c...@apartmentlines.com (Chad Phillips -- Apartment Lines) writes:
> On Oct 15, 2009, at 11:03 AM, Simon J Mudd wrote:
>
> > OK. That looks clear. However it doesn't look as if it's possible to
> > pass parameters this way which means that data can only be passed to
> > the "extension" via variab
Thanks, our wiki is getting a lot better :).
Diego
On Sat, Oct 17, 2009 at 5:54 AM, Michael Collins wrote:
> Diego, when I get some time I will help convert these from IRC to wiki.
> Thanks,
> MC
>
>
> On Fri, Oct 16, 2009 at 8:50 PM, Diego Viola wrote:
>
>> This needs to be converted to docume
On Fri, Oct 16, 2009 at 9:18 AM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:
> 1) you should update again there were a few issues.
> 2) you can set the variable sip_callee_id_name and sip_callee_id number on
> the inbound leg before you answer to control what it says.
>
>
> Thanks for
Thanks for the explanation. I added this to the wiki.
-MC
On Thu, Oct 15, 2009 at 6:50 PM, Michael Jerris wrote:
> It updates the display on a phone if the phone supports this. This works
> on some sip phones right now including polycom and snom.
> Mike
>
> On Oct 12, 2009, at 2:11 AM, Matthew
Diego, when I get some time I will help convert these from IRC to wiki.
Thanks,
MC
On Fri, Oct 16, 2009 at 8:50 PM, Diego Viola wrote:
> This needs to be converted to documentation form too.
>
> http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations
>
> Diego
>
>
_
That's exactly what I needed. Many thanks.
On Fri, Oct 16, 2009 at 8:11 PM, Diego Viola wrote:
> Hi Michael,
>
> You might want to look at the mod_dptools wiki.
>
> http://wiki.freeswitch.org/wiki/Mod_dptools
>
> mod_dptools is the module that contains the dialplan apps.
>
> Regards,
>
> Diego
>
Just a heads up, I have talked to Jeremias from Khomp today and he is
setting up the wiki. I will personally be adding contents to the that wiki
if it ever picks up.
Regards,
jm
On Sat, Oct 17, 2009 at 12:52 AM, Jason White wrote:
> Diego Viola wrote:
> > I'm with Moises and with the other pe
Diego Viola wrote:
> I'm with Moises and with the other people supporting this initiative.
>
> I'm not Brazilian, but they should be able to do whatever they want, after
> all, that's how open source works, if you can do it go ahead and do it.
Correct. We have enough of them as far as English-la
This needs to be converted to documentation form too.
http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations
Diego
On Sat, Oct 17, 2009 at 1:54 AM, Diego Viola wrote:
> s/so all pages look the same/so all the pages have the same look and feel/
>
>
> On Sat, Oct 17, 2009 at 1:47 AM, Dieg
I'm with Moises and with the other people supporting this initiative.
I'm not Brazilian, but they should be able to do whatever they want, after
all, that's how open source works, if you can do it go ahead and do it.
Diego
On Sat, Oct 17, 2009 at 3:05 AM, Moises Silva wrote:
> This mailing list
Hi,
We've been using 13168M in production for some time now (works great).
I want to get us onto the latest build but am having problems getting
NAT to work.
Phones can register; can dial test #, but after 100 seconds the
call is disconnected with error:
2009-10-16 19:52:26.936618 [NOTICE]
>
> This mailing list, the development mailing list, the IRC channel and the
> FreeSWITCH conference all exist - no need for another "forum".
>
>
I think he meant a forum in Portuguese. May be it's hard to believe for some
people, but not everybody speaks English. Programmers tend to do it by
defau
Itamar Reis Peixoto wrote:
> instead of creating thousand of small sites, why don't put the content
> in only place.
>
> freeswitch.org already exist, no need for a new domain.
This mailing list, the development mailing list, the IRC channel and the
FreeSWITCH conference all exist - no need for
Rudá Cunha wrote:
> Hello,
>
> I wanted to create a forum so we can discuss FreeSWITCH.
You're writing to one now. It can be accessed as a mailing list, on the Web at
gmane.org, and via NNTP at gmane.org. I think there's a Web archive somewhere
else, too.
__
s/so all pages look the same/so all the pages have the same look and feel/
On Sat, Oct 17, 2009 at 1:47 AM, Diego Viola wrote:
> My plan to improve the FreeSWITCH wiki
>
>
>1. Correct all typos from all pages.
>2. Delete unused/irrelevant pages.
>3. Make sure we don't repeat stuff (m
My plan to improve the FreeSWITCH wiki
1. Correct all typos from all pages.
2. Delete unused/irrelevant pages.
3. Make sure we don't repeat stuff (multi-tenant/multi-companies,
features/spec-sheet, etc)
4. Define a size for text, titles/subtitles (so all pages look the same).
5.
Hello,
I wanted to create a forum so we can discuss FreeSWITCH.
