Is anyone using a SIP service they like that works with generic SIP
clients like linphone?
- Grant
?
Is there anything in particular that I have to mention at the installation?
Thanks in advance,
Tom
--
sip:3...@perenaster.com
sip:3...@perenaster.com
sip:3...@perenaster.com
sip:3...@perenaster.com
sip:3...@perenaster.com
On Thu, Feb 16, 2012 at 1:56 PM, Grant emailgr...@gmail.com wrote:
Is anyone using a SIP service they like that works with generic SIP
clients like linphone?
I use Callcentric, have for a few years now as my home phone. It
works with Linphone or any generic SIP client or hardware. You can
sign
.
But for the SIP stuff, I have just one client, built the firewall using
fwbuilder (sometimes is more easier), and for instance here's the SIP
part on the nat table:
0 0 DNAT udp -- anyany anywhere
200.*.*.* udp dpt:5060 to:10.0.0.112
Is this wrong
I'm having great success with packet shaping via shorewall but I'm not
sure I have my ports prioritized correctly for SIP and skype. twinkle
is set to use 5060 and 8000 and skype is set to use 23399 for inbound
connections. Should I prioritize 5060, 8000, and 23399 for both SRC
and DEST? I'm
Any preferred SIP softphone on Gentoo? I've used Ekiga on Windows and
Ubuntu, and it's only moderately painful there, but it's very, very,
very old on Gentoo.
If there are no good options, I'll stick with the SIP client on my
android phone, for now. (hardphone isn't an option)
--
:wq
On Thu, 16 Feb 2012 11:56:52 -0800
Grant emailgr...@gmail.com wrote:
Is anyone using a SIP service they like that works with generic SIP
clients like linphone?
- Grant
I like sip2sip.info the most!
--
Willie Matthews
matthews.wil...@gmail.com
Dale writes:
I'm doing my KDE4 upgrades and ran into this:
[...]
Error: This version of PyQt requires SIP v4.13.3 or later
[...]
I notice tho that portage seems to have failed to notice this was
needed. Should I file a bug report or is this just me?
File a bug. There is a DEPEND line
How to make net-misc/asterisk-16.30.1 into binary package so I can install in
on future gentoo boxes.
I think asterisk ver. 16 (still in portage) is the last one still compatible with
sip/iax code all future versions starting with ver.18 are converting sip =>
pjsip
that is not compati
Apparently I'm out of luck as far as using skype or wengophone on a
hardened profile. Is there are kind of a similar service available
that would work on a hardened amd64 system? I understand that I can
use a client like sjphone and connect it to any sip service, but I
like seeing my prepaid
On Thu, 16 Feb 2012 12:51:27 -0800
Grant emailgr...@gmail.com wrote:
Is anyone using a SIP service they like that works with generic SIP
clients like linphone?
- Grant
I like sip2sip.info the most!
I should have said I need to be able to call regular
phone number from the SIP
You can still use
On Sun, 04 Feb 2024 02:47:19 -0500,
Thelma wrote:
> How to make net-misc/asterisk-16.30.1 into binary package so I can install in
> on future gentoo boxes.
>
> I think asterisk ver. 16 (still in portage) is the last one still compatible
> with sip/iax
In v18 sip is still present but deprecated - after this its removed.
There is a conversion script (sip->pjsip) for migration. It required a
few sacrificial chickens and much swearing until I got the upstream
trunk to register (iinet in AU). Its all working good now, the pjsip
con
Mateus Interciso p.zarnick at gmail.com writes:
But for the SIP stuff, I have just one client, built the firewall using
fwbuilder (sometimes is more easier), and for instance here's the SIP
part on the nat table:
0 0 DNAT udp -- anyany anywhere
200
Whats a good SIP/VoIP soft phone for linux (linphone, gnophone,
SimpleH323, ...) I need something simple to set things up with the
least hassle, and then perhaps something good to actually use - if they
dont overlap(!)
