Hi ,
Does anyone encounter this problem ? We have installed Asterisk at Site
A and have 128k Frame Relay over to Site B.
We are using Zyxel 2 port FXS at Site B and Linksys PAP2-NA at Site A.
We are using Ulaw at Site A and G729 at Site B.
When the calls are originated from Site A to Site B,
Doug Lytle wrote:
[EMAIL PROTECTED] wrote:
On Mon, 3 Oct 2005, Corey S. McFadden wrote:
Am I just using the Set() command wrong? It seems pretty
counter-intuitive not to enclose multi-word strings in quotes but if
that's the problem let me know.
Yeah, that's the problem.
Hey guys.
I have to put my * behind a Firewall through nat on the firewall.
The asterisk is running, but for example a register to an outside PSTN
provider won't work.
I enabled nat for the register but i only get Code 120 Send request.
The other problem is, when i try to register with a sip
I have a problem. Incoming calls work without problem
but I cant call out. Using AAH.Gets a busy tone
Anyone who can see a mistake in Outgoing settings
context=from-pstn
host=ipkund1.rixtelecom.se
insecure=very
nat=yes
secret=xxx
type=peer
username=0406082250
Regards
Hi,
Outgoing setting is in zapata.conf. I think you should read the wiki
more ;).
If what you mean by outgoing is another sip extension then you should
look for extension.conf.
Links:
http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
Dear Matt,
Thanks for your great work and the effort documenting the whole process.
I'm sure the whole Asterisk community benefits from this kind of work
and it's really something to end up in the wiki.
Thumbs up!
Best regards,
Vahan
Matt Roth wrote:
List members,
My previous post
On Tuesday 04 October 2005 00:42, Rajesh kumar wrote:
I am using Kphone which works great for my purposes! You can look at
twinklephone as well at http://www.twinklephone.com/
Hi, thanks all for the info, kphone does really wierd stuff and I can't get
twinkle to compile. I'm looking into that
do you think it would make any difference to change the process-priority
if zttest is the only running process except ssh-daemon and the
login-shells ?
[EMAIL PROTECTED] wrote on 30.09.2005 18:11:47:
Are you starting Asterisk with the -p option (high priority?)
Also, do you get a different
René Enskat [Teamware GmbH] wrote:
Hey guys.
I have to put my * behind a Firewall through nat on the firewall.
The asterisk is running, but for example a register to an outside PSTN
provider won't work.
I enabled nat for the register but i only get Code 120 Send request.
The other problem
Anders Svensson wrote:
I have a problem. Incoming calls work without problem but I cant call
out. Using AAH.Gets a busy tone
Anyone who can see a mistake in Outgoing settings
context=from-pstn
host=ipkund1.rixtelecom.se
insecure=very
nat=yes
secret=xxx
type=peer
Hi Giordano,
pls. check the following things:
- edit your /etc/capi.conf (or /etc/isdn/capi.conf) and
adjust the settings according to your card(s)
- call "capiinit"
- check the status by calling "capiinfo". The output should
show the details of your card(s)
- if you're running asterisk as
stevanus wrote:
Hi,
Outgoing setting is in zapata.conf. I think you should read the wiki
more ;).
If what you mean by outgoing is another sip extension then you should
look for extension.conf.
Links:
http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
In article [EMAIL PROTECTED],
Richard Cook [EMAIL PROTECTED] wrote:
I thought maybe someone was using 0.63 with code they developed themselves.
The TDS API has been un-published in 0.63: one is expected only to use
dblib or ctlib, neither of which has any support in Asterisk. For 0.63
you have
I'm finding that I'm a bit disappointed that Asterisk doesn't naturally
forward the Remote-Party-ID from inbound SIP calls (where
trustedrpid=yes) to outbound SIP calls. I guess this is going to be
something we have to use SER for, unless we make our own custom build
(which I'm reluctant to
Hi!
Is there a simple way for an * newbie to force * to
use different sip-trunks for different calls. I have 2 siptrunks, one for
inland calls and one for international calls. All in country numbers starts
with 0 and all international starts with 00. This I have configured in the
You've not said much about your firewall setup. I presume you've opened
up 5060 and RTP ports?
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Asterisk-Users@lists.digium.com
Hi,
Yes i opened 5060 and range -20001
The firewall is not blocking.
I tried to set the externip and localnet but can't register to the pstn
gateway and can't onnect with my nat phones.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag
von
me too looking for softphone...not able to enable kphone
Can anyone please highlight more on it.