That person who registered the domain, I can create and host the forum, so
we can talk there and ask questions
2009/10/16 Itamar Reis Peixoto
> instead of creating thousand of small sites, why don't put the content
> in only plac
Hi Michael,
You might want to look at the mod_dptools wiki.
http://wiki.freeswitch.org/wiki/Mod_dptools
mod_dptools is the module that contains the dialplan apps.
Regards,
Diego
On Sat, Oct 17, 2009 at 12:29 AM, Michael Gende wrote:
> Hey Diego,
>
> You seem to know your way around the docum
Hey Diego,
You seem to know your way around the documentation and clearly have many
contributions. Thus, a question:
Can you point me to doc on dial plan applications (such as transfer, bridge,
etc)? It would help me greatly. Sorry if this is in an obvious place and
I've just not seen it despite
ct to each other, while the PBX part adds
services like voice mail, etc.?
Thank you.
__ Information from ESET NOD32 Antivirus, version of virus signature
database 4516 (20091016) __
The message was checked by ESET NOD32 Antivirus.
htt
Can someone please convert this IRC log to documentation form:
http://wiki.freeswitch.org/wiki/Cacti
Try to explain what the IRC log explains in documentation form, and get rid
of the IRC log.
I'm working on fixing typos and stuff like that on the wiki now, then I will
focus on fixing the format
It already should support at least recv of rfc2833 any time.
as a workaround in googletalk any string that starts with + typed in the
chat box is treated as dtmf by FS
e.g. +1
Once the jingle spec stops being a moving target we will re-investigate
making sure its supported properly.
On Thu, Oct
I'm sure it won't work because we only implemented forwarding t38 with
invite
it would require adding code to say the least.
On Fri, Oct 16, 2009 at 9:35 AM, Peter P GMX wrote:
> Hello Michael
>
> this is a productive system, so I can currently not update to the
> current trunk.
> But the insta
instead of creating thousand of small sites, why don't put the content
in only place.
freeswitch.org already exist, no need for a new domain.
On Fri, Oct 16, 2009 at 5:02 PM, Fabio Ferrari wrote:
> Just to let you know, Jeremias Silva from Khomp registered the
> freeswitchbrasil.org domain. I
> Of course, I was listening to my A.M radio the other day and they said that
> there was this new invention called the Internet that would let people send
> documents to each other electronically. Maybe you should look into that.
> Next thing you know they'll come up with telephones that people do
Just to let you know, Jeremias Silva from Khomp registered the
freeswitchbrasil.org domain. I think there is space to build an wiki
with portuguese content there.
Fabio Ferrari
On Fri, Oct 16, 2009 at 14:20, Roberto Martins wrote:
> we do!.
>
> roberto
> On Oct 16, 2009, at 11:25 AM, João Mesqui
On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang wrote:
> So how would you trigger it from a script dialplan? The only time it seemed
> to work is when I did setVariable("api_after_bridge", "sched_api blah blah
> blah");
> but then it gets executed after the channel's been teared down. I thought
> ap
So how would you trigger it from a script dialplan? The only time it seemed
to work is when I did setVariable("api_after_bridge", "sched_api blah blah
blah");
but then it gets executed after the channel's been teared down. I thought
api_after_bridge means right after the call gets connected.
I nee
You can signup for a wiki account and correct the documentation so
others don't repeat the same issue.
Thanks,
Brian
On Oct 16, 2009, at 12:16 PM, Jerry Richards wrote:
>
> Okay, I think the contrib folder moved up one level. So the Wiki
> installation documentation should probably be updated
Lars Zeb said:
> I tried dial sip:e...@iptel.org. In the pastebin you can see that this
> address came over to the cli as "echo". Is it Bria which is messing up, or
> my FS configuration?
It looks like a Bria problem. If you open the account information and go
to the Advanced tab, is the box che
we do!.
roberto
On Oct 16, 2009, at 11:25 AM, João Mesquita wrote:
> We do. :-)
>
> João Mesquita
>
> On Fri, Oct 16, 2009 at 11:02 AM, Itamar Reis Peixoto
> > wrote:
> I think no.
>
>
> On Fri, Oct 16, 2009 at 10:48 AM, Pedro Prado
> wrote:
> > Hi,
> >
> > Do you have a group of Brazilians
Okay, I think the contrib folder moved up one level. So the Wiki
installation documentation should probably be updated to reflect that.
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Friday, October 16, 2009 9:47 AM
To: 'freeswitc
I am building Freeswitch on a Centos 5.3 machine and the last step below
gets an error because there is no scripts/contrib folder. Anyone know why?
./configure
make
make all install sounds-install uhd-moh-install moh-install
scripts/contrib/trixter/makemodconf.pl modules.conf >
/usr/local/freeswi
1) you should update again there were a few issues.
2) you can set the variable sip_callee_id_name and sip_callee_id number on
the inbound leg before you answer to control what it says.
On Fri, Oct 16, 2009 at 6:31 AM, Helmut Kuper wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hel
The weekly conference call is about to start. Please call in. The agenda and
calling instructions are here:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_16
Looking forward to hearing from everyone!
