Thanks Anthony,
however this rather deteriorated the situation.
Now it works the following
- A calls B
- B enters *4 gets an announcement and enters digits for C (A get MOH)
- C is called
- As soon as C picks up the call, A and C both have no voice (and B is
dropped)
- When A hangs up, C hangs up
All of the example I see allow me to call FROM gtalk.
Help?
Thanks,
David
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Thank You for such an elegant and simple solution that I have not
thought about.
With an exception that I'm using FS 1.0.4 right now and it appears that
something changed in time and following line should use hash instead of
db (when using default 1.0.4 FS config):
action application="set"
Mitch Capper wrote:
I did something like this recently.
Thanks for the feedback. I'll see how Linux can be made to send stuff to a
USB display.
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mercutioviz wrote:
I believe that French and Spanish sounds are in the works by the
community.
The only other sounds I'm aware of are the Russian ones.
Thanks for the tip.
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Sent from the
Thanks Mark
I read this and didn't find a dialplan (do I need one?) to make calls into
gtalk? I mean how would I even dial in? via URI (e.g. some...@gmail.com)?
Wouldn't that just send the call to gmail?
What I am looking for is hard coding a number (e.g. 1010) that would enable me
to call it
Hello,
Obviously it is possible, next time try to search better, the answer is on
the same blog Mark pointed you too:
http://chesterton.id.au/blog/2008/01/02/freeswitch-google-talk-dingaling-jingle-all-the-way/
Thanks for the pointers - I'll head off there now..
regards
Dave
- Original Message -
From: Fred-145 codecompl...@free.fr
To: freeswitch-users@lists.freeswitch.org
Sent: Thursday, November 12, 2009 12:59 PM
Subject: Re: [Freeswitch-users] SPA3102 Won't drop the PSTN line (UK)
Thanks
I overlooked that :)
D.
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Milena
Sent: Thursday, November 12, 2009 3:36 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re:
This is just basic freeswitch dialplan concepts. It has nothing to do
specifically with gtalk. Seems like you need to step back and do some
more reading on the dialplan. ;)
/b
On Nov 12, 2009, at 7:02 AM, David Schwartz wrote:
What I am looking for is hard coding a number (e.g. 1010)
Has anybody every figured out how to get presence working on a Cisco 79x1 w/
FreeSWITCH? I spent quite a bit of time 6+ months ago on it and could never
get it to work.
Peder
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Hi, I'm running a ivr script on FS, the call is from a softswitch to extenal
sip endpoint of FS.
I added two dialplan in public dialplan xml file. as flow:
extension name=ivr_demo2
condition field=destination_number expression=^8$
action application=lua data=../ivr/test.lua/
Russell.Mosemann wrote:
Yes, it should just work. I'd recommend Dahdi (complete), because Zaptel
is not being developed anymore.
Thanks for the links. Turns out this card seems incompatible with the
motherboard I have, so I'll concentrate on the Linksys 3102 instead.
--
View this message in
if you provide a console trace of both situations with console loglevel
debug and put them on pastebin i can tell you what's happening.
On Thu, Nov 12, 2009 at 2:38 AM, Peter P GMX prometheus...@gmx.net wrote:
Thanks Anthony,
however this rather deteriorated the situation.
Now it works the
Take a look at the freeswitch debug log, it should tell you exactly why it hung
up.
Mike
On Nov 12, 2009, at 10:01 AM, Lei Tang wrote:
Hi, I'm running a ivr script on FS, the call is from a softswitch to extenal
sip endpoint of FS.
I added two dialplan in public dialplan xml file. as flow:
Has anyone used a Polycom SoundPoint IP501 or similar hard phone with
FreeSWITCH? I configured one to register with my FreeSWITCH server using one
of the default sip profiles to test and I get [DEBUG] sofia_reg.c:1688 SIP
username 1001 does not match auth username in the log file and the phone
They do it in their own weird way... if you wanna track it down I know
their are examples of it out there.
/b
On Nov 12, 2009, at 8:25 AM, Peder wrote:
Has anybody every figured out how to get presence working on a Cisco
79x1 w/
FreeSWITCH? I spent quite a bit of time 6+ months ago on
Not sure what do you have in your config file for the polycom
exactly? btw you hijacked the Cisco Presence thread by clicking
reply.. and changing the subject please don't do that in the future.
Click new message and input the address for the list.
Thanks,
Brian
On Nov 12, 2009, at 11:41
I'm trying to increase the number of calls per second that I can originate
from FreeSWITCH, but I cannot seem to get more than two-per-second.
(I am trying to use FS to initiate thousands of calls quickly)
switch.conf.xml
I beefed up the max-sessions and sessions-per-second in the
Tina,
How are you originating the calls? from the console? Try bgapi originate...
