On Thu, Sep 08, 2005 at 06:42:49PM -0500, Chris Shipman wrote:
I started working on a program using Ghostscript and Redmon to generate
the tif in windows by a printer.
So far I am using FTP to transfer the tiff and call file. At least until I
figure something better out.
Why don't you
Hi
I discovered that most onboard raid controllers are really software
raid, and it uses the cpu to perform raid functions.
I am not sure how much extra load this introduces, but anyway, its
still not ideal when you need your cpu for transcoding voip stuff.
my 2c.
regards
Clive
On 8 Sep
On Thu, 8 Sep 2005, Matthew Boehm wrote:
Jason Becker wrote:
Hmm, looks like someone in the know needs to update the wiki:
http://www.voip-info.org/tiki-index.php?page=Digium+DS3000P
Wow. Guess I'm not.
I've got a 4 port PRI card in this brand new Dell 1850 3.0Ghz Xeon
On Thu, 8 Sep 2005, Matthew Boehm wrote:
Carlos Antunes wrote:
Have you seen this?
http://www.digium.com/index.php?menu=compatibility
Yes, but I'm not using a Digium card.
Ah - so the difference between your setup and mine is that you are using
Sangoma (presumably) and I'm using
hello everybody,
i used asterisk with 2 ISDN BRI AVM cards in paralel with a panasonic ISDN pbx for testing putpose.
is this also possible to use E1PRI to use in parallel with a simence PRI pbx for test purpose?
|---asterisk
public line|
|---PBX
best regards
shaon
On Thu, Sep 08, 2005 at 04:49:49PM -0400, David Hajek wrote:
- What is the sugested codec for such setup? Now I'm using ULAW, but
realizing it may not be the best choice. Sometimes I can hear broken
audio. Maybe speex is better choice?
Also consider iLBC . gsm consumes less CPU than either
On Fri, Sep 09, 2005 at 08:05:09AM +0200, Clive wrote:
Hi
I discovered that most onboard raid controllers are really software
raid, and it uses the cpu to perform raid functions.
Also: in such a settings you can get comperable performance by using
Linux's built-in software raid. And for
Hi,
I sucked the TE410 in a Siemens dual Xeon machine... lot of irq
problems, digium support said: try the card in another machine.
A cheap amd64 + via K8T800 and TE405 works perfectly...
On Fri, 2005-09-09 at 08:05 +0200, Clive wrote:
Hi
I discovered that most onboard raid controllers are
Hi,
I had written a web application for queue report. In that I had calculate the incoming calls through parse the queue_log and the return info from management API.
But for the realtime refresh about the current status, it is a little affectionto voice quality of our call center system, I want
Canuck15,
No, I hadn't played with the gains. But I've now done so and no difference
unfortunately. Thanks for the suggestion though.
I have discovered that after Asterisk has answered the call and the remote
caller has hung up, if I lift the receiver on a phone connected to the
line
(in
Chris Stenton wrote:
With todays CVS head I am getting the following being sent after a call
has been terminated
on my Cisco 7960. It eventually gives up with a critical error.
chan_sip.c:1132 retrans_pkt: Maximum retries exceeded on transmission
[EMAIL PROTECTED] for seqno 102
(Critical
HI Chris
I am interested, I would like to know how I can have the opportunity to test your program.
On 9/9/05, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Sep 08, 2005 at 06:42:49PM -0500, Chris Shipman wrote: I started working on a program using Ghostscript and Redmon to generate the tif in
This may also cause a hanging SIP channel. You can check it by issuing 'sip
show channels' in CLI.
CCF
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Olle E.
Johansson
Sent: Friday, September 09, 2005 16:52
To: Chris Stenton
Cc: Asterisk Users Mailing
Yes, you missed something:
4 PRIs = 92 Lines per Card * 4 Cards = 368 Lines
Isn't that just in North America? I believe most of the world uses
E1 PRIs with 30 lines per PRI.
right, we are in italy here, 1 PRI == 30 lines (calls)
i guess may be it's a 64bit variable. so you can only use 0-63.
