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 only use
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
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 speex is
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.
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
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
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
___
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 satisfactory
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 compile
I wonder what would be the easiest way to con
Asterisk into logging all activityon ISDN line? Likeincoming calls,
outgoingetc, 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.
TIA,
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 thread.
No
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. It even
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 mailing
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
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
mapping are forwarded to *. So far I'm unable to get
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
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
___
Asterisk-Users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users
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
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
___
Asterisk-Users mailing list
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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 8.0
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 out
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 good
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 phone (POTS)
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
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 latest/CVS
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
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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
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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