-Michael
___
FreeSWITCH-users mailing list
FreeSWI
Yes, it is what I need. But now I have problem with sending dtmf. Here what
I've done:
[r...@centos4-4-vm ~]# telnet localhost 8021
Trying 127.0.0.1...
Connected to localhost.localdomain (127.0.0.1).
Escape character is '^]'.
Content-Type: auth/request
auth ClueCon
Content-Type: command/reply
Rep
If you setup your own stun server it wouldn't do that But the
hostlookup only solves half the problem .. getting the external IP vs
poking holes for RTP which is what stun will do.
/b
On Oct 15, 2009, at 10:35 PM, Mark Campbell-Smith wrote:
> Thanks Brian. Is this something that is pla
Hello Michael
this is a productive system, so I can currently not update to the
current trunk.
But the installed SVN is 14741. Were those changes after 14741? Then
I'll try to find a timeslot at night in order to update freeswitch.
But Freeswitch should forward the UPDATE request in proxy-media m
We do. :-)
João Mesquita
On Fri, Oct 16, 2009 at 11:02 AM, Itamar Reis Peixoto <
ita...@ispbrasil.com.br> wrote:
> I think no.
>
>
> On Fri, Oct 16, 2009 at 10:48 AM, Pedro Prado wrote:
> > Hi,
> >
> > Do you have a group of Brazilians here?
> >
> > Thanks,
> > Pedro Prado
>
>
>
> --
>
I didn't know that. Thanks Russell.
>
>
>
> You've mentioned this twice, now, but a better way to do it is to _not_
> stop FreeSwitch. Instead, run /usr/local/freeswitch/bin/fs_cli, which
> connects to FS, and then you can do everything that you can do directly in
> FS. When you are done, enter "
I think no.
On Fri, Oct 16, 2009 at 10:48 AM, Pedro Prado wrote:
> Hi,
>
> Do you have a group of Brazilians here?
>
> Thanks,
> Pedro Prado
--
Itamar Reis Peixoto
e-mail/msn: ita...@ispbrasil.com.br
sip: ita...@ispbrasil.com.br
skype: itamarjp
icq: 81053601
+55 11 4063 5033
+
Hi,
Do you have a group of Brazilians here?
Thanks,
Pedro Prado
_
Você sabia que com o Hotmail você tem espaço ilimitado para guardar seus
e-mails? Começe a usar já!
http://www.microsoft.c
Try turning up all the sofia debug to 9.
Mike
On Oct 14, 2009, at 2:16 AM, Szasz Szabolcs wrote:
> Hi,
>
> Did anybody set up TLS between Freeswitch and Audiodes MP11X ? I got
> to work TLS between freeswitch and a softphone (phonerlite), but I
> have problem with Audiocodes during the TLS
There was just a bunch of work on UPDATE, can you confirm this is the
same behavior with trunk?
On Oct 14, 2009, at 6:55 AM, Peter P GMX wrote:
> Hello,
>
> we have the following problem.
> 2 Fax machines are communicating via Freeswitch. One is externally
> attached via a Telco who is able to
I would love to see this work in tree, but i am pretty sure it has
never worked. I would gladly accept patches that implement this.
Mike
On Oct 14, 2009, at 2:33 AM, Simon J Mudd wrote:
> br...@freeswitch.org (Brian West) writes:
>
>> You shouldn't have to make clean usually ... doing so mig
sched_api is a fsapi command not a dialplan application, I believe
sched_hangup is both.
Mike
On Oct 13, 2009, at 6:14 AM, Henry Huang wrote:
Hi:
I am using mod_java. And in my script I was able to achieve using:
execute("sched_hangup", "+300 alloted_timeout");
However, when I try to run
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
after updating FS to trunk a few days ago I found that callee's display
is updated serveral times to caller's name after callee picked up. The
first two equal INFO messages looks like this:
INFO sip:2...@85.16.245.213:1040;line=367hfn9i SIP/2.
Russell Mosemann wrote:
>
> Yup, I got the slashes wrong, and I was staring right at the commands when I
> wrote the message. I'll blame it on how late it is here. :-) It should be
> "/quit", "/bye", "/exit" and "/help".
ctrl-d is also recognized and it's easier to type.
__
> When you are done, enter "\quit" or "\bye" or "\exit". See "\help" and
> "help" for more information.
Yup, I got the slashes wrong, and I was staring right at the commands when I
wrote the message. I'll blame it on how late it is here. :-) It should be
"/quit", "/bye", "/exit" and "/help".
--
Michael Gende wrote:
> Hit option 8 and you'll go to a shell. I'm assuming you can get around in
> Linux or Unix (If not, tell me). "ps aux | grep free" and kill the PID for
> FS (pfsense runs FS in the background) Then, invoke FS with
> /usr/local/freeswitch/bin/freeswitch.
You've mentioned this
Hello
I was reading an article the other day about Freeswitch in which the author
used the terms "SIP proxy" and PBX. Is an SIP proxy just a way for SIP
clients to register and connect to each other, while the PBX part adds
services like voice mail, etc.?
Thank you.
--
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