Hi guys,
It's me again, does anyone knows why this doesn't work?
require 'rubygems'
require 'eventmachine'
require 'ESL'
EventMachine.run {
con = EventMachine::start_server 127.0.0.1, 8084 do
fd = con.to_i
esl = ESL::ESLconnection.new(fd)
Hi Jay,
Did you make a wireshark trace yet? You should be able to find out
exactly what's going on there, which protocol is used, etc. We've had
our share of problems with DTMF over SIP trunks as well. Your problems
could also be related to timing issues introduced by multiple gateways.
Do you
Thanks Seven I'll try it very soon.
_
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of seven
Sent: Thursday, May 07, 2009 5:42 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Inboud
I loaded mod_xml_rpc on trunk version 13174, to get voicemail, I go to
http://192.168.1.27:8080/api/voicemail/web
in the challenge box, no matter I input 1009, 1...@192.168.1.27,
1...@localhost, 1...@default, I got Error 403.
However I use freeswitch/works can get the voicemail interface. but
Remko Kloosterman r.klooster...@mtel.nl wrote:
Did you make a wireshark trace yet? You should be able to find out
exactly what's going on there, which protocol is used, etc. We've had
our share of problems with DTMF over SIP trunks as well.
I've just discovered that I'm having a similar
Roger on the previous post asking the question some time ago.
I don't have time to get into this sort of thing, I've got too much on
my plate right now as it is.
Hopefully someone will take up the challenge.
I've cross-posted this to the freeswitch forum.
73 de n2vqm
Best Regards,
Karl
EventMachine is very different to TCPSocket and is definitely not a
drop-in replacement. Take a look at FreeSWITCHeR
(http://code.rubyists.com/projects/fs/repository) and see how they
implemented EventMachine.
More info about EventMachine and specifically #start_server is here:
param name=http-allowed-api value=voicemail/
in the params inside user or domain
On Thu, May 7, 2009 at 5:12 AM, seven dujinf...@gmail.com wrote:
I loaded mod_xml_rpc on trunk version 13174, to get voicemail, I go to
http://192.168.1.27:8080/api/voicemail/web
in the challenge box, no
you may have a sonus infection
try some of the stuff from here under DTMF
http://wiki.freeswitch.org/wiki/RTP_Issues
On Thu, May 7, 2009 at 5:16 AM, Jason White ja...@jasonjgw.net wrote:
Remko Kloosterman r.klooster...@mtel.nl wrote:
Did you make a wireshark trace yet? You should be
To answer the question, channel drivers in FreeSWITCH are called Endpoint
Modules and there is certainly a way
to write them we have nearly a dozen in tree.
On Thu, May 7, 2009 at 6:44 AM, Karl Vesterling k...@ken-ton.com wrote:
Roger on the previous post asking the question some time ago.
Hello
I browsed the wiki + archives of this list, but didn't find an article on
how to add a TDM card to Freeswitch. If I've overlooked it, thank you for
pointing me to it.
I have a one-FXO module OpenVox PCI card. It's correctly detected by Linux
CentOS 5:
# lspci -v
03:00.0
Hello,
I habe the following problem when re-registering to an external SIP
provider during a call which results in immediate call-hangups.
- FS re-registers with nonce
- 2ms later FS re-registers without nonce
- external SIP provider asks for credentials
- FS re-registers with nonce
- External
Looks like they are sending you a bye... not quite sure this is our
problem. Can you provide a sip trace in pcap format for me to see?
/b
On May 7, 2009, at 8:37 AM, Peter P GMX wrote:
Hello,
I habe the following problem when re-registering to an external SIP
provider during a call which
Hi all,
Please if anybody can help me with a problem I'm having. I have 2
gateways Quintum AX and OneAccess 100D both are setup to register with
Freeswitch 1.0.4pre7, when I first switch on the gateways all is well
and then after about 5 min sip register starts falling with sip 401. I
can send
Sounds like the device doesn't challenge correctly. A sip trace would
be nice to see.
/b
On May 7, 2009, at 8:50 AM, Nameer Kazzaz wrote:
Hi all,
Please if anybody can help me with a problem I'm having. I have 2
gateways Quintum AX and OneAccess 100D both are setup to register with
Were they on hold for a bit?
/b
On May 7, 2009, at 10:04 AM, Helmut Kuper wrote:
I scanned my recent FS log files for that message and found that this
error accours a few times a week. I use Snom Phones all with G722 and
SRTP. Any ideas what this could be caused by?
