We've had this combination 206-xSeries and TE110P , but the zttest results
were not in the range above 99,76%
as well we had lots of echo-problems
...we changed to an other hardware platform
[EMAIL PROTECTED] wrote on 03.02.2006 16:49:14:
Hi,
I have an IBM xSeries 206 and now looking at
Hi,
Sorry for keep hammering the list with this annoying question.
Can we use Intel 536 ep (not 537ep that is in wiki) as x100p clone?
I know I've asked it in this list a couple days ago but none responded
so far and I'm getting frustrated pairing it with asterisk as the zaptel
driver could
Does anyone have a neat idea as how to
bill inbound calls per minute(second) real time?
I've been pplaying with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the Connect fee(if I put one)
and keeps it that way no matter how
Hi,
http://www.voip-info.org/wiki/view/Asterisk+sound+files+international
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philippe Lang
Sent: Monday, February 06, 2006 3:46 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] French and
What will be the g729 and g723 codec capacity from
Intel IPP liberary without License?
Because still i am developing all billing and other
application for asterisk so first i want to use these
codecs for test once all our system become stable i
will buy the license.
S0 please let me know how
thanks for the answer.
is this something new in 1.2?
if so, where is it documented, and what is the point of autocreatepeer=yes if this is the case?
-yair
On 2/5/06, C. Zerbo [EMAIL PROTECTED] wrote:
you need to setup a asterisk peer at port 5070 in sip.conf to get the callreplying correctly
hi,
How good or bad is the EC in Asterisk?
Can anyone prove that it works at all and what it's limitations are?
I ask cause I have some problems with this myself which variate from
call to call, and I see from others that Echo Cancel is a quite common
topic.
Jan
On Mon, 2006-02-06 at 10:49 +0100, [EMAIL PROTECTED] wrote:
hi,
How good or bad is the EC in Asterisk?
Can anyone prove that it works at all and what it's limitations are?
I ask cause I have some problems with this myself which variate from
call to call, and I see from others that Echo
hi,
How good or bad is the EC in Asterisk?
Can anyone prove that it works at all and what it's limitations are?
I ask cause I have some problems with this myself which variate from
call to call, and I see from others that Echo Cancel is a quite common
topic.
Jan
I have
Thanks.
I'm able to getting the asterisk calling back to my cellphone. But when I get
to the authentication I get this message when I start to dial in my password:
NOTICE[5178]: rtp.c:509 ast_rtp_read: Unknown RTP codec 96 received
Is this a DTMF failure of some sort?
Thanks again.
virtually all software echo cancelers cannot get double echo removed
completly. It can get the first one but not the second one. There
are
instances where you get a 2nd echo, so ... Asterisk is no exception
from this afaik nothing software only based is.
If you really want good echo
Hello asterisk-users,
Christian Schmidt, 05.02.2006 (d.m.y):
My asterix now accepts calls coming in via CAPI.
I'm so sorry: WHat I wanted to write was:
My asterisk now accepts calls coming in via CAPI.
;-)
Regards,
Christian Schmidt
--
Aus der Kriegsschule des Lebens - Was mich nicht
On Mon, 2006-02-06 at 21:46 +1100, James Harper wrote:
Just an enquiring mind wanting to know, but how is a hardware solution
different to a software solution? The echo cancellers in the Digium
hardware presumably just use the same sort of algorithms as the software
versions, so it is just
Im curious. Does anyone have experienced echo-problems that later where solved
by buying a hardware-echo canceller such as the Wildcard TE411P?
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För James Harper
Skickat: den 6 februari 2006 11:46
Just a Test
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On 2/6/06, Arne Morten Johansen [EMAIL PROTECTED] wrote:
Thanks.
I'm able to getting the asterisk calling back to my cellphone. But when I get
to the authentication I get this message when I start to dial in my password:
NOTICE[5178]: rtp.c:509 ast_rtp_read: Unknown RTP codec 96 received
Sorry for the blank email, here's what I meant to send:
I haven't seen that error before, sorry. A quick search using google
turned this up though:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg08901.html
Not sure if it's relevant in your case. What is asterisk using to
dial
I can't seem to change the default registration for IAX clients:
Feb 6 12:22:52 NOTICE[7883]: chan_iax2.c:5673 update_registry: Restricting
registration for peer 'virbiage' to 60 seconds (requested 3600)
Feb 6 12:23:03 NOTICE[7883]: chan_iax2.c:5673 update_registry: Restricting
registration for
Hi Abdul
You will need to download and install the Intel API which is then used
to compile the patched G723 codec.
Hope this helps.
Kind Regards
Garth
Abdul Lateef wrote:
Hi All,
I have one Carrier which is supporting only G.723.1,
how i can put in my extentions.conf to send calls to
On Mon, 6 Feb 2006, trixter aka Bret McDanel wrote:
Again, I know Sangoma is a sore subject with some on this list, but the
echo cancelation stuff I heard while presented by a Sangoma employee was
not Sangoma specific, although it did include some research into
different hardware/software based
It's a sip channel.
http://www.asteriskguru.com/tutorials/unknown_codec_received.html
This might work, but I don't know where to find the source-code of asterisk.
I've used the ebuilds in gentoo portage to compile asterisk. And I'm not
exactly a linux type of guy, so this is not my field.