ThX
/Gurmi
On 10/4/05, Wayne Gemmell [EMAIL PROTECTED] wrote:
On Tuesday 04 October 2005 00:42, Rajesh kumar wrote:
I am using Kphone which works great for my purposes! You can look at
Alex Lake wrote:
I'm finding that I'm a bit disappointed that Asterisk doesn't naturally
forward the Remote-Party-ID from inbound SIP calls (where
trustedrpid=yes) to outbound SIP calls. I guess this is going to be
something we have to use SER for, unless we make our own custom build
(which
Anders Svensson wrote:
Hi!
Is there a simple way for an * newbie to force * to use different
sip-trunks for different calls. I have 2 siptrunks, one for inland calls
and one for international calls. All in country numbers starts with 0
and all international starts with 00. This I have
Is there anyone who is currently using Asterisk as a production H323
gateway?
And using which combination of asterisk and H323 (chan_h323, chan_oh323?)
The main issue is interoperability with other H323 parties (Cisco AS53xx,
Nextone, etc).
Searching the mailing list it seems that both h323 and
On 10/3/05, Morten Isaksen [EMAIL PROTECTED] wrote:
On 10/3/05, Olle E. Johansson [EMAIL PROTECTED]
wrote:
Does anyone know what stale nonce is?I've answered this question many times, so you should be able to find
the answer...A stale nonce is when a device tries to re-authenticate with a
We have parlay i60 LCR connected to octoBRI card on i60 bri ports.
When i call a number on asterisk and call is connected to i60 over BRI
ports and call goes to PRI line callerid is presented without last 2
digits.
greetings
Milos
___
--Bandwidth and
What is the version of Diax ?
regards
Pierre
2005/10/4, amna saleem [EMAIL PROTECTED]:
Hi!
I am facing some problems with my asterisk-1.0.3.I am using DIAX phones as
clients ,but sometimes they donot register with the asterisk server.Also if
I don`t restart my asterisk frequently the
Found out what's wrong, maybe it can help others.. my linux's package manager
automatically pulled the most fresh version of asterisk, 1.2beta. Downgrading to
1.0.8 solved all problems described below, I get excellent quality and no noise now.
cheers,
Kristof
Kristof Jozsa wrote:
Hi all,
On Mon, 03 Oct 2005 22:09:23 -0600, Stephen Bosch wrote:
Hi, everyone:
I'm in the processing of deciding what IP phones we should use with our
Asterisk server, and I wanted to get feedback from the user community on
the quality, reliability and ease of operation of Snom phones.
What do you have
I have been getting quite a bit of PRI Resets using my Quad PRI Digium
card.
Prior to the resets I am getting similar notices to the following
chan_zap.c:7482 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary
D-channel of span 3
Telco claims the PRI's are fine on their end and that it
You can't put four span in timing, because only one must be like nmaster
sincronization. If one of your telco provide time for your card. Put second
value in all span to 0.
regards,
srsergio
-Mensaje original-
De: Ronald Hartmann [mailto:[EMAIL PROTECTED]
Enviado el: martes, 04 de
Hi guys,
Does anyone know of a way where I can bring a third person in on my
conversation. Say I'm on a IAX or SIP call from a softphone DIAX or IAXCOMM
and am speaking to someone now I want to quickly bring another SIP or IAX
extension into this call so the three of us can speak to each other.
Hello
Does anyone have an example of how to use the MONITOR command from an
AGI-script ?
I have tried different methods, but none of them worked :-(
I'm using Python
MIR
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Asterisk-Users
On Tue, 04 Oct 2005 14:28:47 +0800, Stephen wrote:
Hi ,
Does anyone encounter this problem ? We have installed Asterisk at Site
A and have 128k Frame Relay over to Site B.
We are using Zyxel 2 port FXS at Site B and Linksys PAP2-NA at Site A.
We are using Ulaw at Site A and G729 at Site B.
On Tuesday 04 October 2005 08:32, Ronald Hartmann wrote:
I have been getting quite a bit of PRI Resets using my Quad PRI Digium
card.
You've got a problematic setup for Digium's Zaptel cards.
You're also running an old version of Asterisk. 1.09 is the stable release
and 1.2 is the upcoming
On Tuesday 04 Oct 2005 05:17, [EMAIL PROTECTED] wrote:
On Mon, 3 Oct 2005, Aryanto Rachmad wrote:
I sent an email to Digium support and got only a reply like this:
Although the card is being shown as an 'Unknown Device', it should still
work properly.
