rd SRTP encryption.
>
>
> So, back to my origianal question then. Are there any ATA's that
> support TLS AND SRTP with FreeSwitch?
>
>
> On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri wrote:
>> AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key
>
AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key
exchange to appropriately support SRTP and FreeSWITCH. They do their
proprietary Sipura key exchange only, not sure if Cisco plans on
upgrading the firmware to ever support SDES on the ATAs. They added
support for SDES to their IP Pho
e,
>
> I don't think any of them are plus the T.38 SDP tells me the bitrate
> is 14400, certainly not V.34 speed.
>
> Are you saying the machine even trying to negotiate V.34 poses a problem?
>
> On Thu, Oct 22, 2009 at 2:16 PM, Gabriel Kuri wrote:
>> Ou
Out of curiosity, is it a Super G3 (ie v.34) capable FAX? We've had
nothing but intermittent problems with Super G3 FAXes over T.38, unless
v.34 is strictly turned off on the machine.
Gabe
Kristian Kielhofner wrote:
> On Thu, Oct 22, 2009 at 11:58 AM, Tihomir Culjaga wrote:
>
> One fax machin
We've been testing mod_nibblebill, it's a great module, cheers to the
author!
A couple questions regarding nibblebill:
1) We noticed that when an account is below the minimum balance and a
call is attempted with that account, FS begins to connect the B leg for
the call but then cancels the INV
Brian West wrote:
> Do you have any reason to be doing proxy media?
no, not other than to fix the one way audio issue :)
I'd rather leave proxy_media off.
Gabe
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We're seeing occasional one way audio issues for international calls
going out to one of several carriers. On roughly 2 out of 5 calls
outbound, there is no audio on the the calling party's side, however the
called party indicates they can hear the calling party perfectly well.
NAT is not involved
I heard about this a few days ago, they claim it's not them, but someone
trying to "harm their reputation" ...
http://www.meucci-solutions.com/complaints.asp?id=1
Gabe
Brian West wrote:
> Does anyone else seem to be getting tons of calls from this evil IP?
> They keep ringing me via SIP
> This part is interesting, and the subject of a discussion we had
> recently. A number of systems try that second re-invite after a 488, but
> the SIP specs say the call is pretty much dead after the 488 message is
> exchanged. Are they just hoping that maybe the other end will be
> non-compliant
n on port 443 and are
> encrypted :P
>
> /b
>
> On Mar 30, 2009, at 1:31 PM, Gabriel Kuri wrote:
>
>> eh, how are poeple doing VoIP over there, given use of it outside the
>> UAE is "officially" outlawed by the TRA ? I've heard Etisalat is pretty
>>
eh, how are poeple doing VoIP over there, given use of it outside the
UAE is "officially" outlawed by the TRA ? I've heard Etisalat is pretty
strict with making sure it's blocked going outside the country via a L7
packet inspection device to drop SIP.
~Gabe
Bipin Patel wrote:
> hi,
>
> i current
aries.
>
> When we get some spare time we might go back and turn them on but so far
> we don't have much of a need to.
>
>
> On Fri, Mar 27, 2009 at 1:56 PM, Gabriel Kuri <mailto:gk...@ieee.org>> wrote:
>
> I'm not asking what they do, I'm a
ok, thanks, that pretty much amounts to the gcc defaults. perhaps I
should recompile FS with those defaults and see if the current jira I
have open goes away ;) ...
Gabe
Brian West wrote:
> We usually don't specify anything extra!
>
> /b
>
> On Mar 27, 2009, at 1:56 PM
then you shouldn't use them! ;)
>
>
> /b
>
> On Mar 27, 2009, at 1:19 PM, Gabriel Kuri wrote:
>
>> regarding gcc compiler optimizations, are they generally compatible with
>> FS or should they be removed or does the configure strip them out? just
>> cur
regarding gcc compiler optimizations, are they generally compatible with
FS or should they be removed or does the configure strip them out? just
curious, as I run Gentoo and use such optimizations as "-march=nocona
-O2 -pipe -fomit-frame-pointer"
not sure if they break things or I should be removi
OK, I'll give it a try and report back.
Gabe
Brian West wrote:
> Make current and try again... I haven't seen this crash you have seen...
> if you can run sippcapdump and get the packets that would help also.