CCF
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of René
Mayorga
Sent: Wednesday, September 07, 2005 15:56
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Hosted PBX (vPBX) and
I have test 3.0GHz systems - Intel Desktop board.
I've been testing with a TE405P with looped ports - 1 to 3, 2 to 4. My
test is 20 second long calls with one side playing music on hold, the
other playing gsm prompts. All channels full (60 calls out, 60 in).
Niiice, can I ask what
Hi all,
I am new on this list an I hope the posting is correct.
I am using Debian Sarge, kernel 2.6.13 from www.kernel.org and want to
install the drivers for an AVM Fritz!PCI v2.0 ISDN card.
I used the directions from AVM but Asterisk allways stops with CAPI not
installed!
I am new to
Andres wrote:
8.1.50;tag=as12a1c927
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
As you can see. This is just a NOTIFY message. Probably a Keep Alive.
User-Agent: Asterisk PBX
Event: message-summary
Content-Type:
Some one on another list I subscribe to had a session with an annoying
IVR system at their doctor and posted this link.
http://www.pendulum.org/humor/humor_psych_hotline.html
--
Dave Cotton [EMAIL PROTECTED]
___
--Bandwidth and Colocation sponsored
In article [EMAIL PROTECTED],
Rainer Maier [EMAIL PROTECTED] wrote:
Hi all,
I am new on this list an I hope the posting is correct.
Welcome! I can't help with your ISDN problem, but I wanted to point
out a common posting mistake that newcomers make, which you did also.
When starting a new
Hello
In the following setup:
call coming from a pstn line - into FXO card - asterisk - SIP phone
i get an incredible loud echo in the SIP phone (about 0,5-1s) (everything i
speak into SIP phone microphone i hear in its speaker). The person calling
from PSTN is not getting any echo.
Which
Hi all,
I am new on this list an I hope the posting is correct.
I am using Debian Sarge, kernel 2.6.13 from www.kernel.org and want to
install the drivers for an AVM Fritz!PCI v2.0 ISDN card.
I used the directions from AVM but Asterisk allways stops with CAPI not
installed!
I am new to
[EMAIL PROTECTED] wrote:
In the following setup:
call coming from a pstn line - into FXO card - asterisk - SIP
phone
i get an incredible loud echo in the SIP phone (about 0,5-1s)
(everything i speak into SIP phone microphone i hear in its
speaker). The person calling from PSTN is not
Ahh wow.. that dial plan is seriously messed up... Try the default
one... it will work alot better and give you less lag time between
dialing a number and actually going through.
On 9/8/05, Joseph [EMAIL PROTECTED] wrote:
On Thu, 2005-09-08 at 23:29 +0200, Sander wrote:
What is your problem
Title: BRI debug, national ISDN speech call problem
hello,
I have a Junghanns QuadBRI card in my asterisk server. I'm able to dial connect to local numbers through the ZAP interfaces however when I try to dial national numbers with the according area code the connection fails, an intense
T400P(tormenta2) is based entirely off of the public zaptel spec,
Digium doesn't make them anymore, You can still get almost an exact
copy of the T400P from Varion a clone card maker.
The TE405P has several design and firmware optimizations over the T400P and the TE405P switches lines faster.
Hi,
I am looking for some suggestion on getting remote SIP phone connected to an
asterisk server where their is a NAT on the remote home users network/remote
site and whether I need the asterisk box on a public IP.
I know their is some problem with SIP and NAT (Although certainly not an
On Friday 09 of September 2005 13:14, Andreas Sikkema wrote:
[EMAIL PROTECTED] wrote:
In the following setup:
call coming from a pstn line - into FXO card - asterisk - SIP
phone
i get an incredible loud echo in the SIP phone (about 0,5-1s)
(everything i speak into SIP phone microphone
Thanks Steve most helpful,
Yes I am in the UK and we have master and secondary sockets for implementing
ring. Master sockets have the ring Capacitor and secondary sockets do not, where
it is designed that the ring voltage is already supplied on the 3rd pin.
