Displaying 20 results from an estimated 400 matches similar to: "T.38 "pass-thru""
2002 Jul 11
3
Printing from W2K clients
Hi,
I have Slackware 8 Linux Box with Samba-2.2.5 and HP LJ 1200 printer shared by
samba (with LPRng).
The problemm is: when printing from W2K clients users cannot change
print options (like portrait/landscape page orientation, number of
copies etc). When printing from Win98 clients all is ok.
Could someone help vt with this problemm?
--
Sincerely,
Elman Efendiyev
elman@megacom.com.ua
2004 Sep 13
4
Unknown RTP codec 72 received
Hi all,
I get "Unknown RTP codec 72 received" message in console when call in
progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN
over voicepulse connect (IAX) and to FWD echo test (SIP). But this
message only with one SIP client, others (X-Lite too) not giving this
message. All X-Lite settings are identical. Asterisk is last cvs version
This what I see in console
2004 Jul 24
0
PBX functions and different channels grouping
Hi All,
I need to replace old analog PBX with Asteriskl and X-Lise SIP
SoftPhones as client phones.
First: I have problems with implementation of PBX functions. I need and
unsuccesfully tried theese functions (took info at
http://voip-info.org/wiki-Asterisk+PBX+functions)
Call Pickup: Supported in the standard installation (*8 - defined in
res_parking.c +54)
- Just don't understand how to
2004 Jul 12
0
IP Soft Phone with FAX
Hi,
I need to send and receive faxes over VoIP in realtime.
I mean: user ? calls from VoIP network to fax machine on PSTN, but
starts voice conversation with user B on that fax machine. Then users
agree to send a fax (any direction), pressed "start", completed fax
transmission and then continue a voice conversation.
This is one of generic ways to use analog fax machine.
As I understand
2004 Jul 28
0
D-Link DG-104SH H323 problemm
Hi,
I'm using D-Link DG-104SH (H323 4 port FXS gateway) with analig phone
connected to it and X-Lite softphone as endpoints with *
When I calling from X-Lite to analog phone it's ok
When I dilaling X-Lite from analog phone, X-Lite si ringind but whei I
picked up X-Lite connection drops
IP of DG-104SH is 192.168.1.3, H323 ID is GW1
X-Lite number is 233
Here is * output:
-- Executing
2007 Apr 27
1
SIP<->H323 calls without proxying RTP
Hello,
Could somebody tell me is it possible to use asterisk without RTP proxying
in SIP<->H323 calls?
I mean exactly what canreinvite=yes option do in SIP<->SIP calls.
I don't need a transcoding, only a signaling conversion, and this is
possible with some softswitches, so i wondering what about asterisk.
Same question about H323<->H323 calls
I'm using NuFone
2004 Sep 20
2
H.323 call problemm (no sound)
Hi all,
I'm having trouble with H.323 outbound calls, * connects but there is no
sound in both ways.
I'm using X-Lite as SIP client with GSM codec and dialing to ITSP (which
using cisco, I think) over H.323 with G.729 codec. I have 4 digium G.729
licenses installed and this is onli one call.
I tested my * with another ITSP over SIP and G.729 codec and there was
all ok
Here is my configs
2004 Jul 25
1
Busydetect problems
Hi guys.
I have a XP100P Clone , and the busydetect dont work for me..
PSTN---Asterisk---Sip---Asterisk----PBX
Any call from pstn side dont disconnect ... I have no disconnect supervision and busydetect dont work...
Please Help me.
Zapata.conf
[channels]
echocancel=yes
usecallerid=no
hidecallerid=no
rxgain=0.0
txgain=0.0
signalling=fxs_ks
callprogress=no
context=entrada
channel=>1
2004 Sep 17
1
How would you handle a fax without T.38orG.711uLaw?
asterisk-users-bounces@lists.digium.com wrote:
> Isn't it possible to use T.38 for interconnecting hardware gates
> supporting T.38 with asterisk using SIP REINVITE?