>
> Thanks,
> /b
>
>
> On Mar 20, 2009, at 4:41
20, 2009, at 4:28 PM, Gabriel Kuri wrote:
>
>> hey folks, I'm trying to configure PCMU fallback for T.38.
>>
>> The originating endpoint (Linksys SPA-2102) sends an INVITE to FS with
>> G729 and PCMU in the sdp. the INVITE to the provider includes G729 and
>> PCMU a
hey folks, I'm trying to configure PCMU fallback for T.38.
The originating endpoint (Linksys SPA-2102) sends an INVITE to FS with
G729 and PCMU in the sdp. the INVITE to the provider includes G729 and
PCMU as part of the sdp as well (absolute_codec_string=G729,PCMU) ...
m=audio 16458 RTP/AVP 18 0
is the purpose of start_dtmf to detect it inband and convert it
> to 2833. Unless you disabled 2833.
>
> /b
>
> On Mar 13, 2009, at 3:36 PM, Gabriel Kuri wrote:
>
>> shoot me if I'm on the wrong track, but is it possible to use the
>> start_dtmf app to do inb
shoot me if I'm on the wrong track, but is it possible to use the
start_dtmf app to do inband dtmf detection and "convert" the inband DTMF
to rfc2833, as opposed to using the dtmf detection on a Linksys or
Grandstream ATA?
the reason I ask is the dtmf detection on these ATAs seems to falsely
catch
Someone can problem correct me if I'm wrong, however I believe a recent
change was made to the configure script to try and autodetect ODBC. The
configure script may be hitting a false positive as described in this
similar thread from a few days ago ...
http://lists.freeswitch.org/pipermail/freeswi
It sounds like your build environment is whacked, is this a fresh
checkout of trunk or did you overwrite an existing directory? You might
want to scrap that directory and try a fresh checkout.
I just freshly checked out a copy of trunk into a new dir and ran
./bootstrap, ./configure, and make with
sible but the uh, performance,
> would be interesting...
>
> Can anyone call me out on this assumption?
>
> On Tue, Feb 17, 2009 at 7:16 PM, Gabriel Kuri wrote:
>> awesome work! on a slightly related [embedded] note, do you know if any
>> work has been done to port FS to
awesome work! on a slightly related [embedded] note, do you know if any
work has been done to port FS to any of the Analog Blackfin MCUs? I'd be
interested in hearing if anyone has had any such luck.
Gabe
Kristian Kielhofner wrote:
> FreeSWITCH now compiles in AsLinux:
>
> http://www.astlinux.or
I'm about 99% positive that if https is enabled for remote provisioning,
the web server needs an SSL certificate signed by the Linksys
Enterprise CA, otherwise the phone will reject it.
Gabe
> but it shouldn't matter too much if they're using https or not, as long
> as the ata doesn't authenti
p. I
> will report back when I do.
>
> Thanks so much -
>
> Library Mark
>
> Quoting "Gabriel Kuri" :
>
>> I believe you need to make sure the Ethernet cable is unplugged from the
>> phone when trying to dial that string.
>>
>> Now I'
It depends on the whether you pass the option to the Linksys/Cisco
Profile Compiler to generate the config file. In any case, that
shouldn't be an issue.
Gabe
Brian West wrote:
> Don't they cryptographically sign the config also?
>
> /b
>
> On Feb 17, 2009, at 2:
me other hardware fix for this?
>
> Quoting "Gabriel Kuri" :
>
>> Have you tried resetting the phone via the built-in IVR menu?
>>
>> Pick up the handset and dial 73738#
>>
>> This should reset the phone to factory defaults, assuming that company
Have you tried resetting the phone via the built-in IVR menu?
Pick up the handset and dial 73738#
This should reset the phone to factory defaults, assuming that company
didn't lock this feature out.
Gabe
Mark wrote:
> Hello, folks - I hope that I can reach someone who knows the answer to
Hey Folks:
For a FS box that's generally handling higher amounts of
inbound/outbound call traffic (say 500 - 700 calls) and registrations
(30 - 50 per second), is it recommended to split off the signaling and
media traffic onto separate NICs for performance reasons? Or is it
better to split all th
I don't know if this helps, but I attached a config file generated by
CUCM for a 7975. I don't believe CUCM writes out all the possible config
options into the XML file, although it does write out quite a bit.