I am attempting to get my first
Your config files?? Zaptel zapata ? Maybe you have a setting wrong i had the
same problem but then with pri lines now it's gone. You can hear yourself as
loud as the other person that is calling you? And what sipphone do you use
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
Did you search the maillist archives for hybrid echo cancellation?
Hello
In the following setup:
call coming from a pstn line - into FXO card - asterisk - SIP phone
i get an incredible loud echo in the SIP phone (about 0,5-1s) (everything
i
speak into SIP phone microphone i hear in its
Not all providers use crc4 you can try to remove the entry
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens altus
Verzonden: vrijdag 9 september 2005 7:24
Aan: Baris Simsek
CC: asterisk
Onderwerp: Re: [Asterisk-Users] pri gateway
These are my configs for
On Friday 09 September 2005 00:25, Darren Wright wrote:
Anyone care to elaborate on the differences between the T400P and the
TE405P?
This is described on Digium's site.
In a nutshell: newer, more efficient design, utilizes PCI burst mode and can
reduce load on your server. They even have
On Friday 09 of September 2005 13:38, Sander wrote:
Your config files?? Zaptel zapata ? Maybe you have a setting wrong i had
here they are:
zapata.conf:
[channels]
context=incoming
signalling=fxs_ks
usecallerid=yes
cidsignalling=v23
cidstart=ring
callerid=asreceived
busydetect=yes
On Thu, Sep 08, 2005 at 07:28:34PM +, Mike Hemstock wrote:
On Tuesday 06 September 2005 15:27, Mike M wrote:
Imagine what a network of systems composed of Asterisk, ham radio, wifi,
generators, batteries, and a reserve of fuel could have done for the
Gulf coast. I have all of the
On Friday 09 of September 2005 13:39, Soner Tari wrote:
Did you search the maillist archives for hybrid echo cancellation?
well, yes i googled a lot beforehand, came across the hybrid issue, but from
what i unerstand, the hybrid is a piece of hardware that sits on the X100P
card. I'm not sure
Dear All,
I'm having problem with spandsp/txfax,
I'm not able to send a multi paged tiff file,
the fax machine receives the first page of the document and
complains about communication problem.
The file what I'm trying to send has 2 pages and
is received generated by spandsp/rxfax.
Searched in
Hi,
I am a beginner in asterisk. Implementing it in my dept in India
using TDM400b card with asterisk, zaptel, libpri version latest of CVS
HEAD
Callerid on my system is coming tough.
Asterisk doesnot finishes the callerid spill and Cancells it.
After going through code in Callerid.c and
Did you search the maillist archives for hybrid echo cancellation?
well, yes i googled a lot beforehand, came across the hybrid issue, but
from
what i unerstand, the hybrid is a piece of hardware that sits on the
X100P
card. I'm not sure what can be done about it - the card doesn't seem to
I'd recommend the following link for the start:
http://www.voip-info.org/tiki-index.php?page=Causes+of+Echo
I have read the echo related info on voip-info. But this didn't help me much.
thats why i send my initial post to this list. I know the problem is related
to the FXO card, but none of
spa3k is really an spa3000 (k = 000).
Try:
www.sipura.com
www.voxilla.com
www.voipsupply.com
or any number of other suppliers of voip equipment.
Oh Ok I guess I was taking it too literally!!!
With a pair of SPA3000's, would I not even need *?
Depends on what you are trying to
On 08/09/05, Mark Phillips [EMAIL PROTECTED] wrote:
Bloddy 2E's; always wrong.
Mark G7LTT/KC2ENI
I know some G7s who are occasionally wrong, too :-)
Peter G4MJS / 9M6BAA
--
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
On 09/09/05, Mike M [EMAIL PROTECTED] wrote:
On Thu, Sep 08, 2005 at 07:28:34PM +, Mike Hemstock wrote:
On Tuesday 06 September 2005 15:27, Mike M wrote:
Imagine what a network of systems composed of Asterisk, ham radio, wifi,
generators, batteries, and a reserve of fuel could have
Hi !