To be honest, I am not happy with
Patrick Friedel wrote:
Rich Adamson wrote:
My office has been running Asterisk 1.0.8 and a TDM04B for a few
months now without too much trouble. After a while we discovered
that after a certain period (about a month), asterisk stopped
acknowledging inbound calls. Since I was out of the
On 3 Oct 2005, at 22:54, Matt Roth wrote:
List members,
It has been a while, but I once implemented a simple shared database
over NFS, so
dredging my memory produced the following:
Future Plans and Unresolved Issues
==
I wrote Windows software for
I have just realized while trying to research asterisk not acking
incoming calls
that the tdm400b card is stamped rev H, but when I issue the zap show status
command in the manager interface, it indicates Wildcard TDM400P REV E/F
Board 1
Which do I believe??
--
Best Regards
Greg Cirino
Hello,
Would like to use IAX /IAX2 to transport 30 channels inter Asterisk.
I don't have any experience with that, so can someone help ??
How much bw do I need for simultaneous calls and is there any latency for SIP
G711 to IAX2 and vice-versa , ... etc ?
Thanks in advance for any info,
Geo
Bob Goddard wrote:
From the pci.ids file. Digium should email the details to Martin.
We are well aware of that. A quick scan of that file will show that we
already have the IDs for the dual/quad-span cards in the master database.
The issue with the TDM400P and the single-span cards is that
Currently we are working with Telco Providers to provide 911 and e911 with
all the bells and whistles, including CNMAN features. This will enable you
to deliver 911 calls to PSAP with out having to tell them your location. Get
ready to manage DB ... check out REDSKY software
-Original
Cirelle Enterprises wrote:
I have just realized while trying to research asterisk not acking
incoming calls
that the tdm400b card is stamped rev H, but when I issue the zap show
status
command in the manager interface, it indicates Wildcard TDM400P REV E/F
Board 1
Please contact Digium
I'm trying to screen the call transfert to my cell phone using a exemple found on the web.
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
It work partially: while I'm prompted to accept the the caller still
her the music. But wether I accept the call or reject it I'm put in
communication with
Hi,
I am running asterisk on Fedora Core 3, Configured few extension, I receive
frequent error message on * console as
-- Got SIP response 481 Call Leg/Transaction Does Not Exist back from
xxx.xxx.xxx.xxx.
This only comes from two extensions which I configured
Any idea what does this error
* answers the call, but if the incoming caller hangs up, * does not
release the line.
Is there a polarity reversal on hangup (those clicks you hear maybe)? If so
then you may find that using the CVS-HEAD version of Asterisk will help
hugely. Put hanguponpolarityswitch=yes in your zapata.conf
Right at the end of your Zapata.conf you have:
#include zapata_additional.conf
hanguponpolarityswitch
;Include genzaptelconf configs
#include zapata-auto.conf
Remove that hanuponpolarityswitch as you already have
hanguponpolarityswitch=yes earlier on, and I don't know what having the
second one,
I've been trying to work with the dynamic feature support.. IE adding
codes like *2 to features.conf that can trigger a dialplan
application to run.
I've been unable to get goto to work properly. AGI also seems to
not function correctly if called as a feature.
Anyone else playing around
Kevin P. Fleming wrote:
Cirelle Enterprises wrote:
I have just realized while trying to research asterisk not acking
incoming calls
that the tdm400b card is stamped rev H, but when I issue the zap show
status
command in the manager interface, it indicates Wildcard TDM400P REV
E/F Board 1
I guess you could post your config files here and hope that someone
feels inclined to look them over! ;-)
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Asterisk-Users@lists.digium.com
Can anyone tell me if there is a Calling Card Platform in which I can use
in conjuction with Asterisk that can give me Authentication via the caller
id of the user. I don't want a PIN based Calling Card system, but the
software to be able to recognize the caller ID information and
authenticate the
On Tue, 2005-10-04 at 08:35 -0500, [EMAIL PROTECTED] wrote:
Can anyone tell me if there is a Calling Card Platform in which I can use
in conjuction with Asterisk that can give me Authentication via the caller
id of the user. I don't want a PIN based Calling Card system, but the
software to be
Hi!
Where can a newbie find some info about how to set up
an auto attendant extension?