BillK
--
gentoo-user@gentoo.org mailing list
HIi,
I searching for a tool, which can be used to test VoIP clients for
being SIP compliant (RFC 3261 and others) and robust.
Does anyone know such a tool ?
Thank you very much for any help in advance!
mcc
--
gentoo-user@gentoo.org mailing list
!!! Multiple package instances within a single package slot have been pulled
!!! into the dependency graph, resulting in a slot conflict:
dev-python/sip:0
('ebuild', '/', 'dev-python/sip-4.8.1', 'merge') pulled in by
=dev-python/sip-4.8.1 required by ('ebuild', '/',
'dev-python/PyQt4-4.5.1
in new
slots), Size of downloads: 390,805 kB
!!! Multiple package instances within a single package slot have
been pulled
!!! into the dependency graph, resulting in a slot conflict:
dev-python/sip:0
('ebuild', '/', 'dev-python/sip-4.8.1', 'merge') pulled in by
=dev-python/sip-4.8.1 required
downgrades, 117 new, 117 in new
slots), Size of downloads: 390,805 kB
!!! Multiple package instances within a single package slot have been
pulled
!!! into the dependency graph, resulting in a slot conflict:
dev-python/sip:0
('ebuild', '/', 'dev-python/sip-4.8.1', 'merge') pulled in by
=dev
My bluetooth headset was working great with twinkle, but now it's out
of the tree and I can't find another SIP client that works with a
bluetooth headset. Does anybody know of one? linphone is said to
work, but I can't make it happen even after an exhaustive effort.
- Grant
Well, I just tried something that seems to work,
but has me confused or missing the routine reading
of new portage features.
Anyway upon a routine update (using portage 2.2_rc33
and sets for kde4) I got a message:
All ebuilds that could satisfy =dev-python/sip-4.8.1 have been masked.
One
On Mon, 22 Aug 2011 08:03:02 -0400, Michael Mol wrote:
Any preferred SIP softphone on Gentoo? I've used Ekiga on Windows and
Ubuntu, and it's only moderately painful there, but it's very, very,
very old on Gentoo.
Ekiga 3.2.7 is the latest on their web site and it has been in portage
for just
Yes, was caught out recently by the replacement of sip with pjsip -
currently on v21.0.2 and working (sip only, simple home setup) Also had
some weird problems with two versions installed (so asterisk started on
old working version even though new one was installed - once I ran
depclean
man quickpkg
On 4/2/24 15:47, Thelma wrote:
How to make net-misc/asterisk-16.30.1 into binary package so I can
install in on future gentoo boxes.
I think asterisk ver. 16 (still in portage) is the last one still
compatible with sip/iax code all future versions starting with ver.18
Grant wrote:
Apparently I'm out of luck as far as using skype or wengophone on a
hardened profile. Is there are kind of a similar service available
that would work on a hardened amd64 system? I understand that I can
use a client like sjphone and connect it to any sip service, but I
like
James wireless at tampabay.rr.com writes:
OOPS,
Further goolgling led me to bug
218874:
reemerging dev-python/sip fixed this bug for me.
fixed and closed.
James
Dale rdalek1967 at gmail.com writes:
You wouldn't happen to have that slot of python in your world file would
you?
nope,
world file only contains this entry relate to python:
dev-python/sip
James
When did they implement switch-over from sip to pjsip?
I'm using AudioCode boxes.
I emerged and tried to load asterisk ver.18 but the audiocode would not
register. I suppose ver.16 is the end of the line for me.
On 2/2/24 16:39, William Kenworthy wrote:
Yes, was caught out recently
On Tue, 07 Aug 2007 08:11:06 -0400, Colleen Beamer wrote:
On my laptop, I did an update world. One of the programs that the
update wants to install is 'sip'. I connect to my mirror and then the
mirror goes out to www.riverbankcomputing.com, but gets a 404 error.
Further to my previous mail
My bluetooth headset was working great with twinkle, but now it's out
of the tree and I can't find another SIP client that works with a
bluetooth headset. Does anybody know of one? linphone is said to
work, but I can't make it happen even after an exhaustive effort.