I have a same problem here, tried even more different versions of *,
libtiff, spandsp, and lot of different hardware (X100P,TE110P,mISDN ISDN
BRI, and TDM400) and a lot of different faxes (but mostly Panasonic ones) on
a various landlines (over couple of POTS lines, over ISDN BRI where
The operative word here being occasionally. Of course, bad spelling
doesn't count.
flameprooftrousers And as for those half baked M3's ...
/flameprooftrousers
Peter Bowyer wrote:
On 08/09/05, Mark Phillips [EMAIL PROTECTED] wrote:
Bloddy 2E's; always wrong.
Mark G7LTT/KC2ENI
I know
[EMAIL PROTECTED] wrote:
Ah - so the difference between your setup and mine is that you are using
Sangoma (presumably) and I'm using Digium. Looks like the Digium is
significantly more efficient then.
It could also be that I'm using Net-SNMP to query my cpu usage and even
when the machine
I store my speed dial numbers in the astdb key speeddial with the number and
then name separated by a -.
This dial plan works fine:
[speed-dial]
exten = _*0XX,1,Set(temp=${DB(speeddial/${EXTEN:2})})
exten = _*0XX,2,Cut(number=temp,,1)
exten = _*0XX,3,Goto(house-phones,${number},1)
The log
echotraining=yes
echotraining=800
This looks odd to me, I would use just:
echotraining=800
Gain setting are important of course. You could use ztmonitor for that.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing
I'm not writing a printer driver so I probably couldn't use the idea.
I've always disabled CUPS.
Regards,
Chris
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, September 09, 2005 1:04 AM
Subject: Re: [Asterisk-Users]
Hi Montrealers !
I would like to create a usergroup for Montreal's asterisk users.
If you are interested, contact me and we'll schedule a beer/coffee meeting
downtown next week.
Sincerely,
Adrien
--
Adrien Laurent - CIO
514-284-2020 ext 202
[EMAIL PROTECTED]
www.modulis.ca
Hold on here folks,
I'm guessing that the original poster of this thread isn't a member of
his local RAyNet team.
Whilst I don't profess to be an expert at this I have been doing
emergency radio for quite some time and have seen service at the
Lockerbie bombing, Docklands bomb, Ground Zero
On Friday 09 September 2005 09:05, Matthew Boehm wrote:
It could also be that I'm using Net-SNMP to query my cpu usage and even
when the machine is idle, SNMP reports about 20% CPU usage which is
incorrect.
I'm sorry but if your Dell Xeon 3.0GHz is topping out at 50% CPU for 40 ulaw
calls
im really newbie, and i have a siemens digital pbx work in my work. i
have 4 outside lines and the pbx has a E1/PRI card. what i need to ask
my siemens provider(techinicians) to do in the pbx?
i only have in my pbx the 9 to get a line to go outside is very simple. but i
dont know what i
need to
Hi,
You might want to join MLUG which has a lot of VOIP users/experts.
http://www.mlug.ca
Andre Courchesne - Consultant
http://www.net-forces.com
Home of the RockHopper Firewall/Server
Adrien Laurent wrote:
Hi Montrealers !
I would like to create a usergroup for Montreal's
Chris Shipman schrieb:
What build of SpanDSP did you use?
spandsp-0.0.2pre18
I'm working on a windows program
so users can print to a local printer which will be forwarded to the
asterisk server to be faxed.
So far the program FTPs a Tiff to the Asterisk server to be faxed
On Friday 09 of September 2005 15:23, Soner Tari wrote:
echotraining=yes
echotraining=800
This looks odd to me, I would use just:
echotraining=800
I have commented the first echotraining. Not that it changed anything ;)
I have also just compiled 1.2.0-beta1 asterisk. As far as my
In article [EMAIL PROTECTED],
John Hill [EMAIL PROTECTED] wrote:
I store my speed dial numbers in the astdb key speeddial with the number and
then name separated by a -.