> I'm not shure but but think its's might be possible because after
> reinvite traffic goes directly from one gate to anotger, not over
> Asterisk
We've seen a problem here with asterisk. Wehn
2004 Jul 24
1
Please help I fear I have missed something very important! but what?
Sorry about this, I have been struggling with the basics of my asterisk
config.
I set up two sip peers and two phones. And I set up lots of dial masks
for outgoing calls, all my outgoing calls were working great, however
incoming calls were a different matter altogether, I cannot get incoming
calls to work. So I have gone back to a very basic FWD config, with one
phone which as far as I am aware
2007 Feb 11
0
TE110P working hardware configurations
Helo,
I have a troubles getting to stable work of Digium TE110P card (mailed some
time earlier in the list) - I can't get 100% pseudo zap interface accuracy
(zttest), so getting HDLC aborts and call drops. I tried number
motherboards, hardware and software configs according to info in wiki, thisl
list and number of websites - no luck.
So I ask everyboby who successfully use Digium TE110P card
2007 May 11
0
Asterisk crashes
Hello,
I have very annoying problem with asterisk 1.4.4:
Every evening when I have peak load asterisk crashes, "peak load" is only
over 20-30 sip-to-h323 simultaneous calls. Nothing special in logs after
crash. Load average never was higher than 0.3, asterisk never uses more than
12% CPU (according to top). Tried SVN versions - same result. Both h323 and
sip peers has only one codec
2004 Nov 23
1
Fax over SIP Problems (sorry for this topic ...)
Hello everyone!
I tried to send a fax over SIP with an Asterisk Server in the middle (no
Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway
is external). Whenever I start sending a Fax to a PSTN destination, the
Call gets answered and asterisk tries to build a native bridging:
-- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and
SIP/xxx-3ef8
Then the following
2007 Feb 12
3
Trixbox vs. Custom install
Hello,
I'm following the thread "Asterisk@Now vs Trixbox", and I have a
similar question: if someone is going to install Asterisk, FreePBX
and A2Billing, should you advice him/her to use Trixbox ... or a
custom "step by step" installation on a distribution of his/her choice?
Thanks
Stefano
2007 Feb 07
9
Digium TE110P
Helo,
I have problem with Digium TE110P connected to CISCO 3640 (port on
NM-HDV-2E1-60) wth PRI E1 link. I use CISCO now for testing but when I
tried with real PBX problem was exactly same.
I have this messages in Asterisk conole and log sometimes:
NOTICE[1115] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel
of span 1
Usually 2-5 such messages in series, can be repeated after 10
2005 Dec 09
5
Memory overcommit
I have been using Xen on a daily basis on a production (but not
critical) machine for a number of months now. It''s looking really good.
One thing that I have not yet seen anyone mention as a feature that I
would really like to see is the ability to overcommit memory. I have 2G
of RAM in my machine. I would like to give a developer his own virtual
domain to sandbox his application
2004 Dec 21
7
Cannot transfer with Cisco or Snom
I am having a hell of a time with transfers.
First the Snom issues:
The transfer button on the Snom 220 does not work. I have read about
setting break key off in the advanced page of the web config but the Snom
220 has no such option. At the moment I am having to use the # transfer
hack which makes this phone look really stupid to have buttons on it that
cannot be used. Anyone know how to
2004 Apr 21
1
sip 4 fedora
Good day all
I'm still looking for a SIP client that will work on fedora core 1?
Thanks
2007 Feb 16
7
Summary of "Trixbox vs. custom install"
Hello everybody. First of all thanks to all the people giving their
opinion on the subject I proposed: "Trixbox vs. custom install".
You've all been very helpful.
I try to summarize what has emerged from the various messages.
Forgive me if I miss or forget something or if I simplify too much
some of your messages...
- Elman Efendiyev says that you should install from sources if
2004 May 13
4
BGM Music
Is there any way to play background music on a sip phone
while the phone is not in use like many legacy pbx's offer?
Could you take 7960 and use the 6th line in a similar fashion
to the all setup maybe?
Thoughts ideas?
--
respectfully, Joseph - (606) 477-2355 x140
------=============