- Grant
It works
n register to it.
Conversion scrip (sip->pjsip) will not do any good if the hardware (AudioCodes
boxes, Sipura and other units) are not compatible with pjsip.
Is IAX is gone as well in newer versions?
I have an impression this is the end of old Asterisk that Digium started; very,
very sad :-/
It had g
.
All I know is, compared to other 'server'-oriented distros (or distro
variant), Gentoo has the least amount of memory usage :-)
Rgds,
--
Pandu E Poluan
~ IT Optimizer ~
Visit my Blog: http://pepoluan.posterous.com
--
sip:3...@perenaster.com
sip:3...@perenaster.com
sip:3
When I try to register provision/register equipment to asterisk I get an error
message:
[Dec 23 12:32:51] WARNING[21685]: db.c:350 ast_db_put: Couldn't execute
statement: SQL logic error
-- Registered SIP 'pstn-5665' at 10.0.0.110:5060
> Saved useragent "Audiocodes-Sip-Ga
On 12/23/2020 12:47 PM, the...@sys-concept.com wrote:
> When I try to register provision/register equipment to asterisk I get an
> error message:
>
> [Dec 23 12:32:51] WARNING[21685]: db.c:350 ast_db_put: Couldn't execute
> statement: SQL logic error
> -- Registe
On Fri, 02 Feb 2024 19:29:24 -0500,
Thelma wrote:
>
> When did they implement switch-over from sip to pjsip?
> I'm using AudioCode boxes.
>
> I emerged and tried to load asterisk ver.18 but the audiocode would not
> register. I suppose ver.16 is the end of the line for me.
that could satisfy =dev-python/sip-4.8.1 have been masked.
One of the following masked packages is required to complete your request:
- dev-python/sip-4.8.1 (masked by: ~amd64 keyword)
So I just added this line to package.keywords:
=dev-python/sip-4.8.1
and it got passed that problem
.
The VoIP PBX has a gateway; it is using sip trunks to provide phone
service. However, it will be severely locked down on install, I will
only let it talk to the sip trunk provider and its update server and
nothing else.
Does "I will only let the VoIP talk to the SIP trunk provider and
and
Jami softphones on android using SIP to a single asterisk.
I incrementally upgrade asterisk mostly by a clean install carrying over
config files into an LXC instance (using a golden master setup) which
has been trouble free until pjsip - and thats more my fault in missing
that SIP was
On Tuesday 07 August 2007 13:23, Neil Bothwick wrote:
On Tue, 07 Aug 2007 08:11:06 -0400, Colleen Beamer wrote:
On my laptop, I did an update world. One of the programs that the
update wants to install is 'sip'. I connect to my mirror and then the
mirror goes out
On Monday 13 August 2007 21:41:31 Mateus Interciso wrote:
Actually, I need a fully transparent bridge, for for instance, correcly
using a SIP phone, which even with siproxd, it doesn't work, so, NAT and
Masquerade, won't help me. I'm pretty sure I can transform the gentoo box
in a transparent
x show registry" the are registered with each
other,
otherwise the call wouldn't go through at all.
Here is the output, from both asterisks; one that works and one that doesn't:
===NOT WORKING=
"main-asterisk" (NOT WORKING):
== Using SIP RTP CoS mark 5
> 0x7f
Hi,
On my laptop, I did an update world. One of the programs that the
update wants to install is 'sip'. I connect to my mirror and then the
mirror goes out to www.riverbankcomputing.com, but gets a 404 error.
Anyone else seen this issue? I know what a 404 error is. Would
changing mirrors
Heh, of course now I see this mail
^_^
-Pariksheet
On Tue, Aug 26, 2008 at 3:54 PM, James [EMAIL PROTECTED] wrote:
James wireless at tampabay.rr.com writes:
OOPS,
Further goolgling led me to bug
218874:
reemerging dev-python/sip fixed this bug for me.
fixed and closed.
