
 
 
 
  Jair Santos wrote:
 
  I've been trying to install. The problem is the lack of 
 documentation
  on
  how to integrate with FS.
 
  Jair Santos
 
 
 
 
 
 
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On
  Behalf Of xbipin
  Sent: Thursday, September 18, 2008 3:25 AM
  To: freeswitch-users@lists.freeswitch.org
  Subject: [Freeswitch-users] billing platform
 
 
 
  has any1 been able to use ASTPP billing with freeswitch and
  if so then my next question, is it possible to use it on a
  win2k3 platform and not linux as i have freeswitch running on
  a win2k3 server since a few days and works fine, i just need
  a prepaid type of script that basically stops a call once its
  credit is over. i will add the credit etc to the account manually.
 
  im ready to post a bounty if any1 willing to do it. the
  script just needs to end the call once it computes that
  remaining time is 0 based on available time which is based on
  the cost per minute and available credit and avoid calling
  again which needs to be done as soon as call is placed so the
  call is never connected if account has no credit.
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  --
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  [EMAIL PROTECTED]
 
  Aleph Communications
  www.aleph-com.net
 
 
 
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 -- 
 wasim h. baig | principal consultant | convergence pk | +92 300 8508070 |
 as
 you scope creep, so shall we reap ...
 
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Re: [Freeswitch-users] plz compile latest snapshot for windows along with msi

2008-09-19 Thread xbipin

the FS site says FS supports TLS etc so wouldnt it be good if the windows
binary were compiled with TLS as by default they r not so guys like me who
actually download the msi and install and run it can atleast have TLS
support by default on windows platform rather than them trying to make it
work and then later posting in mailing list as if its a bug etc, thatway it
can save a lot of posts that r done repeatidely and also it would be better
if ppl added to wiki as soon as they got some of their issues solved.
If i could just get the msi with TLS support then atleast i would try some
advanced features and then later post it as wiki for all those who r
beginners and willing to use FS on windows. I have already created the wiki
offline, just waiting to try out that tLS and then ill post it as a wiki
under beginners guide for windows platform.



Carlos Talbot wrote:
 
 I'm uploading the msi to the site I have write access to. It should be
 sync'd up on the files.freeswitch.org tonight. Remember to backup your
 config folder before uninstalling!
 
 Carlos
 
 
 On Mon, Sep 15, 2008 at 10:40 AM, Michael Jerris [EMAIL PROTECTED] wrote:
 

 On Sep 15, 2008, at 9:25 AM, xbipin wrote:

 
  hi,
 
  this is for the freeswitch developers, can u plz compile the latest
  snapshots of freeswitch and provide the msi installer for the latest
  snapshot coz i have been using the installer which was last compiled
  on 15
  aug and its has been quiet some time that an updated installer is
  available.
 
  I was using the last installer on my win2k3 server, works fine
  except that
  it crashes on shutdown and the sip client hears a very fast saying
  IVR when
  dialed the 5000 number and the IVR is broken up, meaning it can be
  heard but
  with some bits missing or lets say with jitter, any way to make rtp
  packets
  much smoother or audio much smoother as i was running openSBC on the
  same
  server and it routes packets pretty fine so its got nothing to do
  with the
  server or the route from the client to the server.
  --
  View this message in context:
 http://www.nabble.com/plz-compile-latest-snapshot-for-windows-along-with-msi-tp19492923p19492923.html
  Sent from the Freeswitch-users mailing list archive at Nabble.com.

 I spoke to the guy who makes the msi files yesterday and he said he
 would get a newer copy up there.  That being said, if you download
 msvc express edition and advanced installer, both free, you can make
 your own msi any time you like.


 Mike


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Re: [Freeswitch-users] billing platform

2008-09-19 Thread xbipin

whats the amount ur looking for this bounty, mayb guys like me who wanna use
it as a proper softswitch can pitch in a little and can also help in testing
and coding.


Darren Wiebe wrote:
 
 As the author of ASTPP I'd be very interested in negotiating a bounty to
 speed things up.  User authentication is working with Freeswitch as well
 as lcr and call rating.  It's largely a matter of documenting and testing.
 
