Hello,
I'm looking for a way to have G.729 codec working on Solaris/x86.
Binary codec from Digium is not compiled for Solaris/x86 (only sparc).
Are there any alternative (free or commercial) G.729 implementations,
which would work?
I saw something from Intel and got it to compile on Linux,
Hello,
are there any (possibly experimental) asterisk debian packages (or at
least a debian/ directory to build our own)?
Previously I used to modify debian/ directory from earlier version,
but 1.4 changed build process, so this is not that easy.
Thank you,
Juraj.
___
this SIP provider in my sip.conf
(about 10). All other providers are working correctly, this one
(except for these excessive registrations) is working too (all of the
accounts). I've been told by my voice provider, that they are also
using Asterisk on their side.
I've tried upgradi
Hello,
I would like to create some form of reporting of call quality. Is
there a way to collect quality of RTP data (for SIP calls) to gather
some statistics (packet loss, ...). I would like to know when calls
are of lower quality and if I should blame ISP, operator or look for
some problems o
Hello,
Did you try a combination of qualify=yes in sip.conf and then try the
ExtensionState in the manager?
yes, I have qualify=yes in the IAX config for peers I'm interested in.
Seems like if qualify=yes or 2000... whatever, is not set then asterisk will
not always know the state of the pho
,
Juraj Bednar.
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Hello,
Google is your friend:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+ExtensionState
not today.
I mentioned in my original mail, that ExtensionState is unrealiable
too. Sometimes I quit my softphone and I see extension as "Idle"
(status 0), sometimes I log in
Hello,
I would like to somehow get the presence of IAX2 and SIP users from
Asterisk Manager API or using any other means.
I tried watching for PeerStatus event, but it seems unrealiable
(http://bugs.digium.com/view.php?id=7833).
I tried defining hint for user and sending ExtensionState event,
o my question is how to make it work. What am I doing wrong? Or is there a simpler solution? Is there a reason why incominglimit=1 in
iax.conf does not work? I have asterisk 1.2.7 (with security patches). Any help appreciated. Thanks,
Hello,
I would like to ask, if there's a way to transfer a call from some
external program? I would like to build something like Asterisk Flash
Operator Panel, with the ability to transfer a call using drag and drop.
So I would like to connect to asterisk command line interface and
transfer one s
Hello,
I have a polycom ip600 and eyebeam. When I call from polycom to
eyeBeam, everything, including audio works. When I call the other side
(from eyeBeam to polycom), I get no audio. In both cases, eyeBeam shows
the same codec: g711u. Also sip show channels shows ulaw codec for both
sides and
Hello,
has anyone seen or created a Debian Sarge package for 1.2beta1?
I saw some for Sid, but I prefer not upgrading glibc right now, as
this is a production server (Asterisk on it will be for testing).
Thanks,
Juraj.
___
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Hello,
> I went over the code. AES128 is the only algorithm that is suppored
> today. More importantly there are some concerns on the vulnerability as
> discussed in
> http://lists.digium.com/pipermail/asterisk-security/2005-August/60.html.
> People are using UDP VPNs to satisfy customer requi
Hello,
> What's the status on using eyebeam with Asterisk, does it still
> require a patch to Asterisk to support the video component? I'm
> intererested in starting to use Video and audio telephony but wary of
> anything that requires patches.
cvs head works out of the box, just enable the h.32
Hello,
I use iaxcomm-latest from the iaxclient.sf.net page (binary
release) on linux, also tried Mac OS X version with the same result
and Asterisk 1.0.9 from Debian. Iaxcomm has a huge latency -- tens of
seconds, constantly changing over time. It was run on two different
machines, always to a
Hello,
I added queuing support (based on SQLite database to store the queue)
for my SIP Messaging patch. Works with eyeBeam, probably lots of bugs,
but it's at least something.
I created page about installation on the tips and tricks of voip-info.org:
http://www.voip-info.org/tiki-index.php?p
Hello,
> There's some work on creating a multiprotocol solution for instant
> messaging within Asterisk, but it will not be in the coming v1.2.
is the work somewhere as a patch to be tried or in some other form,
even if it's not coming to 1.2?
Juraj.
___
Hello,
I just got my Soekris 4801 box for use with Asterisk, but not as a
primary Asterisk server.
> * [EMAIL PROTECTED] (Is @home or regular better?)
If you want to run from CF, I recommend running some distribution
(that does not take much space) and your own Asterisk... I'm not even
sur
Hello,
> > I found, that in CVS Head, in chan_sip.c, there's some support of
> > asterisk. I have one question -- how does it map extensions to sip
> > user names? When my client "subscribes" to other extensions' presence,
> > they see all users online, but it may be because of voicemail
> > fall
Hello,
I found, that in CVS Head, in chan_sip.c, there's some support of
asterisk. I have one question -- how does it map extensions to sip
user names? When my client "subscribes" to other extensions' presence,
they see all users online, but it may be because of voicemail
fallback. Is there a wa
Hello,
> There's this device called VoiceBlue GSM gateway.
> It talks gsm on one side and SIP on the other side.
> Have a look at:
> http://www.voip-info.org/tiki-print.php?page=How+to+connect+VoIP+GSM+gateway+to+Asterisk+PBX
yep, but it is very expensive, I found. Even cellphone + cellsocket +
F
some. Any ideas?
Sincerely,
Juraj Bednar.
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Hello,
I was again asked to try to add support for presence
(SUBSCRIBE/NOTIFY) and IM using SIMPLE. I have few questions:
a.) are there any, at least partial projects, patches, anything,
that at least partly implements presence and/or IM to chan_sip? I
don't care about presence on other cha
Hello,
> > I know there are wifi sip phones out there but I have a
> > question, are any of these phones "anti explosive"? By that I
> > mean, there are certain regulations about phones or cel
> > phones that are not recommended to operate in environments
> > like gas stations due to sparks and th
Hello,
> > We have been running Asterisk for about a month now and one of the
> > things I miss the most is the ability to se who's online and
> > available and who's not. Surely, there's the manager interface, but
> > unless you'd want to install extra software on each client computer,
> > this i
Hello,
I would like to use Asterisk with fayn.cz service. They should be
using a standard H.323
protocol, but I have no more information about this. They provide a
softphone and/or rebranded
H.323 telephone, but I don't know any H.323 settings nor if the
firmware in the phone is
modified. Has
Hello,
> Has anyone successfully used a standard bluetooth enabled system to
> connect to a standard bluetooth enabled mobile phone (not the bluetooth
> to FXS converters) to create an audio path for phone calls with
> asterisk, if so is there a writeup on what was done so that others can
> replic
Hello,
I would like to implement a home GSM gateway using asterisk. What
would you recommend me as a low-cost hardware for creating a gsm
channel? I found 2n gsm gateway, that supports sip and chan_blue for
bluetooth connections. Any recommendations?
Basically, I want to end calls to some GSM
ort in asterisk?
Any pointer to documentation describing this is welcome.
One more question -- is there a video conferencing support (like
meetme, but for video)?
I also found some development pages, but without code...
Thanks,
Juraj Bednar.
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