James
Hi all,
Apparently my SIP port 5060 is in use. Is there a Linux command to tell me
what process is using this port. I have a vague memory of seeing such a
command but just cannot remember or find it.
Many thanks in advance
Dave
--
gentoo-user@gentoo.org mailing list
On 6/15/06, Dave S [EMAIL PROTECTED] wrote:
Hi all,
Apparently my SIP port 5060 is in use. Is there a Linux command to tell me
what process is using this port. I have a vague memory of seeing such a
command but just cannot remember or find it.
netstat -l -p
-Richard
--
gentoo-user@gentoo.org
2006/6/15, Dave S [EMAIL PROTECTED]:
Hi all,
Apparently my SIP port 5060 is in use. Is there a Linux command to tell me
what process is using this port. I have a vague memory of seeing such a
command but just cannot remember or find it.
Many thanks in advance
Dave
Are you looking
Dave S wrote:
Hi all,
Apparently my SIP port 5060 is in use. Is there a Linux command to tell me
what process is using this port. I have a vague memory of seeing such a
command but just cannot remember or find it.
netstat -tulpen
netstat -tulpen | grep 5060
Alexander Skwar
--
BOFH Excuse
Im using xlite on my gentoo workstation and have no trouble with it.
Mit freundlichen Grüßen
Kai Zemke
**
IMPORTANT: The contents of this email and any attachments are confidential.
They are intended for
On Fri, 3 April 2015, at 12:25 am, Mick michaelkintz...@gmail.com wrote:
When people you need to communicate with on MSWindows boxes only know how to
manage Skype-ware and you don't run a SIP proxy server yourself, your choices
reduce somewhat.
We'll have to reincarnate Alexander Graham
relating to that and some other features.
The VoIP PBX has a gateway; it is using sip trunks to provide phone
service. However, it will be severely locked down on install, I will
only let it talk to the sip trunk provider and its update server and
nothing else.
Does "I will only let the VoIP
the gentoo
box, and I NATed the connection. So, basically, this is it.
You'll have to continue using NAT. Drop all bridge-related configuration
(i.e. keep away from brctl), configure the external interface to forward
connections.
Then you have to care for incoming connections. For a good SIP
On Mon, 13 Aug 2007 22:17:03 +0100, Mike Williams wrote:
On Monday 13 August 2007 21:41:31 Mateus Interciso wrote:
Actually, I need a fully transparent bridge, for for instance, correcly
using a SIP phone, which even with siproxd, it doesn't work, so, NAT
and Masquerade, won't help me. I'm
and sipconfig,
but have no idea what to do about those issues.
equery depends PyQt4
shows noting depends on this package
Any suggestions as to unmerging it, fixing it,
or just wait a few days, sync up and see if it is
fixed?
Ideas?
$ qlist dev-python/sip
...
/usr/lib64/python2.5
On 11 Jan 2010, at 18:31, Mark Knecht wrote:
...
MagicJack is just a USB FXS and a SIP client, except that it's tied
into a
proprietary service. There are lots of companies that sell VoIP
using the
SIP standard and I would expect there are many cheaper than
MagicJack,
because they don't
Hello,
Can anybody hint what on Earth this following message means, I've
never seen anything like it:
* Messages for package dev-python/sip-4.9.3-r1:
* ERROR: dev-python/sip-4.9.3-r1 failed:
* PYTHON(): '--active' option cannot be used in ebuilds of packages
supporting installation
a gateway; it is using sip trunks to provide phone
service. However, it will be severely locked down on install, I will
only let it talk to the sip trunk provider and its update server and
nothing else.
Dan
have to continue using NAT. Drop all bridge-related
configuration (i.e. keep away from brctl), configure the external
interface to forward connections.
Then you have to care for incoming connections. For a good SIP setup
with more than one SIP client, I'd highly suggest looking at SIP
proxies like
On Wednesday 22 October 2008 12:10:15 Daniel Pielmeier wrote:
2008/10/22 Alex Schuster [EMAIL PROTECTED]:
I think I have the same conflict. I just solved it by putting this into
package.keywords:
~dev-python/PyQt4-4.4.3
~dev-python/sip-4.7.7
[...]