 Darren Wiebe
 [EMAIL PROTECTED]
 
 xbipin wrote:
 are u interested in a paid implementation as im willing to post a bounty
 for
 it if there r others like us willing to use it with freeswitch. wikipbx
 is
 also good for a start but one thing i dont understand is if freeswitch is
 made for cross platform then y r other projects related to it limiting it
 to
 just linux?



 Jair Santos wrote:
   
 I've been trying to install. The problem is the lack of  documentation
 on
 how to integrate with FS.

 Jair Santos

  
  
  


 
 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On 
 Behalf Of xbipin
 Sent: Thursday, September 18, 2008 3:25 AM
 To: freeswitch-users@lists.freeswitch.org
 Subject: [Freeswitch-users] billing platform



 has any1 been able to use ASTPP billing with freeswitch and 
 if so then my next question, is it possible to use it on a 
 win2k3 platform and not linux as i have freeswitch running on 
 a win2k3 server since a few days and works fine, i just need 
 a prepaid type of script that basically stops a call once its 
 credit is over. i will add the credit etc to the account manually.

 im ready to post a bounty if any1 willing to do it. the 
 script just needs to end the call once it computes that 
 remaining time is 0 based on available time which is based on 
 the cost per minute and available credit and avoid calling 
 again which needs to be done as soon as call is placed so the 
 call is never connected if account has no credit.
 -- 
 View this message in context: 
 http://www.nabble.com/billing-platform-tp19549891p19549891.html
 Sent from the Freeswitch-users mailing list archive at Nabble.com.


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 [EMAIL PROTECTED]
 
 Aleph Communications
 www.aleph-com.net
 
 
 
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[Freeswitch-users] billing platform

2008-09-18 Thread xbipin

has any1 been able to use ASTPP billing with freeswitch and if so then my
next question, is it possible to use it on a win2k3 platform and not linux
as i have freeswitch running on a win2k3 server since a few days and works
fine, i just need a prepaid type of script that basically stops a call once
its credit is over. i will add the credit etc to the account manually.

im ready to post a bounty if any1 willing to do it. the script just needs to
end the call once it computes that remaining time is 0 based on available
time which is based on the cost per minute and available credit and avoid
calling again which needs to be done as soon as call is placed so the call
is never connected if account has no credit.
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Re: [Freeswitch-users] billing platform

2008-09-18 Thread xbipin

are u interested in a paid implementation as im willing to post a bounty for
it if there r others like us willing to use it with freeswitch. wikipbx is
also good for a start but one thing i dont understand is if freeswitch is
made for cross platform then y r other projects related to it limiting it to
just linux?



Jair Santos wrote:
 
 I've been trying to install. The problem is the lack of  documentation on
 how to integrate with FS.
 
 Jair Santos
 
  
  
  
 
 
 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On 
 Behalf Of xbipin
 Sent: Thursday, September 18, 2008 3:25 AM
 To: freeswitch-users@lists.freeswitch.org
 Subject: [Freeswitch-users] billing platform
 
 
 
 has any1 been able to use ASTPP billing with freeswitch and 
 if so then my next question, is it possible to use it on a 
 win2k3 platform and not linux as i have freeswitch running on 
 a win2k3 server since a few days and works fine, i just need 
 a prepaid type of script that basically stops a call once its 
 credit is over. i will add the credit etc to the account manually.
 
 im ready to post a bounty if any1 willing to do it. the 
 script just needs to end the call once it computes that 
 remaining time is 0 based on available time which is based on 
 the cost per minute and available credit and avoid calling 
 again which needs to be done as soon as call is placed so the 
 call is never connected if account has no credit.
 -- 
 View this message in context: 
 http://www.nabble.com/billing-platform-tp19549891p19549891.html
 Sent from the Freeswitch-users mailing list archive at Nabble.com.
 
 
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[Freeswitch-users] re gexp help

2008-09-17 Thread xbipin

with the following expression i cant dial any sipbroker number, eg:
*01118

^(^\*\d+)$
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Re: [Freeswitch-users] re gexp help

2008-09-17 Thread xbipin

removing the second ^ i can dial the screeming monkeys numbe which is

*266300 but not the test sipbroker number which is *01118



Ivan C Myrvold wrote:
 
 Remove the second ^ and see if that helps.
 