Brian West
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
today a colleague of mine told me that sometimes calls were disconnected
without any obvious reasons. In FS's log I found this:
2009-05-07 15:52:22 [ERR] switch_rtp.c:1656 rtp_common_read() Error:
SRTP unprotect failed with code 7 (auth
Hello Helmut,
I also have problems with my Snom300s and Snom320s and G711 and SRTP.
They may be related to this problem, but I am not sure.
The phones disconnect the media stream after a while (2..10 minutes)
because the Snom media port is blocked all of a sudden.
I have opened a bug report at
Yes, I've seen this http://wiki.freeswitch.org/wiki/SIP_TLS.
I was just curious if the only way to have true end to end secure
communications with FS would have to be a SIP trunk from one FS system to
another encrypted SIP system on the other with no POTS/PRI/BRI circuits used in
transit. I'm
Well its not so easy to take a lineman's handset and eavesdrop on a T1/
BRI/PRI/DSS1 circuit.. that takes way more hardware but POTS you
just need a lineman's handset.
But yes true secure will need to be end to end.
/b
On May 7, 2009, at 10:35 AM, Paul wrote:
Yes, I've seen this
You're right. Digital circuits are not so easy to tap into as opposed to POTS.
I'm thinking about this issue because I'm wondering how the US Govt. sets their
DRSN lines up to be secure. From what I read, only Raytheon and Telecore have
DRSN-use JITC-approved switches. It seems that list may be
If I understand correctly the OpenVox card is a zaptel clone, which means
you'd need to download the drivers, install them and then do the OpenZAP
install. See the OpenZAP page on the wiki.
-MC
On Thu, May 7, 2009 at 5:46 AM, Fred-145 codecompl...@free.fr wrote:
Hello
I browsed the wiki +
I see, but it should work with ESL too right?
Diego
On Thu, May 7, 2009 at 7:55 AM, Mikael Aleksander Bjerkeland
mik...@bjerkeland.com wrote:
EventMachine is very different to TCPSocket and is definitely not a
drop-in replacement. Take a look at FreeSWITCHeR
SIP TLS will protect the SIP session information with static keys via a
certificate, assuming of course the call is direct between two peers.
It will do nothing for the actual voice channel.
There is SRTP, which can be used to create a cryptographic context over
RTP. However, the key question is
mercutioviz wrote:
If I understand correctly the OpenVox card is a zaptel clone, which means
you'd need to download the drivers, install them and then do the OpenZAP
install. See the OpenZAP page on the wiki.
Thanks for the tip. I'll give it a shot. In the mean time, I'm interested in
On Thu, May 7, 2009 at 9:26 AM, Fred-145 codecompl...@free.fr wrote:
mercutioviz wrote:
If I understand correctly the OpenVox card is a zaptel clone, which means
you'd need to download the drivers, install them and then do the OpenZAP
install. See the OpenZAP page on the wiki.
Thanks
Antonio Gallo wrote:
Alix cases are like 6/9 € from their shop site. I think its easy to find
someone who work with aluminium that can make for you custom boxes for
like like 6/20 € at pcs
Unfortunately, none of the PCEngines cases (www.pcengines.ch/order1.php?c=2)
allow for a PCI slot,
Hey David!
You should come by to this year's ClueCon!
We still have some speaking slots left.
On Thu, May 7, 2009 at 11:08 AM, David Sugar dy...@gnutelephony.org wrote:
SIP TLS will protect the SIP session information with static keys via a
certificate, assuming of course the call is direct
If I can find funding for travel presently I would.
Anthony Minessale wrote:
Hey David!
You should come by to this year's ClueCon!
We still have some speaking slots left.
On Thu, May 7, 2009 at 11:08 AM, David Sugar dy...@gnutelephony.org
mailto:dy...@gnutelephony.org wrote:
SIP
I had this same problem but eventually overcame it. I modified the docs at
http://wiki.freeswitch.org/wiki/Amazon_ec2
accordingly. I think the problem I had was the internal vs. external IP
address, as you've alluded to at the bottom of your message.
In addition to the changes you've made, I
Ok, this seems to work:
require 'rubygems'
require 'eventmachine'
module CallingCard
def post_init
send_data sendmsg\ncall-command:
execute\nexecute-app-name: answer\n\n
send_data sendmsg\ncall-command:
execute\nexecute-app-name: playback\nexecute-app-arg:
It seems like EM (EventMachine) can't be used with ESL.
16:41 diegoviola thedonvaughn: i see, so ESL itself can't be used with EM?
16:41 wyhaines You can't just hand the socket from EM to ESL.
16:42 thedonvaughn prolly not
16:42 thedonvaughn and if tmm1 doesn't know how, then i'm going to say
Hi there,
I know there's a feature that allows us to create a uuid and then use
that uuid to do an originate (api create_uuid). Is there any badness if
I don't do the create_uuid and just do the originate using my own
uuid.