If you're looking for another option, let me know and I can see if I can
enable it and w
I've been experiencing the same since I upgraded to the latest rev of
trunk from rev 3, but I've been waiting to collect more data before
opening a jira and try to at least narrow down which revision broke things.
Laurent - If you open a jira, I'll add the output of the core dumps,
debugging i
I've tried to do the same and in my own experience, most carriers don't
accept 302 redirects. What I've seen is they take the 302 as a failure
and move on to the next switch, so worse case with 3 switches, it will
take 2 retries before hitting the switch you want them to redirect to.
Gabe
Dennis
p in /root ?
Gabe
Brian West wrote:
> If you're running SELinux then you'll need to correct that on your
> machine to allow FreeSWITCH to write to /tmp
>
> /b
>
> On Dec 9, 2008, at 2:43 AM, Gabriel Kuri wrote:
>
>> I've been experimenting with mod_fax
I've been experimenting with mod_fax and discovered it doesn't appear to
receive faxes unless freeswitch is running as root? it fails trying to
open the tiff file for writing (see the logs below). I'm using the
dialplan as prescribed in the wiki without any changes and the user the
freeswitch proce
How about:
"The mailbox of the person you are trying to reach is full and can not
accept new messages at this time. Please try your call again later. Goodbye"
~Gabe
Tamas Cseke wrote:
> Hello,
>
> I would like to use a disk quota in users' voicemail
> (http://jira.freeswitch.org/browse/MODAPP-1
il, but if the phone is actually ringing and no one picks
up in 30 seconds, send it to voicemail?
Thanks
Gabe
Brian West wrote:
> Try pre_answer before bridge.
>
> /b
>
> Sent from my iPhone
>
> On Nov 28, 2008, at 3:03 PM, Gabriel Kuri <[EMAIL PROTECTED]> wrote:
>
my iPhone
>
> On Nov 28, 2008, at 3:03 PM, Gabriel Kuri <[EMAIL PROTECTED]> wrote:
>
>> I have a phone that is registered to FS but is no longer available
>> (Internet connection down, phone turned off, etc.). The registration
>> still exists in the sip_registration
I have a phone that is registered to FS but is no longer available
(Internet connection down, phone turned off, etc.). The registration
still exists in the sip_registrations table (not expired yet), but the
phone is not reachable on the network.
According to my dialplan, if the bridge to the phone
Marc,
I'll chime in since I'm currently in the process of building a very
similar environment...
I currently have two FS boxes using xml_curl for configuration,
dialplan, and directory data. All sip session info and voicemail data is
stored in the mysql db which is on two multi-master mysql boxes
0.
m=audio 25454 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
So is this an underlying issue with the linksys spa units or FS?
Gabe
Gabriel Kuri wrote:
> I'm having an issue with the linksys spa devices when ena
Sorry, I should've included that in the original mesg...
The logs I posted were from trunk-r10055. I updated this morning to
trunk-r10081, still the same issue.
~Gabe
Brian West wrote:
> Which version of FreeSWITCH are you using?
>
> /b
>
> On Oct 20, 2008, at 2:25 AM
I'm having an issue with the linksys spa devices when enabling inbound
proxy media mode (inbound-proxy-media=true) and late negotiation
(inbound-late-negotiation=true) in the sofia profile. The spa
immediately sends a BYE when the call is answered by the called party.
For whatever reason, it works
void this issue with transcoding?
Someone posted a similar question on jira, but it doesn't look as though
it had been answered ...
http://jira.freeswitch.org/browse/MODENDP-46
Thanks for the help,
Gabriel Kuri
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I was experiencing the same exact issue and I tracked it down to the
MySQL interactive_timeout and wait_timeout variables. I had both of the
variables set to 900 seconds, which was too low compared to the interval
the phones are set to register with the server. Basically I discovered
you need to se
e
using G.726-32 and I had files encoded with G.726-32 in a folder
'sounds/voicemail/g726-32/', how do I tell FS to use those sounds files
based on the user's codec?
Thanks,
Gabriel Kuri
Lucas Cornelisse wrote:
> Hi Gabriel,
>
> The problem I see is that g729 is protec
ayback the files
for the appropriate codec?
Thanks Much...
Gabriel Kuri
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