 Ivan
 
 Den 17. sep.. 2008 kl. 12:52 skrev xbipin:
 

 with the following expression i cant dial any sipbroker number, eg:
 *01118

 ^(^\*\d+)$
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[Freeswitch-users] cant dial through registered gateway in windows 2k3

2008-09-16 Thread xbipin

i have the following in my gateway.msl file kept under
conf/directory/default/

include
  user id=id
gateways
  gateway name=sip.server.com







  /gateway
/gateways
params
  
/params
  /user
/include

on freeswitch start it registers also and in my dialplan i have:

extension name=india
  condition field=destination_number expression=^919825967120$
  action application=bridge data=sofia/gateway/sip.server.com/$1/
  /condition
/extension

but when i call that number, its unable to bridge to the gateway and it says
route not available
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Re: [Freeswitch-users] cant dial through registered gateway in windows 2k3

2008-09-16 Thread xbipin

actually i sent the message but in the output it showed the config as blank
so i tried removing the xml format but it still showed the same so after
multiple editing it showed fine so it got submitted 6 times, sorry for that,
i didnt mean to hijack it or something like that.



Brian West-3 wrote:
 
 On Sep 16, 2008, at 9:44 AM, xbipin wrote:


extension name=india
  condition field=destination_number  
 expression=^919825967120$
  action application=bridge data=sofia/gateway/sip.server.com/ 
 $1/
  /condition
/extension
 
 Well your regular expression won't capture anything so $1 will be blank.
 
 So either wrap it in () ie ^(1234)$ or just use the number in place of  
 $1.
 
 Also what does the console say?
 
 /b
 
 PS: you do realize you sent the same email 6 times!
 
 
 
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Re: [Freeswitch-users] plz compile latest snapshot for windows along with msi

2008-09-16 Thread xbipin

thanks for the msi, its updated and works great



Michael Jerris wrote:
 
 
 On Sep 15, 2008, at 12:03 PM, xbipin wrote:
 

 i have msvc 2005 professional but im finding it hard or almsot  
 impossible to
 compile it under windows as it gives a lot of errors and msot  
 dependencies
 missing.
 i have the sdk pack also installed but its the same.
 The replacement to TLS under windows which i cant manage to get it  
 running
 and might not even with my linksys ATA, i have instaleld openSBC for  
 the
 voip encryption to bypass my block.
 i would really appreciate it if the msi was done by u guys, atleast  
 it would
 work fine compared to me compiling and trying to fix errors in it.


 
 
 If your having build errors on msvc could you please report them.   
 What dependencies are missing?
 
 Mike
 
 
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Re: [Freeswitch-users] cant dial through registered gateway in windows 2k3

2008-09-16 Thread xbipin

y do i get this warning when the first client registers to the server?

2008-09-16 19:19:07 [WARNING] sofia.c:75 sofia_handle_sip_r_notify() delete
subscriptions for failed notify
and also on shutdown udner windows, FS still crashes like usual

and also can u plz let me know if there is any wiki on the dialstring
formatting, i cant seem to find it thats y the error in dialing out of
gateway till u corrected it, thanks a lot, works fine with the change u
showed me.



Brian West-3 wrote:
 
 On Sep 16, 2008, at 9:44 AM, xbipin wrote:


extension name=india
  condition field=destination_number  
 expression=^919825967120$
  action application=bridge data=sofia/gateway/sip.server.com/ 
 $1/
  /condition
/extension
 
 Well your regular expression won't capture anything so $1 will be blank.
 
 So either wrap it in () ie ^(1234)$ or just use the number in place of  
 $1.
 
 Also what does the console say?
 
 /b
 
 PS: you do realize you sent the same email 6 times!
 
 
 
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Re: [Freeswitch-users] cant get TLS running at all

2008-08-29 Thread xbipin

when can this be done as im eagerly waiting to test out freeswitch.



Michael Jerris wrote:
 
 The biggest gotcha here is you need to have versions compiled with  
 the same compiler version as the rest of freeswitch or strange things  
 can result.  Maybe we will have to make some pre-compiled versions and  
 allow them for download so we can cleanly integrate this into the build.
 
 Mike
 
 On Aug 28, 2008, at 12:01 PM, UV wrote:
 
 I don't have those files yet
 Long story
 Will let you know when I do. ;-)

 Or even better, when I'll have a working SLN for openssl...

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of  
 xbipin
 Sent: Friday, August 29, 2008 1:44 AM
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] cant get TLS running at all


 if ur using the TLS funstion successfully then can u plz give me those
 files?