Regards
Anders Svensson
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I have just installed spandsp-0.0.2 onto my Asterisk 1.0.9
I am using an old Intel 536EP (actually found drivers that work)
BUT...all my faxes are coming in landscape mode
Has anyone come across this?
any fixes?
Shawn
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--Bandwidth and Colocation
Hello.
It's possible to use IODBC instead UNIXODBC with realtime?
As I see, the res Makefile load a odbcinst.h file, but
in IODBC there's not this file.
I change the res Makefile (iodbcinst.h instead odbcinst.h)
and the make create the res_odbc.so.
But when asterisk boot it don't start showing:
I have been battling echo since we installed a new system at one of
our clients. I am using a single span digium card. I believe this is
the first time someone has setup a PRI in this area (its way out in
the middle of nowhere). We get slight echo on all calls, and when
calling some numbers (long
1) What do these two notices mean?
Oct 4 09:34:30 NOTICE[49584]: chan_iax2.c:5777
socket_read: Rejected connect attempt from
198.22.67.70, request '[EMAIL PROTECTED]' does not
exist
Oct 4 09:34:51 NOTICE[49584]: chan_iax2.c:5777
socket_read: Rejected connect attempt from
66.234.228.170,
Hello,
How would I setup where I call into my number and
press say 911 and then it would ask for a pass and
would accept it and then would prompt for a number so
I could call out of my number on the road?
Joshua
__
Yahoo! Mail - PC Magazine
On 10/3/05, Matt Roth [EMAIL PROTECTED] wrote:
Before writing these scripts, I have two questions that need answered:1) How can I tell when a file is complete on the NFS server?
We do something similar with a perl script, it grabs the file sizes of
the 2 legs of the recording, then waits 5
I'm using a SNOM 360 with Ver 4.3 software.
Asterisk is Asterisk CVS-D2005.05.02.22.00.00-05/04/05 (BRI Stuff +
Head)
I've used the wiki info to set up some lines to monitor some internal
extensions.
When the extension is rung - the lamp comes on, when the call is
answered, the lamp goes
On Tuesday 04 October 2005 10:37, Tad Heckaman wrote:
the middle of nowhere). We get slight echo on all calls, and when
calling some numbers (long distance calls but still in the local
area), we get very loud echo. The person calling out can hear their
own voice at the same volume about a half
Hey Guys,
I have a new task to tackle. I need to make asterisk save me as much $$$ From
Ma Bell As possible. Here is the Scenario. I have 50 storefronts throughout the
state of Michigan.
Each location has 2 voice lines in a hunt group and a FAX Line with DSL on it.
I also have a large
Hello,
I have a block of 25 DIDs and have 10 phones on the
network. I want when a person tries to call out for *
to pick a number for the CIDN and I want to make sure
that the number isn't duplicated while it's in use.
Joshua
__
Yahoo! Mail - PC
Add direct-inward-dial to your dial peer and it should work fine.
-Greg
On Mon, 2005-10-03 at 15:48 -0700, Tim Pozar wrote:
I would think I could do this but for some reason I am stymied.
I have a PRI from RCN connected to a cisco 3640 (in my day cisco is
all lower case :-)). My config
zaptel Disabled echo canceller because of tone (rx) on channel 16
I just did dmesg -c and thats what I got... I think there was a call
allready in progress. Whats that about?
On 10/4/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Tuesday 04 October 2005 10:37, Tad Heckaman wrote:
the middle
On Tue, 2005-10-04 at 07:44 -0700, Crystal Stream, Incorporated wrote:
Hello,
How would I setup where I call into my number and
press say 911 and then it would ask for a pass and
would accept it and then would prompt for a number so
I could call out of my number on the road?
Joshua
show
After doing a quick search, it appears that maybe a bellsouth echo can
is turning my echo can off? How do I tell zaptel to leave it on,
regardless if it recieves that tone?
On 10/4/05, Tad Heckaman [EMAIL PROTECTED] wrote:
zaptel Disabled echo canceller because of tone (rx) on channel 16
I just
Announcing Voice over IP Directory Services
(http://www.voipDS.org)
Hi All,
I am sure many of you are aware that by using VOIP
devices one can make peer-to-peer calls. By devices I
mean software phones, hardware phones and Asterisk.
This feature is available, out of the box, in
majority of
I would check your /etc/ld.so.conf file and make sure that you have the
library path for the IODBC libraries in there...
Then run ldconfig
and try reloading asterisk again.
Hello.
