we use SIP trunks with t.38 (supposedly)
which patton (1 or 2 FXS) would you recommend ?
how about audiocodes mp-11x ? I rather stay away from the grandstream and
the likes...
-gabriel
On Fri, 15 Jun 2012 17:09:22 -0400, Tony Graziano
wrote:
> We use Patton fxs gateways.
>
> Fa
(dtmf
issues among other things - those where connected to a different phx not
sipx)
thanks,
-gabriel
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" and the cisco phones are experimental. We also know the SIP
firmware on the cisco is not on par with other phones or with their own
sccp fw. In the mean time some of us have used these phones with other
other open source PBX systems or call manager we're just tring to make
them wor
ider.
any idea what these entries might be ? Or which log file should I look for
more info ?
thanks
-gabriel
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In my tests I found 8.5.4 to be the most stable
also didn't have any problems with making/receiving calls between cisco or
other phones (grandstream/innomedia/snmo/softphones)
-gabriel
On Sun, 28 Mar 2010, Rhon wrote:
Hi Ben,
I'm using sip 8.5(3) firmware. Using Cisco 7970G ph
just add your sipx server in the filed "processNodeName" on "Node Name"
and then it will work.
it took a while for me to find a fw that doesn't have issues so good luck
with that ;)
-gabriel
On Fri, 26 Mar 2010, Rhon wrote:
Hi,
Thanks for the link. I've al
(playback_terminators=1230*)
---
On Wed, 17 Feb 2010, gabriel wrote:
Hello list,
I have the following issue with the voicemail: every now and then, when
somebody soft spoken leaves a voice mail they get interrupted by the silence
detection threshold and unfortunately they don’t bother to re-record and
Hello list,
I have the following issue with the voicemail: every now and then, when
somebody soft spoken leaves a voice mail they get interrupted by the
silence detection threshold and unfortunately they don’t bother to
re-record and speak louder instead they hang up. If there where for me I
until this gets fixed I patched it with a cron job
for i in `find /var/sipxdata/mediaserver/data/mailstore -name "*xml" -print | grep
saved`; do sed -i 's/sip:.*/sip:2...@sipx.company.net\>\<\/from\>/g' $i; done;
-gabriel
On Fri, 29 Jan 2010, Todd Hodgen
ings, I
don't have any of those, even disabled the conferencing altogether.
-gabriel
On Fri, 29 Jan 2010, Todd Hodgen wrote:
I found that when I pulled two specific messages out of my voicemail, the
problem went away. It has worked ever since I did that with no problems.
Running the same
ind a workaround rather that re-installing the stable version
I do not have any conference recordings at the moment. Was trying to
listen to the saved messages dialing in from outside over the sip trunk
but I'll be onsite in a bit an will try it from a local extension as well.
-gabriel
On F
right, it's the dev version.
in the mean time I searched on sipx-dev and saw the bug :(
not sure what the conf recordings had to do with it but will try what you
said now.
-gabriel
On Fri, 29 Jan 2010, Todd Hodgen wrote:
> What version are you running?
>
> If you are runnin
so I can listent to the voice mail messages only from the web interface,
the call gets disconnected when trying it from the phone.
I see this error in the log:
---
"2010-01-29T21:15:04.782000Z":18063:sipXivr:INFO:sipx.company.net:Thread-359::sipxivr:"Retrieve::playMessages
SAVED Mail
out doesn't have one assigned
is it really that they don't support this or am I doing something wrong ?
-gabriel
On Sun, 24 Jan 2010, Pizza Napoletana wrote:
> On Jan 24, 2010, at 6:52 PM, M. Ranganathan wrote:
>>> But Speakeasy gave me a whole bunch of parameters
so on the browsers I had the 8443 godaddy 2048b cert working fine for a
while now (with intermediates)
then the voicemail PIN change issue came up.
my fix:
1. downloaded from godaddy the root CA + two intemediates
(https://certs.godaddy.com/anonymous/repository.seam)
valicert_class2_root.crt
I just configured it this morning and running tests now.
I haven't got the incoming calls to go directly to each user based on the
user DID yet and still working on that but everything else worked out
fine so far. (users calling outside with their own external number, good
voice quality, DTMF w
I've had the same problem. I am building a new pbx and I have about 10
phones (7971) registered, after a few days I noticed about 3 phones have
lost the registration, not sure why and still investigating.
-gabriel
On Sat, 2 Jan 2010, Nathan Nieblas wrote:
I’ve just recently dep
wow, that worked right away, thanks for the tip !
now working on other stuff like directory xml services etc.
apparently there is a bug opened for this issue we've been having
http://track.sipfoundry.org/browse/XX-5358
- gabriel
On Tue, 15 Dec 2009, Dylan Ebner wrote:
> Gabriel-
&g
e any help, I am a newbie and trying to switch from *
to sipx
-gabriel
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