 UV-5 wrote:

 The compiled version from openssl site doesn't include the lib  
 files (I
 have
 no idea why). You'll have to compile those yourself. Unfortunately,  
 you
 can
 only do that using Perl, so it's quite a cumbersome process -  
 especially
 for
 the newbie programmer.


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of  
 xbipin
 Sent: Friday, August 29, 2008 1:10 AM
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] cant get TLS running at all


 so do i need to compile sofia or openssl as the openssl u told me,  
 i have
 the
 compiled version directly from the site for windows platform. does  
 it need
 to be re compiled or how is it all as i am a newbie in programming  
 and
 compiling



 UV-5 wrote:

 Bipin,

 I'm working on an openssl build under VS2008 but it's not easy.
 If you wish to try compiling sofia with TLS, I suggest you  
 download the
 latest openssl build (0.9.8i) and compile it (you'll need Perl to do
 that).
 If done correctly, this should give you the necessary lib files to  
 link
 with
 sofia.

 Good luck.


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 xbipin
 Sent: Thursday, August 28, 2008 7:44 PM
 To: freeswitch-users@lists.freeswitch.org
 Subject: [Freeswitch-users] cant get TLS running at all


 hi,

 actually to try out freeswitch to the fullest, i need to get it  
 running
 using TLS and that also on a windows server as my isp blocks plain  
 SIP
 and
 my dedicated server is windows 2003. the problem is i cant get TLS
 running
 on windows using freeswitch and what i was told and what i read  
 was sofia
 isnt compiled using TLS, i have openssl installed as well as  
 certificates
 installed but can any1 plz guide me on how to compile sofia using  
 TLS on
 windows or if some1 could send me a compiled sofia for windows that
 supports
 TLS so i can atleast get it running and then help in testing other
 features
 etc.
 i was also told its a build cycle issues and not a bug so i have  
 reported
 this on jira, if some1 could enable TLS support on the windows build
 process
 or any sort of help is very much appreciated.

 Bipin
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Re: [Freeswitch-users] cant get TLS running at all

2008-08-29 Thread xbipin

is it possible to use the library provided by gnutls project? did u try that
out?



UV-5 wrote:
 
 The compiled version from openssl site doesn't include the lib files (I
 have
 no idea why). You'll have to compile those yourself. Unfortunately, you
 can
 only do that using Perl, so it's quite a cumbersome process - especially
 for
 the newbie programmer.
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of xbipin
 Sent: Friday, August 29, 2008 1:10 AM
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] cant get TLS running at all
 
 
 so do i need to compile sofia or openssl as the openssl u told me, i have
 the
 compiled version directly from the site for windows platform. does it need
 to be re compiled or how is it all as i am a newbie in programming and
 compiling
 
 
 
 UV-5 wrote:
 
 Bipin,
 
 I'm working on an openssl build under VS2008 but it's not easy.
 If you wish to try compiling sofia with TLS, I suggest you download the
 latest openssl build (0.9.8i) and compile it (you'll need Perl to do
 that).
 If done correctly, this should give you the necessary lib files to link
 with
 sofia.
 
 Good luck.
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 xbipin
 Sent: Thursday, August 28, 2008 7:44 PM
 To: freeswitch-users@lists.freeswitch.org
 Subject: [Freeswitch-users] cant get TLS running at all
 
 
 hi,
 
 actually to try out freeswitch to the fullest, i need to get it running
 using TLS and that also on a windows server as my isp blocks plain SIP
 and
 my dedicated server is windows 2003. the problem is i cant get TLS
 running
 on windows using freeswitch and what i was told and what i read was sofia
 isnt compiled using TLS, i have openssl installed as well as certificates
 installed but can any1 plz guide me on how to compile sofia using TLS on
 windows or if some1 could send me a compiled sofia for windows that
 supports
 TLS so i can atleast get it running and then help in testing other
 features
 etc.
 i was also told its a build cycle issues and not a bug so i have reported
 this on jira, if some1 could enable TLS support on the windows build
 process
 or any sort of help is very much appreciated.
 