This dial plan works fine:
[speed-dial]
exten = _*0XX,1,Set(temp=${DB(speeddial/${EXTEN:2})})
exten =
It seems that using AstFax would mean that you would have to have a
dedicated email server for faxing.
AstFax expects the number in the email address.So all emails would have to
be piped to the program.
Which maybe fine in some circumstances.
Am I wrong?
regards,
Gain setting are important of course. You could use ztmonitor for that.
the asterisk server is a racked machine with no sound card. so can't use
the
ztmonitor. If everything fails i'll dig it out and try this
You don't need a soundcard to use ztmonitor, what do you mean by that?
Marek, you
Echo can is just a few weeks away , and if I recall correctly, the
Sangoma echo can will effectively monitor for, and handle, echo up to
128MS, whereas on a Quad Span Digium card I think you only get echo can
up to 16MS.
Cory J Andrews
Partner / Purchasing
+++
VOIPSupply.com -
Hi,
I want to extend my asterisk stuff and buy some Planet devices, to be
certain I'm going to buy PLANET VIP-050 with FXO and FXS modules. Has anyone
heard about it. Is it compatible with Asterisk, or it would cause a lot of
problems. Dose anyone have some experience with it??
All the
Or shitcan the onboard raid and get a real hardware raid controller like
a 3ware card (if you are stuck on IDE / SATA.) Reduces complexity.
On Fri, Sep 09, 2005 at 09:23:44AM +0300, Tzafrir Cohen said:
On Fri, Sep 09, 2005 at 08:05:09AM +0200, Clive wrote:
Hi
I discovered that most
On Fri, Sep 09, 2005 at 01:46:57PM +0100, Peter Bowyer wrote:
On 09/09/05, Mike M [EMAIL PROTECTED] wrote:
On Thu, Sep 08, 2005 at 07:28:34PM +, Mike Hemstock wrote:
On Tuesday 06 September 2005 15:27, Mike M wrote:
Imagine what a network of systems composed of Asterisk, ham
Cory,
As I understand it the echo detection will only run in the first 100?
Milliseconds. The statement that it will 'monitor for' echo is
misleading. It will detect echo at the start of a call like all other
current echo cancellation, correct?
On another note, Sangoma tech support is good. I
I agree, use either SCSI with hardware raid with a battery backed cache
or use sata/ide with linux software raid. Linux raid SCSI also works
well, but if you go for the scsi drives might as well get the controller
too.
The firmware raid on the cheap sata/ide cards have left me stranded
several
On Friday 09 of September 2005 16:08, Soner Tari wrote:
Gain setting are important of course. You could use ztmonitor for that.
the asterisk server is a racked machine with no sound card. so can't use
the
ztmonitor. If everything fails i'll dig it out and try this
You don't need a
I know that the MGCP stack will work over the 10/100 port as I've been
deploying them for the past few months and using them to terminate voice
traffic for various customers using a Class5 VoIP capable softswitch. It
almost seems like the adtran TA6xx doesn't reply back to the asterisk
No that just means you are not calling ztmonitor properly.
Try running ~# ztmonitor 1 -v
Jared Armstrong
OmniSpear, Inc.
Web Network Solutions
-Original Message-
From: Marek Zachara [mailto:[EMAIL PROTECTED]
Sent: Friday, September 09, 2005 10:43 AM
To: Asterisk Users Mailing List -
On Fri, 2005-09-09 at 08:28 -0600, Damon Estep wrote:
On another note, Sangoma tech support is good.
I can second that, 3/4 years ago I installed one of their cards and
couldn't get it running. I phoned them and they talked me through for
around 2 hours until it was really working. I had the
I'll have to have Doug refresh me on it, I don't want to mislead
anyone. He was using the term Dynamic Echo Cancellation. Yes, their
support is very good given the majority of the company are
hardware/software engineers. They offer pretty much an unconditional
guarantee that their hardware
Hi,
When a SIP client registers on Asterisk server, why it expires after
certain amount of time?