This is maybe the PyQt issue I did
with Mepis Linux 3.4-3
http://www.mepis.org/
1(747)632-4973 SIP
Get Gizmo 1 cent per minuet calling
http://www.gizmoproject.com/
--
gentoo-user@gentoo.org mailing list
(FXS) to asterisk box.
Either get an FXS card, or an ATA to turn it into a SIP device.
--
Mike Williams
--
gentoo-user@gentoo.org mailing list
On Thursday 15 June 2006 18:21, Dave S wrote:
Hi all,
Apparently my SIP port 5060 is in use. Is there a Linux command to
tell me what process is using this port. I have a vague memory of
seeing such a command but just cannot remember or find it.
IIRC, it's fuser with the -n tcp option, eg
On Thursday 15 June 2006 17:21, Dave S wrote:
Hi all,
Apparently my SIP port 5060 is in use. Is there a Linux command to tell me
what process is using this port. I have a vague memory of seeing such a
command but just cannot remember or find it.
Many thanks in advance
Dave
Thanks for all
/site-packages/numpy/core/include -I. -o
qwt_version_info.o qwt_version_info.cpp
g++ -Wl,-O1 -o qwt_version_info qwt_version_info.o-L/usr/lib64/qt4 -lQtCore
-L/usr/lib64 -L/usr/lib64/qt4 -lgthread-2.0 -lglib-2.0 -lpthread
sip: Deprecation warning: ../sip/iqt5qt4/IQtModule.sip:32: %Module version
2008/10/22 Alex Schuster [EMAIL PROTECTED]:
I think I have the same conflict. I just solved it by putting this into
package.keywords:
~dev-python/PyQt4-4.4.3
~dev-python/sip-4.7.7
So I upgraded to PyQt4-4.4.3, and this depends on the splitted Qt ebuilds,
while PyQt4 up to version 4.4-r1
dumping my wired line.
MagicJack is just a USB FXS and a SIP client, except that it's tied
into a proprietary service. There are lots of companies that sell VoIP
using the SIP standard and I would expect there are many cheaper than
MagicJack, because they don't bundle it up in a package that's
Arttu V. wrote:
Hello,
Can anybody hint what on Earth this following message means, I've
never seen anything like it:
* Messages for package dev-python/sip-4.9.3-r1:
* ERROR: dev-python/sip-4.9.3-r1 failed:
* PYTHON(): '--active' option cannot be used in ebuilds of packages
supporting
var/cache/portage/tree/media-libs/libsdl/libsdl-1.2.15-r9.ebuild)
1445464904: ::: completed emerge (5 of 13) media-libs/libsdl-1.2.15-r9 to /
1445464904: >>> emerge (6 of 13) dev-python/sip-4.16.9 to /
1445464904: === (6 of 13) Cleaning
(dev-python/sip-4.16.9::/var/cache/portage/tree/de
with
a
cabled telephone?
The USB version for regular in-house phones.
Thinking about dumping my wired line.
MagicJack is just a USB FXS and a SIP client, except that it's tied into a
proprietary service. There are lots of companies that sell VoIP using the
SIP standard and I would expect there are many
microphone
(sip:[EMAIL PROTECTED]) ... It's frustrating at this point in time, but I
hope this is as easily solvable as the webcam issue.
Make sure your microphone is unmuted (alsamixer). If applicable, make
sure that your capture device is set to microphone (alsamixer -V
capture).
--
gentoo-user
, but might be related to bug190582
try either emerge PyQT-3.17.3
or
downgrade dev-python/sip to 4.5.2-r1
Best of luck,
W
--
You're very sure of your facts, he said at last, I
couldn't trust the thinking of a man who takes the Universe
- if there is one - for granted.
Sortir en
to do.
I'm not quite sure, but might be related to bug190582
try either emerge PyQT-3.17.3
or
downgrade dev-python/sip to 4.5.2-r1
Best of luck,
W
The last one worked!