 Bipin
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[Freeswitch-users] cant get TLS running at all

2008-08-28 Thread xbipin

hi,

actually to try out freeswitch to the fullest, i need to get it running
using TLS and that also on a windows server as my isp blocks plain SIP and
my dedicated server is windows 2003. the problem is i cant get TLS running
on windows using freeswitch and what i was told and what i read was sofia
isnt compiled using TLS, i have openssl installed as well as certificates
installed but can any1 plz guide me on how to compile sofia using TLS on
windows or if some1 could send me a compiled sofia for windows that supports
TLS so i can atleast get it running and then help in testing other features
etc.
i was also told its a build cycle issues and not a bug so i have reported
this on jira, if some1 could enable TLS support on the windows build process
or any sort of help is very much appreciated.

Bipin
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Re: [Freeswitch-users] cant get TLS running at all

2008-08-28 Thread xbipin

so do i need to compile sofia or openssl as the openssl u told me, i have the
compiled version directly from the site for windows platform. does it need
to be re compiled or how is it all as i am a newbie in programming and
compiling



UV-5 wrote:
 
 Bipin,
 
 I'm working on an openssl build under VS2008 but it's not easy.
 If you wish to try compiling sofia with TLS, I suggest you download the
 latest openssl build (0.9.8i) and compile it (you'll need Perl to do
 that).
 If done correctly, this should give you the necessary lib files to link
 with
 sofia.
 
 Good luck.
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of xbipin
 Sent: Thursday, August 28, 2008 7:44 PM
 To: freeswitch-users@lists.freeswitch.org
 Subject: [Freeswitch-users] cant get TLS running at all
 
 
 hi,
 
 actually to try out freeswitch to the fullest, i need to get it running
 using TLS and that also on a windows server as my isp blocks plain SIP and
 my dedicated server is windows 2003. the problem is i cant get TLS running
 on windows using freeswitch and what i was told and what i read was sofia
 isnt compiled using TLS, i have openssl installed as well as certificates
 installed but can any1 plz guide me on how to compile sofia using TLS on
 windows or if some1 could send me a compiled sofia for windows that
 supports
 TLS so i can atleast get it running and then help in testing other
 features
 etc.
 i was also told its a build cycle issues and not a bug so i have reported
 this on jira, if some1 could enable TLS support on the windows build
 process
 or any sort of help is very much appreciated.
 
 Bipin
 -- 
 View this message in context:
 http://www.nabble.com/cant-get-TLS-running-at-all-tp19197345p19197345.html
 Sent from the Freeswitch-users mailing list archive at Nabble.com.
 
 
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Re: [Freeswitch-users] cant get TLS running at all

2008-08-28 Thread xbipin

if ur using the TLS funstion successfully then can u plz give me those files?


UV-5 wrote:
 
 The compiled version from openssl site doesn't include the lib files (I
 have
 no idea why). You'll have to compile those yourself. Unfortunately, you
 can
 only do that using Perl, so it's quite a cumbersome process - especially
 for
 the newbie programmer.
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of xbipin
 Sent: Friday, August 29, 2008 1:10 AM
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] cant get TLS running at all
 
 
 so do i need to compile sofia or openssl as the openssl u told me, i have
 the
 compiled version directly from the site for windows platform. does it need
 to be re compiled or how is it all as i am a newbie in programming and
 compiling
 
 
 
 UV-5 wrote:
 
 Bipin,
 
 I'm working on an openssl build under VS2008 but it's not easy.
 If you wish to try compiling sofia with TLS, I suggest you download the
 latest openssl build (0.9.8i) and compile it (you'll need Perl to do
 that).
 If done correctly, this should give you the necessary lib files to link
 with
 sofia.
 
 Good luck.
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 xbipin
 Sent: Thursday, August 28, 2008 7:44 PM
 To: freeswitch-users@lists.freeswitch.org
 Subject: [Freeswitch-users] cant get TLS running at all
 
 
 hi,
 
 actually to try out freeswitch to the fullest, i need to get it running
 using TLS and that also on a windows server as my isp blocks plain SIP
 and
 my dedicated server is windows 2003. the problem is i cant get TLS
 running
 on windows using freeswitch and what i was told and what i read was sofia
 isnt compiled using TLS, i have openssl installed as well as certificates
 installed but can any1 plz guide me on how to compile sofia using TLS on
 windows or if some1 could send me a compiled sofia for windows that
 supports
 TLS so i can atleast get it running and then help in testing other
 features
 etc.
 i was also told its a build cycle issues and not a bug so i have reported
 this on jira, if some1 could enable TLS support on the windows build
 process
 or any sort of help is very much appreciated.
 