It's possible to use IODBC instead UNIXODBC with realtime?
As I see, the res Makefile load a odbcinst.h file,
On Tuesday 04 October 2005 10:59, Tad Heckaman wrote:
zaptel Disabled echo canceller because of tone (rx) on channel 16
I just did dmesg -c and thats what I got... I think there was a call
allready in progress. Whats that about?
Was that after the first or second dmesg -c?
Procedure:
dmesg
Can someone explain what is meant by CVS Head?
I see references to it a lot, but don't know what it means!
I am having a problem with *, and it has been suggested that installing CVS
Head might help, but I don't what it is, or where to get it!
TIA
Leigh
Upgrade Asterisk. Versions of HEAD post 8-29-05 have this functionality built in.
On 10/4/05, Mark Elkins [EMAIL PROTECTED] wrote:
I'm using a SNOM 360 with Ver 4.3 software.Asterisk is Asterisk CVS-D2005.05.02.22.00.00-05/04/05(BRI Stuff +
Head)I've used the wiki info to set up some lines to
How would I setup where I call into my number and
press say 911 and then it would ask for a pass and
would accept it and then would prompt for a number so
I could call out of my number on the road?
How about in whatever context you define your inbound number, add this exten:
exten =
Hi,
Has anyone seen this before? The phones are
registered OK, and they can take incoming
calls, but all I get is a fast busy whatever
I dial. I've tried regular numbers, *98, etc.
Looking at the Asterisk Command Line Interface, I
don't see any text outputted when I try to dial out.
I wonder
You know what.. I have sporadic echo issues too and I just checked my
dmesg and also see that! What's this all about?
zaptel Disabled echo canceller because of tone (rx) on channel 3
zaptel Disabled echo canceller because of tone (rx) on channel 2
zaptel Disabled echo canceller because of tone
On Tuesday 04 October 2005 11:02, Tad Heckaman wrote:
After doing a quick search, it appears that maybe a bellsouth echo can
is turning my echo can off? How do I tell zaptel to leave it on,
regardless if it recieves that tone?
The tone to disable echo cancellers is a good thing, not a bad one.
Balaji NJL wrote:
Announcing – Voice over IP Directory Services
(http://www.voipDS.org)
To make this global, where any VOIP user could make
peer-to-peer call to any other VOIP user, we need the
following
a central repository which stores peer connection
information of all users
an
I'm trying to get
ast_rxfax and ast_txfax compiling with Asterisk 1.2.1 beta. The two
ast_?xfax files don't compile:
gcc -pipe
-Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g
-Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6
-march=i686
-fomit-frame-pointer
I just ran the command once. I just called one, and I heard myself in
the background, but I did not get that message in dmesg. Still, that
message about it being disabled worries me because I get really bad
echo on SOME calls. I got that disabled echo cancel message 4 times in
my dmesg, but I am
Can someone explain what is meant by CVS Head?
Leigh,
CVS Head is the latest-and-greatest development version of Asterisk.
CVS stands for Concurrent Versioning System and is the system used by
the developers to track and coordinate changes in the programming.
You can obtain the CVS Head
BJ Weschke wrote:
Upgrade Asterisk. Versions of HEAD post 8-29-05 have this functionality
built in.
Some of it is currently broken, but there is a patch in the bug tracker
that fixes status notification for Eye-beam. haven't tried with Snom.
/O
___
Trying to compile BRIstuff 0.2.0-RC8o. Ran the download.sh and
compile.sh scripts to automate the process.
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
Well, I dont receive faxes anyway, it goes to a POTS line. I also
disabled it in the zconfig.h file and recompiled, but I haven't
installed it yet.
So going back to my original email... Anything else that might be
causing my issues?
On 10/4/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On
Hi:
I have one TDM40B and one TDM04B on my Asterisk box.
Both were working fine. Then, all of the FXS ports
started to make echo sounds when I make FXS to IAX or
SIP connection. All of the FXS ports fail to make
bridging to the FXO channels. And when I try to make a
call from the console to the
On Tuesday 04 October 2005 11:26, Matt wrote:
You know what.. I have sporadic echo issues too and I just checked my
dmesg and also see that! What's this all about?
*STOP*
You will receive these messages if you send or receive faxes. I asked for
this particular procedure to be executed
Since Colin Anderson -- in a previous thread -- asked the question about
whether ADSI was dead, I thought it was worth discussing.