Thank you.
emilio
--
[EMAIL PROTECTED] mailing list
and switched to Verizon.
Try: http://www.cellreception.com as a starter to narrow things down.
nice link
I'm going to find a portable wifi device, and see how it goes for
sip/Voip connections...
I'll have to figure out how to map/hook the wifi-to-home connections
over the Vonage line (unlimited
free with Mepis Linux 3.4-3
http://www.mepis.org/
1(747)632-4973 SIP
Get Gizmo 1 cent per minuet calling
http://www.gizmoproject.com/
--
gentoo-user@gentoo.org mailing list
/relnotes/stnlnxrn.htm
Cisco wants ~$40k per install to upgrade to a version that runs on a
modern OS. So while we write our own SIP Proxy Server... we are stuck
using rh7.1.
Any help would be great! Thanks!
(and yes, we have looked at SER but there are a number of key features
missing for us
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Bryan Whitehead wrote:
FWIW, The software is CSPS:
http://www.cisco.com/univercd/cc/td/doc/product/voice/sipproxy/relnotes/stnlnxrn.htm
There are other SIP Proxies in Portage you may like to test:
net-misc/partysip
net-misc/siproxd
- --
Arturo
I recently switched laptops and although the new one is faster, the
sound is much worse. lspci reports the sound card this way:
ALi Corporation M5451 PCI AC-Link Controller Audio Device (rev 02)
The problem is I need to use a sip phone (sjphone) and I can barely
make out what the other person
to use a sip phone (sjphone) and I can barely
make out what the other person is saying. Is there any way I can try
to improve the sound quality/performance? I am using the in-kernel
alsa drivers. Is it worth a try to emerge the ones in portage instead
or is that unlikely to help?
- Grant
Have
/nick.gpg.asc GPG Key ID: 04E4653F
GPG Fingerprint: 9732 D7C7 A441 D79E FDF0 94F6 1F48 5674 04E4 653F
SIP : [EMAIL PROTECTED]PSTN: 0560 0030509
pgpVpH4vHRyEn.pgp
Description: PGP signature
don't
think it's necessary, and mine works without it.
Good luck,
-Nick
--
GPG Key : www.njw.me.uk/nick.gpg.asc GPG Key ID: 04E4653F
GPG Fingerprint: 9732 D7C7 A441 D79E FDF0 94F6 1F48 5674 04E4 653F
SIP : [EMAIL PROTECTED]PSTN: 0560 0030509
pgpPn37WVQUzH.pgp
Description: PGP
and i'm going to use it to connect it PSTN line to asterisk box. so my problem is how do i connect my other normal anolog phones(FXS) to asterisk box.Either get an FXS card, or an ATA to turn it into a SIP device.
--Mike Williams--gentoo-user@gentoo.org mailing list
On Thursday 15 June 2006 12:21, Dave S wrote:
Hi all,
Apparently my SIP port 5060 is in use. Is there a Linux command to tell me
what process is using this port. I have a vague memory of seeing such a
command but just cannot remember or find it.
netstat -anp | grep :5060
More useful would
,
Apparently my SIP port 5060 is in use. Is there a Linux command to tell me
what process is using this port. I have a vague memory of seeing such a
command but just cannot remember or find it.
Many thanks in advance
Dave
--
gentoo-user@gentoo.org mailing list
--
gentoo-user@gentoo.org mailing
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Dave S wrote:
Hi all,
Apparently my SIP port 5060 is in use. Is there a Linux command to tell me
what process is using this port. I have a vague memory of seeing such a
command but just cannot remember or find it.
Many thanks in advance
Dave
in a small, inexpensive
linux-centric pbx(voip) solution, in addition to the
fax line, which technologies (hardware and software)
would you recommend? [sip, asterick.]
The max number of phone lines is (6).
thoughts and suggestions are most welcome,
James
--
gentoo-user@gentoo.org mailing list
and sip, portage is happy again. I still wonder what exactly the
error was, but not too much.