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I also agree. You want a raid controller that has it's own CPU. You want
hot spare, hot swapping, status lights, etc. to be handled by that
controller. If you have a hot spare you want automatic cutover to that
spare drive. You are not limited to SCSI with these controllers. Some
manufactures
Can anybody see a way of detecting the current number of
retries remaining to a call file in the extension context that it is calling?
E.g. If I want to schedule a fax and I want to feed an email
back to the sender stating that the number is busy 2/5 retries remaining?
Steve
Ben,I assume from your posts that you are in an area serviced by a small independent phone company. We have this situation as well, and you might very well have to pay much higher rates than other areas. You might try contacting Long Distance carriers (we use Paetec) and find out if they can work
On Fri, Sep 09, 2005 at 09:31:06AM -0400, Mark Phillips wrote:
Generators require fuel which is always in short supply and batteries
die out quickly.
Fuel and batteries and power efficient systems need planning and
management. Don't overlook solar panels as an energy source. They
need to
I have successfully been using OH323 v0.6.5 with Asterisk 1.0.x.
I now need to move to CVS HEAD in order to use some features that
are not in v1.0.x, and am trying to compile OH323 to use with it.
On the InaccessNetworks site, it ways that OH323 v0.7.1 is for HEAD.
However, when I compile it, it
It's not that easy then everytime you want to change someting for testing
you have to ask them to change something i can give you the software for
programming siemens pbx if you want
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Pablo Allietti
Hello,
In the examples on voip-info.org DBSet and DBGet are used to store
configuration variables such as immediate call forwarding settings etc.
I would like to store these seetings in a mysql database, so that they are more
easily accessible from a user configuration page on a webserver.
yes you are right. It does run this way. However this still does not solve the
echo issue. I can see that RX is following TX with about 70% of the signal
strength - but what to do about it? :(
Marek
On Friday 09 of September 2005 16:46, Jared Armstrong wrote:
No that just means you are not
Hello Folks!
in my sip-logs i see that asterisk uses the User-Agent ID Asterisk
PBX:
SipClient: Received: 16:34:03.023
-
BYE sip:[EMAIL PROTECTED]:44343;transport=udp SIP/2.0
Max-Forwards: 10
Record-Route: sip:213.2XX.XXX.XX8;ftag=as2eb3c466;lr=on
Via: SIP/2.0/UDP
[EMAIL PROTECTED] wrote:
I would like to store these seetings in a mysql database, so
that they are more easily accessible from a user
configuration page on a webserver. Since these settings need
to be checked in the dialplan for each call to the extension,
it seems a bit to much to have to
Hi David,
I just looked at my iax.conf on one of my boxes in Argentina and
actually there are no jitterbuffer settings indicated so I'm assuming it
is using Asterisk defaults.
We are experimenting with G.729 on these IAX trunks also and I just
realized I have no accurate means of measuring
Hi,
have looked around for some documentation what effect a reload has on a
running system but I can't find any relevant information. What I would
like to know is what type of configuration changes (if any) that will
interfere with already established calls if I do a reload. I'm only
using
Forgot the version:
Asterisk 1.0.7
On Thu, 2005-09-08 at 17:49 -0400, David Hajek wrote:
Nice. Thanks.
What Asterisk version? Can you lookup jitterbuffer settings?
Thanks a lot.
___
--Bandwidth and Colocation sponsored by Easynews.com --
On Fri, Sep 09, 2005 at 10:55:52AM -0400, Paul wrote:
I also agree. You want a raid controller that has it's own CPU. You want
hot spare, hot swapping, status lights, etc. to be handled by that
controller. If you have a hot spare you want automatic cutover to that
spare drive. You are not
The original dial plan was:
(*xx|[3469]11|0|00|[2-9]xx|1xxx[2-9]xxS0|.)
