> But now i'm stumbling on another problem.. Asterisk seems to want
> to send the SIP udp packets directly to the SIP clients.
> In the case of a SIP user/client behind a NAT, this obviously doesn't
> work.
Have you tried reinvite=no in your [ser] section of sip.conf?
P
_
> > My Asterisk box also does NAT for internal network, and
> > establishes site-to-site VPN tunnel(s). As a result I have
> > several internal interfaces with private addresses on them, and
> > only one public interface. By trial-and-error I've found out that
>
> This can be a tricky one. If you
My Asterisk box also does NAT for internal network,
and establishes site-to-site VPN tunnel(s). As a result I have several internal
interfaces with private addresses on them, and only one public interface. By
trial-and-error I've found out that FWD (SIP) won't work unless I disable my VPN
tu
> I seem to be having problems with IAX clients based on the iaxClient
> library. I have been working on my own client (an augmentation to the
> Call Manager I released last week) and it seems to regularly miss
> incoming calls entirely. It also occasionally misses the drop signal
Same here. Gen
> The library has several DSP features, including AGC, denoising, and
> echo cancellation. These are all provided via integration with
> preprocessing from the SPEEX library. I don't know if DAN allows you
> to turn on/off echo cancellation or not. However, the echo
> cancellation code from spee
> > I'm interested. I'm running chan_capi 0.3.0 with Fritz PCI ISDN card.
Using
> > DIAX as softphone and dialing out to PSTN generally results in good
sound
> > quality at softphone end (no echo), but PSTN end experiences quite a bit
of
> > echo. I have enabled echosquelch in capi.conf, but it doe
Hi,
I'm interested. I'm running chan_capi 0.3.0 with Fritz PCI ISDN card. Using
DIAX as softphone and dialing out to PSTN generally results in good sound
quality at softphone end (no echo), but PSTN end experiences quite a bit of
echo. I have enabled echosquelch in capi.conf, but it does not seem
What is the status on echo cancellation in
Asterisk/CAPI? I know Zaptel drivers will do echocancel, and chan_capi does have
support for Eicon's echo cancellation, but what about the rest? I found in
mailing list archives a patch description that will mute RX channel whenever
signal level is
About once every day my * goes nuts and "asterisk
-r" responds with "broken pipe". All calls are dropped immediately, even
extension 600 (echo). Killing the process and restarting asterisk helps... until
next day. I'm running 0.5.0 release on RH9. Any ideas what's wrong, and what can
I do to
> > Is it possible to incorporate iLBC codec, some hotels only allow 28.8
> > dial-up links, and then your product will be really useful on the
> > road
>
> How _does_ * work on dialup? I have never tried. I know you have an
> immediate 200-300ms lag but how is it otherwise?
I have very sati
> As promise, the new prerelease (0.9.2) is now available for download from
> the followiing locations:
...
> Please send me your feedback.
Using FQDN instead of IP address would be great! (my Asterisk is on dynamic
IP)
Keep up the good work!
Peter
__
I wonder what would be the easiest way to con
Asterisk into logging all activity on ISDN line? Like incoming calls,
outgoing etc, even if these calls did not originate/terminate at Asterisk
server? I'm using chan_capi if that matters (it should), with Fritz PCI &
S-type ISDN connection.
TI
> I'm testing * (CVS-09/16/03-02:07:49 with zaprtc 0.0.1) with Fritz!PCI
> (chan_capi 0.3.0), and have a couple of funny things - I wonder if anyone
> else has seen them:
Hmm, I'm running plain vanilla * v0.5 and have no problems with that
particular card, same version of chan_capi. Did you compil
> > Well, I happen to be one of those very specific cases... ;) and looks
> > like
> > will have experiment with it myself. Although I'd hate to re-invent
> > the
> > wheel.
>
> Checking e-mail this morning it looks like we have two independent
> "fixes" that both do what has been suggested in this
> http://lists.digium.com/pipermail/asterisk-users/2003-October/024968.html
> >
> > Any idea when these "hacks" will appear in CVS?
>
> We should all hope "never". That's why you call it a "hack"
> because it works for only one very specific case and would break
> SIP under Astrisk for most people
> That's for pointing out Walter Snel "hack".
> Adding his two additional features would not be
> hard.
http://lists.digium.com/pipermail/asterisk-users/2003-October/024968.html
Any idea when these "hacks" will appear in CVS?
Peter
___
Asterisk-Users m
hough SIP can be forced
> to work by slightly breaking it.