I experimented and made the following originate call:
api originate
I would use real UUID's not something as simple as abcdefg cuz FS will
hang up on you if you collide.
/b
On May 7, 2009, at 3:47 PM, Simon Tang wrote:
Hi there,
I know there's a feature that allows us to create a uuid and then
use
that uuid to do an originate (api create_uuid). Is there
On Thu, May 07, 2009 at 01:47:26PM -0700, Simon Tang wrote:
Hi there,
I know there's a feature that allows us to create a uuid and then use
that uuid to do an originate (api create_uuid). Is there any badness if
I don't do the create_uuid and just do the originate using my own
uuid.
I
How about something like this :
{origination_caller_id_number
=16041234567,originate_timeout=60,origination_uuid=$
{create_uuid()}}sofia/gateway/icall/16041234567 park
You can call an api with ${apiname()} and the results will go into its
place.
/b
On May 7, 2009, at 4:18 PM, Simon Tang
Also fix your evil mail server cuz I tried tried to email you and it
bounced back.
You have failed to detail your scenario more... options used .. how
you connect things.. every detail counts no matter how small.
Then collect it up and put it on Jira.
/b
On May 7, 2009, at 8:37 AM,
it's up to you what you do,
break it you buy it =D
On Thu, May 7, 2009 at 4:26 PM, Brian West br...@freeswitch.org wrote:
How about something like this :
{origination_caller_id_number=16041234567,originate_timeout=60,origination_uuid=${create_uuid()}}sofia/gateway/icall/16041234567
park
Anthony Minessale anthony.miness...@gmail.com wrote:
you may have a sonus infection
try some of the stuff from here under DTMF
http://wiki.freeswitch.org/wiki/RTP_Issues
Thank you for the suggestion.
I tried both the Sonus and Cisco settings in the external profile (running
sofia profile
Hi guys,
Nevermind with the ESL and EM thing.
I was wondering what the getBody() getHeader() and other ESL stuff
does behind the scenes, in raw socket, do you know?
Thanks,
Diego
On Thu, May 7, 2009 at 4:44 PM, Diego Viola diego.vi...@gmail.com wrote:
It seems like EM (EventMachine) can't be
Diego Viola diego.vi...@gmail.com wrote:
Hi guys,
Nevermind with the ESL and EM thing.
I was wondering what the getBody() getHeader() and other ESL stuff
does behind the scenes, in raw socket, do you know?
Why not read the source code? This is free software and open-source, after
all.
Its just status
Also please do not hijack threads.. Click new message and input the
address freeswitch-users@lists.freeswitch.org
Thanks,
Brian
On May 7, 2009, at 8:13 PM, Lars Zeb wrote:
I have installed from the 1.0.4pre7 tarball on a openSuse 11.1.
Why is it that after I launch
Check out the wiki for more info.
Wiki.FreeSWITCH.org
On Thursday, May 7, 2009, Lars Zeb larc...@yahoo.com wrote:
I have installed from the 1.0.4pre7 tarball on a openSuse 11.1.
Why is it that after I launch freeswitch and type in either 'show' or
'status' at the console, it responds with
A quick update: I can still reproduce the profile startup failure under
revision 13246, but I haven't hit the segfault again.
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
Lars Zeb larc...@yahoo.com wrote:
I have installed from the 1.0.4pre7 tarball on a openSuse 11.1.
Why is it that after I launch freeswitch and type in either 'show' or
'status' at the console, it responds with 'Unknown command', but it does
accept 'shutdown'?
Maybe the mod_commands module
Sorry for the hijacking, too anxious.
I typed only one of the two commands - both Unknown.
2009-05-07 17:52:52 [CONSOLE] switch_core.c:1322
switch_core_init_and_modload()
FreeSWITCH Version 1.0.4pre7 (13238M) Started.
Crash Protection [Disabled]
Max Sessions[1000]
Session Rate[30]
SQL
Weird did you modify anything in modules.conf?
/b
On May 7, 2009, at 9:01 PM, Lars Zeb wrote:
Sorry for the hijacking, too anxious.
I typed only one of the two commands – both Unknown.
2009-05-07 17:52:52 [CONSOLE] switch_core.c:1322
switch_core_init_and_modload()
FreeSWITCH Version
It works. Thanks.
But when I click the flash Play, it shows undefined. When I click
download, it shows another challenge box where I had to input the same
user/pass. Is that a problem?
On May 7, 2009, at 8:09 PM, Anthony Minessale wrote:
param name=http-allowed-api value=voicemail/
in
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