 Bipin
 -- 
 View this message in context:
 http://www.nabble.com/cant-get-TLS-running-at-all-tp19197345p19197345.html
 Sent from the Freeswitch-users mailing list archive at Nabble.com.
 
 
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Re: [Freeswitch-users] latest freeswitch for windows doesnt run using TLS

2008-08-23 Thread xbipin

thanks a lot for all the help, i was thinking bugs were supposed to be
reported to this mailing list and that jira.freeswitch, im totally new to it
so cant figure out how to use it.
The last time i downloaded the latest snapshot and compiled it on windows, i
had openssl already installed on my system and the certificates already
generate but mayb im doing something wrong, it would be great if some1 could
post a small wiki on it on how to compile using openssl or if some1 could
just compile the msi with the openssl support.

Regards,
Bipin



Anthony Minessale-2 wrote:
 
 Also,
 
 Now that Brian has politely helped you let me give you some advice.
 
 Next time, leave out the rude part about how you have reported this many
 times and nobody fixed it for you.
 Let me be the first to offer you our triple-your-money-back guarantee!
 
 Clearly you did not report your issue to jira http://jira.freeswitch.org
 or
 we would know about it.
 
 
 
 On Wed, Aug 20, 2008 at 9:16 AM, Brian West [EMAIL PROTECTED] wrote:
 

 On Aug 20, 2008, at 7:44 AM, xbipin wrote:

 
  hi,
 
  i failed to run the latest freeswitch msi package using TLS on port
  5061 on
  windows xp, i have reported this multiple times but havent got any
  positive
  feedback from others as well as no solution to the bug.

 Have you reported this to jira.freeswitch.org?  This is the first I
 have heard of this.


 
  the bug is, under windows xp when u install and the run freeswitch,
  it works
  fine in standard sip mode but as soon as u turn the tls function to
  true and
  then re run it, sofia gives a thread error i think error number is
  552 or
  something like that, basically what i understand from other posts is
  that it
  cant listen on port 5061 but i got nothing running on that port at
  all so
  then whats causing it not to run?

 TLS isn't enabled by default on ANY built for one single reason.  You
 have to make your TLS certificates.  And on windows I would bet that
 Sofia isn't compiled with OpenSSL support and thats the reason its not
 working.


 
  this has been the issue since the first version of freeswitch i have
  tried
  to the latest one present, i even compiled the source on windows and
  then
  tried but its the same so just hope that some1 could get a solution
  for it.
  What others have reported that it run fine on linux using tls but
  doesnt on
  windows using TLS.

 Open a bug on Jira under build system issues.  Its not a bug with the
 code but with the build system on windows.

 
 
  Regards,
  Bipin

 Brian West
 sip:[EMAIL PROTECTED] [EMAIL PROTECTED]




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 FreeSWITCH http://www.freeswitch.org/
 ClueCon http://www.cluecon.com/
 
 AIM: anthm
 MSN:[EMAIL PROTECTED] [EMAIL PROTECTED]
 GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED]
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 sip:[EMAIL PROTECTED] [EMAIL PROTECTED]
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 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED]
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Re: [Freeswitch-users] latest freeswitch for windows doesnt run using TLS

2008-08-20 Thread xbipin

hi,

i failed to run the latest freeswitch msi package using TLS on port 5061 on
windows xp, i have reported this multiple times but havent got any positive
feedback from others as well as no solution to the bug.
the bug is, under windows xp when u install and the run freeswitch, it works
fine in standard sip mode but as soon as u turn the tls function to true and
then re run it, sofia gives a thread error i think error number is 552 or
something like that, basically what i understand from other posts is that it
cant listen on port 5061 but i got nothing running on that port at all so
then whats causing it not to run?
this has been the issue since the first version of freeswitch i have tried
to the latest one present, i even compiled the source on windows and then
tried but its the same so just hope that some1 could get a solution for it.
What others have reported that it run fine on linux using tls but doesnt on
windows using TLS.