>
> roy
>
> On Mon, 2003-10-27 at 10:00, Peter Zeltins wrote:
> > I'm trying to set up * server behind NAT. The box is set up as DMZ in
> > my DSL router, i.e. all incoming connections without explicit port
&
I'm trying to set up * server behind NAT. The box
is set up as DMZ in my DSL router, i.e. all incoming connections without
explicit port mapping are forwarded to *. So far I'm unable to get this setup to
work for either IAX or SIP (tried IAXComm & XLite softphones on public IP
address). Data
Hi all,
I've got my dirty hands on (free!) Vocaltec 4-port FXO/FXS gateway. It is
used unit, I managed to configure correct IP settings in it but am somewhat
at loss how to integrate it into my existing Asterisk network. I have no
H323 gatekeeper, no Vocaltec Network Manager software, and am not f
> Does this thread help?
>
> http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html
>
Thanks, this is exactly what I was looking for. I tried experimenting with
different codecs myself, and GSM seems to be the only one that works...
neither iLBC or Speex went thru. I'm using XLite
I wonder how IAX compares to SIP bandwidth-wise? I've tried both over
overseas IP connection, and somehow SIP seemed to work better.
Peter
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What does this error message mean?
WARNING[262160]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect
process 2 frames
I've been getting these a lot lately, sound quality seems to have suffered.
I'm using I4L driver with Fritz PCI ISDN card. However, even the regular
echo test sounds a bi
> For smaller systems, you'd have to go a NetJet ISDN BRI card ($150? for
two lines)
Try BT Speedway BRI ISDN, ~20$ on ebay
Peter
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Title: Message
All you really should need is:
modprobe hisax type=27 protocol=2
id=isdn0
and in modem.conf:
driver=aopendriver=i4ltype=i4l; ISDN
example;group=1msn=xxxdevice => /dev/ttyI0device
=> /dev/ttyI1
Has anyone got the
BT Speedway (AVM Fritz) card working on a RedHat 8
> My Eicon ISDN card turned up today so - plugged it in and went through
> the modem.conf. It reports unable to open /dev/ttyI0
>
> The problem is I have never used ISDN with Linux - let alone a telephony
> app - and I have no idea even where to start. Some pointers would be
> appreciated.
Check o
> Calling * via SIP produces very good sound. Calling * via the chan_capi
> produces horrible sound. However, if I dial 500 in the demo menu to
> connect to the IAX at digium the sound is good again. ie:
>
> ISDNCall->AVM-B1-Card->Asterisk = All prompts sound horrible
> SIP->Asterisk = Prompts are
> What problems do you have with the chan_capi install?
>
> I am not a hardcore linux guru but it wasn't too hard to setup chan_capi..
Missing capi.h etc. I guess these are installed by CAPI driver, but I had
problems compiling these... for some reason I couldn't just compile a kernel
module beca
> What problems do you have with the chan_capi install?
>
> I am not a hardcore linux guru but it wasn't too hard to setup chan_capi..
Missing capi.h etc. I guess these are installed by CAPI driver, but I had
problems compiling these... for some reason I couldn't just compile a kernel
module beca
> Is it possible to use * as a gateway in the following setup:
>
>LAN (with Windows NT/Linux PCs)
> |
> Ethernet (IP)
> |
> Linux PC with * and AVM Fritz! ISDN Adapter
> |
>ISDN
> |
>Someone with a analog/digital phon
Why wouldn't "SIP show channels" display lag & jitter, it's always 0ms? Is
there a "deeper reason" for this or this is just something not implemented
yet?
Peter
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> Yesterday I updated my pwlib, openh323 and Asterisk from CVS. After making
> "clean opt" in pwlib and openh323 and make "clean install" in Asterisk i
get
> an "Undefined symbol" error when I try to start Asterisk. As far as I can
RTFM. Use specified versions of pwlib & openh323 instead of lates
I've checked everything (pwlib + openh323 + asterisk) out of CVS, compiled,
and chan_h323 module does not load with "undefined symbol
_ZTI19H323AudioCapability". What could be the problem?
Peter
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I'm trying to link up Cisco MGCP-enabled router (residential gateway) with
Asterisk, and it looks like some sort of protocol mismatch, could it be MGCP
0.1 vs 1.0?
Look at this (x.x.x.99 is the router, x.x.x.98 is Asterisk):
MGCP read:
NTFY 2 aaln/[EMAIL PROTECTED] MGCP 0.1
X: 0
O: hd
from 192.
I'll have MGCP hardphone that needs to dial pre-defined number as soon as it
goes off-hook. So far I'm lost as to how (if at all) this can be implemented
in Asterisk. Any pointers?
TIA,
Peter
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I'm still a newbie in Asterisk, just yesterday installed it for home use (so
I can call home while travelling). Using AVM A1 (BT Speedway) ISDN card.
Anyway, I find it very hard to locate supporting information for Asterisk.
User's Handbook is still a draft, this mailing list is not easily
searchab
I can't get my Asterisk to register/place calls with FWD. Here's what I have
in my SIP.CONF:
register => [EMAIL PROTECTED]/1
[fwd]
type=friend
secret=somesecret
host=fwd.pulver.com
username=1
fromuser=1
fromdomain=fwd.pulver.com
I'm using CVS version of Asterisk, checked it out last
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