So I change it to:
(xx.|*xx.|#xx.)
I don't think it is complicated, beside it works with Sipura-3000, and I
don't see a reason why shouldn't it work with Sipura-2002.
I contact Sipura technical support
Zeeshan Zakaria wrote:
Hi,
When a SIP client registers on Asterisk server, why it expires after
certain amount of time?
Because it is the way SIP registrations work. For more information, find
a SIP book or read the SIP RFC 3261.
SIP phones need to re-register every once in a while to tell
Banging my head against a brick wall trying to get a working H.323
implementation for CVS-HEAD. (The ONLY H.323 I have had working is
OH323 v0.6.5 with CVS-STABLE - see my other post regarding compile
problems on OH323 for HEAD)
So, I thought, lets try this wonderful chan_woomera (dubbed H.323
On Fri, Sep 09, 2005 at 05:39:22PM +0200, Sander wrote:
thanks Sander but i have the soft, and i can enter to the pbx conf and
modify all settings, but i dont know how settings i need to change.
It's not that easy then everytime you want to change someting for testing
you have to ask them
Did you try to get a milliWatt test phone number from your telco? It
was really easy for me. I called the business office and told them that
my new digital pbx was having some awful echo trying to deal with their
lines; all I needed was a milliWatt test line to balance my receive and
Olle E. Johansson wrote:
SIP phones need to re-register every once in a while to tell the server
where it can be reached. If you have a soft phone on a laptop that you
move from network to network - home, office, airport, Barnes Noble etc
- you want to be reached on the IP address you use
Hi
Asterisk 1.0.9-BRIstuffed-0.2.0-RC8m is connected to a T-Concept
XI520 System. Phone calls on both directions do work, but transfers
are not possible. Asterisk recognizes that some sip phone requests a
transfer. Is it possible to forward this transfer request to the
XI520? Users of analog and
Do you want to connect the asterisk with pri or with internal isdn? And what
model pbx do you have? then i can tell you how to configure? Maybe some
screenshots with it
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Pablo Allietti
Verzonden: vrijdag 9
On Friday 09 of September 2005 19:04, Mojo with Horan Company, LLC wrote:
Did you try to get a milliWatt test phone number from your telco? It
was really easy for me. I called the business office and told them that
Well, unfortunately not everyone has similiarly helpful telco providers ;)
To
Which is kinda annoying because grandstreams (at least 1.0.5.23 firmware
anyway) don't do that... I have to powercycle the one on my desk once
an hour if I want it to ring on incoming calls otherwise asterisk
'forgets' about it (the cisco in the other room seems to be fine).
I'd love an
In article 200509091317.j89DGtY3019393 at commserver.noach.com,
John Hill jhill at noach.com wrote:
I store my speed dial numbers in the astdb key speeddial with the number
and
then name separated by a -.
This dial plan works fine:
[speed-dial]
exten =
Seriously however, even if i could get some reference signal, how can i tune
the card apart from changing the rx/tx gain? even with these two down to -6.0
dB i'm still getting awful lot of echo ... The card is a simple X100P clone.
In my situation, before I found the reference signal, I found
On Fri, 9 Sep 2005, Matthew Boehm wrote:
[EMAIL PROTECTED] wrote:
Ah - so the difference between your setup and mine is that you are using
Sangoma (presumably) and I'm using Digium. Looks like the Digium is
significantly more efficient then.
It could also be that I'm using
On Fri, Sep 09, 2005 at 07:20:58PM +0200, Sander wrote:
uuauuu that will great!
i cant undertand too much about internal connection because. i have a PC
with a te110p and my siemens is a hipath 3500 with a s2m/pri card is a
E1 card. but i dont know how to connect between them. i have always the
Hello.
Is there any know issue with Asterisk 1.0.9 concerning intermittent SIP
registration issues.
My SIP hard phone (aastra 9133i) and soft phone (xlite) keep losing
registration so calls to them go direct to VM although calling to other
phones from them works fine.
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