--matt
Voice Broadcasting - http://www.hellohunter.com/voice_blast.php
On Fri, Nov 13, 2009 at 12:57 AM, t...@a2unlimited.com wrote:
I'm trying to increase the number of calls per second that I can originate
Matt,
Thank you so much!
bgapi did the trick.
- Tina
Tina,
How are you originating the calls? from the console? Try bgapi
originate...
--matt
Voice Broadcasting - http://www.hellohunter.com/voice_blast.php
On Fri, Nov 13, 2009 at 12:57 AM, t...@a2unlimited.com wrote:
I'm trying to
Hi all,
I'm currently building a proof-of-concept box using Freeswitch. Coming
from Asterisk/Kamalio/OpenSER it looks very cool so far, very complete.
The plan is to make some sort of SIP router, some would call it an SBC I
guess. There will be no PBX stuff, just gateways that talk to each
I am using Polycoms (430 and 501) with FreeSwitch. How do you provision
them? Via WEB or config files?
If you use config files than I can send you some sample files.
Regards, __Yehavi:
On Nov 12, 2009, at 11:41 AM, Adam Ford wrote:
Has anyone used a Polycom
Take a look at mod_easyroute.
On Thu, Nov 12, 2009 at 1:14 PM, Robin Vleij vi...@fx-services.com wrote:
Hi all,
I'm currently building a proof-of-concept box using Freeswitch. Coming
from Asterisk/Kamalio/OpenSER it looks very cool so far, very complete.
The plan is to make some sort of SIP
I am using xml_cdr to generate CDR results from FreeSWITCH servers, and
I've noticed that failed call attempts are not showing up in the results.
Whereas the failed attempt is showing up in the Master.csv file.
For example, I've initiated some outbound calls that show up in the
Master.csv as
Hello everyone!
I'v got strange problem with incomplete call via tcp transport. When I
perform bridged call from one ua (no matter what transport udp or tcp)
through FS this call's leg b message sequence (over tcp) lacks finishing SIP
message what in it's turn cause the call to be disconnected by
HI all,
i have connected Freeswtich(mod event socket) through telnet(tcp) 8021 port,
when i am trying to connect freeswtich it it taking 20 seconds to get
response from FS,
can i able to reduce tcp response time?
thanks
Srinivasula Reddy K
___
On 11/12/09 9:59 PM, Rupa Schomaker wrote:
Hi!
Take a look at mod_easyroute.
Cool, I remember quick-reading about that module and thinking nah,
not needed. Then when the plan changed and I needed the large amount of
routes it didn't struck me that easyroute is what I need for what I want
to
I was trying to configure it just on the phone itself, but apparently even
though it says Auth. User on the phone setting, it doesn't actually set the
auth username according to the web interface. After using the web interface
to configure the phone it works now.
Thank you for your
tack on a ;transport=tcp
/b
On Nov 12, 2009, at 4:27 PM, RobertT wrote:
action application=bridge data=sofia/external_call/
$1%${domain_name}/
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On Thu, Nov 12, 2009 at 4:32 PM, Robin Vleij vi...@fx-services.com wrote:
On 11/12/09 9:59 PM, Rupa Schomaker wrote:
If I read it right, this is suited for complete nrs. So would I have a
system connected with lots of DIDs, I would put them in easyroute. Then
for systems with lots of number
enable the b leg logging
On Thu, Nov 12, 2009 at 3:19 PM, t...@a2unlimited.com wrote:
I am using xml_cdr to generate CDR results from FreeSWITCH servers, and
I've noticed that failed call attempts are not showing up in the results.
Whereas the failed attempt is showing up in the Master.csv
What exactly are you typing when you connect? Also, which version of FS?
-MC
On Thu, Nov 12, 2009 at 2:32 PM, srinivasula reddy
srinivas.ksvre...@gmail.com wrote:
HI all,
i have connected Freeswtich(mod event socket) through telnet(tcp) 8021
port, when i am trying to connect freeswtich
Or, of course, there is always mod_xml_curl. Basically, XML dialplan
on the fly. Call comes in, FreeSWITCH sends XML request via HTTP to a
web application server, web application server responds with XML
routing response, FreeSWITCH routes the call.
On Thu, Nov 12, 2009 at 5:53 PM, Rupa
On Thu, Nov 12, Orien Love wrote:
Since I have not had any replies about the atom board I am guessing
that nobody has used one, Could somebody tell me what is a good CPU
speed / Memory / FSB be?
I really do not have a large budget and cannot afford to buy
something that will not
Hi Jason
Thanks for your response, I setup the configuration with 2 proxies based on
the example of the freeswitch wiki.
I looked at freeswitch.log and found the following line.
Dialplan: sofia/internal/1...@74.207.249.79 Action
set(effective_caller_id_number=1222333)
Dialplan:
but FS does use tcp for that call leg - RX 1167 bytes ... from *tcp* ...:
And after all there can be other SIP transports combinations FS should
interconnect...
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