Regards,
Bipin



xbipin wrote:
 
 hi,
 it works under linux but not under windows, when u enable tls in
 internal.xml, then sofia gives a error, some thread error in windows, i
 have the certificates generated and put in the ssl folder
 
 
 
 Brian West-3 wrote:
 
 I just did from a fresh install and it works fine.
 
 ./gentls_cert setup
 
 ./gentls_cert create -cn 192.168.0.163 -alt DNS:192.168.0.163
 
 Flip the false to true.. restart and it came right up.
 
 /b
 
 On Aug 3, 2008, at 2:56 PM, xbipin wrote:
 

 i did as explained but i got the same problem the other guy got in  
 ubuntu,
 error loading internal profile, any solution to it



 \UV-5\ wrote:

 As far as I know, there is no ability to generate certificates for  
 TLS on
 any windows release.
 Since I also needed that, I shared my experience in this temporary  
 wiki
 page
 until an automated script is written for windows (brutally  
 mimicking the
 gentls_cert script):
 http://wiki.freeswitch.org/wiki/Generating_TLS_certificates_%28win32%29

 Hope it helps.

 UV

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of  
 xbipin
 Sent: Thursday, July 31, 2008 8:00 PM
 To: freeswitch-users@lists.freeswitch.org
 Subject: [Freeswitch-users] latest freeswitch for windows


 can any1 plz compile the latest freeswitch for windows with the  
 ability to
 generate certificates for TLS protocol as i am just unable to do it  
 with
 my
 little knowledge for C  and visual studio.
 basically i looking for he latest windows freeswitch with TLS, as in
 plugins
 etc included for generating SSL certifiates.
 if any1 can do it then i can host it and provide very good  
 bandwidth for
 others to download.
 -- 
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Re: [Freeswitch-users] latest freeswitch for windows

2008-08-03 Thread xbipin

i did as explained but i got the same problem the other guy got in ubuntu,
error loading internal profile, any solution to it



\UV-5\ wrote:
 
 As far as I know, there is no ability to generate certificates for TLS on
 any windows release.
 Since I also needed that, I shared my experience in this temporary wiki
 page
 until an automated script is written for windows (brutally mimicking the
 gentls_cert script):
 http://wiki.freeswitch.org/wiki/Generating_TLS_certificates_%28win32%29
 
 Hope it helps.
 
 UV
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of xbipin
 Sent: Thursday, July 31, 2008 8:00 PM
 To: freeswitch-users@lists.freeswitch.org
 Subject: [Freeswitch-users] latest freeswitch for windows
 
 
 can any1 plz compile the latest freeswitch for windows with the ability to
 generate certificates for TLS protocol as i am just unable to do it with
 my
 little knowledge for C  and visual studio.
 basically i looking for he latest windows freeswitch with TLS, as in
 plugins
 etc included for generating SSL certifiates.
 if any1 can do it then i can host it and provide very good bandwidth for
 others to download.
 -- 
 View this message in context:
 http://www.nabble.com/latest-freeswitch-for-windows-tp18751293p18751293.html
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[Freeswitch-users] latest freeswitch for windows

2008-07-31 Thread xbipin

can any1 plz compile the latest freeswitch for windows with the ability to
generate certificates for TLS protocol as i am just unable to do it with my
little knowledge for C  and visual studio.
basically i looking for he latest windows freeswitch with TLS, as in plugins
etc included for generating SSL certifiates.
if any1 can do it then i can host it and provide very good bandwidth for
others to download.
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[Freeswitch-users] Windows compiled binary doesnt generate certificate for tls

2008-07-28 Thread xbipin

i have been looking for the freeswitch windows binary from long and i found
that at freeswitch so i downloaded it etc and works perfect except i wanna
test it by installing it on my server but voip is blocked in my country so i
have to use tls for signalling but when i enable that in internal.xml then i
cant find a way to generate a certificate for it so the ssl folder remains
empty. can any1 tell me is there a way to copy a certificate file to that
folder as i have 2 certificate generated from voxilla for tls and srtp but
both just have the mini cert portion and key file, so how do i put this into
a file and use it from freeswitch as the gen_cert command doesnt work to
generate certificate files as i think the script is missing for windows or
something like that, no idea as im new to this. basically some1 help me just
get the device to register using tls, the srtp portion ill look into later
on. i did a trace and found out that the tls packets from linksys spa3102
reach the server but nothing happens as certs missing in freeswitch
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