We've been using Nortel Vista 350s in our office up until now. The
phones are from Telus, I don't know if there's any way to unlock them.
It would appear Telus hasn't
Thank you!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nathan Pralle
Sent: 04 October 2005 11:43
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CVS Head - Obtaining Newbie Question
Can someone explain
Cirelle Enterprises wrote:
we also experienced this with asterisk 1.0.9 and rev H of the tdm with
4 fxo modules
we were restarting asterisk every night via cron and this still happened
in our case, 3 out of 4 fxo modules (2,3,4) crapped out and stopped
ack'ing incoming
calls (outgoing
Is there a way to slow down or speed up the speed at which SayDigits
rattles off a series of digits?
Thanks,
Nathan
--
-
Nathan E. Pralle
Give the Director a Serpent Deflector
www.nathanpralle.com
-
On Tuesday 04 October 2005 11:54, Tad Heckaman wrote:
Well, I dont receive faxes anyway, it goes to a POTS line. I also
disabled it in the zconfig.h file and recompiled, but I haven't
installed it yet.
I thought this was a TE110P, and not a TDM4xx or X101P?
So going back to my original
Hello,
How do I make sure the G.729 codec is being utilized
fully and not just as a passthru? I've registered it
and followed the install instructions
__
Yahoo! Mail - PC Magazine Editors' Choice 2005
http://mail.yahoo.com
--- Olle E. Johansson [EMAIL PROTECTED] wrote:
Balaji NJL wrote:
Announcing Voice over IP Directory Services
(http://www.voipDS.org)
Sorry to be negative, but this kind of services came
up in tons
when e-mail was a new service. All of the addresses
registred in there
is now
Hi, list!
I'm playing on an [EMAIL PROTECTED] installation, since a month or two.
I've had no trouble setting it up 'n running.
I've bought a couple of DLINK's DPH-140S SIP Phones, to use with Asterisk.
From this phones, I can make receive calls with no trouble, but, when I
try to use some
On Tuesday 04 October 2005 06:09, Stephen Bosch wrote:
Hi, everyone:
I'm in the processing of deciding what IP phones we should use with our
Asterisk server, and I wanted to get feedback from the user community on
the quality, reliability and ease of operation of Snom phones.
What do you
Tim,
Thank you for the information. I will keep it in mind when implementing
my mixing and archiving system and share the results with the list when
it is complete.
Thanks,
Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
tim panton wrote:
On 3 Oct
Hi Shawn,
Could you explain what you mean by 'orientation'. Are your faxes rotated 90
degrees?, are they compressed in the longitudinal plane?
Send me one of your landscaped tiff files offlist and I'll try to see whart
is going on.
Craig
- Original Message -
From: Shawn Porter
on asterisk command line do a
show translations
-bill
On 4-Oct-05, at 12:29 PM, Crystal Stream, Incorporated wrote:
Hello,
How do I make sure the G.729 codec is being utilized
fully and not just as a passthru? I've registered it
and followed the install instructions
On Monday 03 October 2005 13:50, Paul Dugas wrote:
This is a wierd one. Can't figure it out. I have an SPA-3000 at the
house handling my incoming line. It's setup to direct the incoming call
to asterisk. Works great 99% of the time.
A few times a day, I'm getting calls that ring once
Title: RE: [Asterisk-Users] DPH-140S SIP Phone oddities
Change your DTMF setting to rfc2833. You may be using an incompatible type with your phones.
..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
from the CLI, show g729 -- In my situation, with polycom 501s, if a
phone calls another internal phone and canreinvite is set to yes, this
does not count against your licenses 'cause the phones are now the only
devices in the conversation. you can still find in my phones' status
menu what
Correct, this is a TE110P. I disabled the echocan disable thing in the
config file, but I havent actually installed it since it is in
production.
Is there someway to make a backup of the modules before I reinstall
zaptel? I want to easily jump back to the point before I changed some
of the
On 10/4/05, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
from the CLI, show g729 -- In my situation, with polycom 501s, if a
phone calls another internal phone and canreinvite is set to yes, this
does not count against your licenses 'cause the phones are now the only
devices in the
I managed to compile
app_rxfax and app_txfax against the latest asterisk (1.2 beta 1). When
trying to load the app_rxfax module I get this error:
[app_rxfax.so]Oct 4 12:52:25
WARNING[3701]: loader.c:314 __load_resource:
/usr/lib/asterisk/modules/app_rxfax.so: undefined symbol:
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