Wonko
are built as an SoC or, to be
more precise, as a System-in-a-Package (SiP).
signature.asc
Description: OpenPGP digital signature
Neil Bothwick neil at digimed.co.uk writes:
That sounds a good guess, what does the Xorg.log show?
Well KDE4 is working fine now.
dev-python/sip was the key. Somehow none of the updates(syncs) actually
updated it to the latest stable version (4.8.2) So
dev-python/PyQt4 and hence some
.
The Rhine is a fountain of youth: one sip and you won’t get old.
pgpKIr65FTIj3.pgp
Description: PGP signature
possible → how long it’s still possible
When people you need to communicate with on MSWindows boxes only know how to
manage Skype-ware and you don't run a SIP proxy server yourself, your choices
reduce somewhat.
--
Regards,
Mick
signature.asc
Description: This is a digitally signed message
would be a multi-part blog post.
My recommendation would be to use sip-proxy if it meets your needs, Asterisk
or Kamailio if you need something heavier.
--
:wq
signature.asc
Description: This is a digitally signed message part.
to be able to get to
call. (I think I'm not logged into the directory or something)
In ekiga i put - sip:[EMAIL PROTECTED] (then I press connect) to the
windows netmeeting. Doesn't work.
Under Netmeeting, I put in sip:[EMAIL PROTECTED] and it give me error etc.
Seems like no matter how, I can't get
On Mon, Jan 11, 2010 at 2:00 PM, Stroller
strol...@stellar.eclipse.co.uk wrote:
On 11 Jan 2010, at 18:31, Mark Knecht wrote:
...
MagicJack is just a USB FXS and a SIP client, except that it's tied into
a
proprietary service. There are lots of companies that sell VoIP using the
SIP standard
-packages --sipdir=/usr/share/sip
--assume-shared --no-timestamp --qsci-api --enable=QtCore
--enable=QtNetwork --enable=QtScript --enable=QtXml --enable=QtGui
--enable=QtDesigner --enable=QtScriptTools --enable=QtTest
--enable=QtDBus --enable=QtDeclarative --enable=QtOpenGL --enable=phonon
--enable=QtSql
/lib/calibre/calibre/gui2/progress_indicator/__init__.py,
line 15, in module pi_error) RuntimeError: Failed to load the
Progress Indicator plugin: the sip module implements API v9.0 to
v9.1 but the progress_indicator module requires API v8.1
Thank you Greetings Silvio
You probably updated
what I do I can't
get Ekiga's echoing system to echo back my microphone
(sip:[EMAIL PROTECTED]) ... It's frustrating at this point in time, but I
hope this is as easily solvable as the webcam issue.
Make sure your microphone is unmuted (alsamixer). If applicable, make
sure that your
playback
through my speakers (using alsamixer) but no matter what I do I can't
get Ekiga's echoing system to echo back my microphone
(sip:[EMAIL PROTECTED]) ... It's frustrating at this point in time, but I
hope this is as easily solvable as the webcam issue.
Make sure your
On Tue, 07 Aug 2007 08:11:06 -0400, Colleen Beamer wrote:
On my laptop, I did an update world. One of the programs that the
update wants to install is 'sip'. I connect to my mirror and then the
mirror goes out to www.riverbankcomputing.com, but gets a 404 error.
Anyone else seen this issue
-3
http://www.mepis.org/
1(747)632-4973 SIP
Get Gizmo 1 cent per minuet calling
http://www.gizmoproject.com/
--
gentoo-user@gentoo.org mailing list
I recently switched laptops and although the new one is faster, the
sound is much worse. lspci reports the sound card this way:
ALi Corporation M5451 PCI AC-Link Controller Audio Device (rev 02)
The problem is I need to use a sip phone (sjphone) and I can barely
make out what the other
need to use a sip phone (sjphone) and I can
barely make out what the other person is saying. Is there any
way I can try to improve the sound quality/performance? I am
using the in-kernel alsa drivers. Is it worth a try to emerge
the ones in portage instead or is that unlikely to help
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