I
I often see this happening:
- ChanIsAvail returns Zap/94
- I dial out via it.
- And then Moving call from channel 94 to channel 101
Why is it moving to channel 101?
Edwin
--
Edwin Groothuis |Personal website: http://www.mavetju.org
[EMAIL PROTECTED]| Weblog:
Hi
I must be missing something here:
do the zaptel tarball (1.2.2 and 1.2.3) miss the file .version? Without
it version.h will be generated with an empty version number and some bad
things will happen, IIRC.
I've added it mnually to the Debian package. But I have a feeling that I
have somehow
On Mon, 2006-02-06 at 04:01 -0800, [EMAIL PROTECTED] wrote:
On Mon, 6 Feb 2006, trixter aka Bret McDanel wrote:
Again, I know Sangoma is a sore subject with some on this list, but the
echo cancelation stuff I heard while presented by a Sangoma employee was
not Sangoma specific, although it
You can get asterisk source from http://www.asterisk.org/
Arne Morten Johansen wrote:
It's a sip channel.
http://www.asteriskguru.com/tutorials/unknown_codec_received.html
This might work, but I don't know where to find the source-code of asterisk. I've used the ebuilds in gentoo portage to
I have a Motorola Razr successfully connected to asterisk using a
bluetooth dongle and chan_bluetooth. Here's some issues I've run
across:
- You have to ignore the instructions in bluetooth.conf, saying to run
sdptool search --bdaddr xx:xx:xx:xx:xx:xx 0x111F to determine the
correct channel to
well I've heard that there are open source IP phones given away for free
in WALMART, I'm seriously thinking to get couple of 'em!!
What phone would this be? I didn't notice any, but there's 5-6
Wal-Marts within an hour's drive, I'd love to try to find some. Never
can have too many. Are they
Hi all,I have an * box dual Xeon, 4Gb ram, 2 A104.Normally I use gsm codec, but to allow using faxes, I let some users to use g711 as default codec.My question is:Is it possible to detect what a certain call is?
So if is a phone call I'll use gsm, if is a fax I'll use g711.Thanks to all--
On 2/6/06, Joseph Tanner [EMAIL PROTECTED] wrote:
well I've heard that there are open source IP phones given away for free
in WALMART, I'm seriously thinking to get couple of 'em!!
What phone would this be? I didn't notice any, but there's 5-6
Wal-Marts within an hour's drive, I'd love to
[EMAIL PROTECTED] wrote:
On Mon, 6 Feb 2006, trixter aka Bret McDanel wrote:
Again, I know Sangoma is a sore subject with some on this list, but the
echo cancelation stuff I heard while presented by a Sangoma employee was
not Sangoma specific, although it did include some research into
Guys! I was only teasing you! Sure there couldn't be any open sourse IP phones because they are not software! PLUS WALMART sells breathing air if he ever had the chance, not to mention IP Phones!!!
Truely/
Joe
From: Gonzalo Servat [EMAIL PROTECTED]Reply-To: Asterisk Users Mailing List -
Funny funny. In this day of free (after rebate) PAP2s, a free (again,
I assumed after rebate) IP phone seemed plausible. BTW, check
walmart.com, they do indeed sell ip phones.
I guess I'll just have to use one of my free DTA310s or my free PAP2 instead.
Joseph Tanner
On 2/6/06, Gonzalo Servat
Jose,
There are No open source IP phones, I was only joking, I assumed you should know what an open source is.
Truely/
Joe
From: Joseph Tanner [EMAIL PROTECTED]Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comTo: Asterisk Users Mailing List -
Yes, I actually just removed the VPM. After doing so, I had echo at
the beginning of the call, but it trained out after a second at least
making it usable. I plan on contacting digium support this morning. Do
you know if there are any docs specific to this card or the vpm module
itself?
On Feb 6, 2006, at 5:04 AM, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Im curious. Does anyone have experienced echo-problems that later
where solved by buying a hardware-echo canceller such as the
Wildcard TE411P?
Yes. I turned off all echo can on the wildcards and bought external.
On 2/6/06, Joseph Tanner [EMAIL PROTECTED] wrote:
Funny funny. In this day of free (after rebate) PAP2s, a free (again,
I assumed after rebate) IP phone seemed plausible. BTW, check
walmart.com, they do indeed sell ip phones.
I guess I'll just have to use one of my free DTA310s or my free
I have problem with DTMF signal, in brazil use deferent
tones level, between low tone and high tone
In low tone use + or 1 dB regarding of high
tone, but in asterisk low and high frequencies send a level tone equals
How to I can change this.
The folks at Sangoma- they know more about echo than most of us will ever forget- you want to speak with David M.I recently upgraded 6 Sangoma 104's to 104d's to resolves intermittent echo issues. The 104d is a magnificent marvel by folks who understand hardware. Holdthe 104d beside a TE411P
Jose,
Close, check the bottom of my messages, and the name sent along with
my email address; it's Joseph not Jose.
There are No open source IP phones, I was only joking, I assumed you should
know what an open source is.
There are no open source routers, no open source PBXs, no open source
Hi,
I have tried both the stable version ARI-00.04.006 and the development
version ARI-00.05.018 with the same results. I can see call detail
records just fine but I cannot see any voicemail. I am using the
voicemail extension and password to log in but I still do not see
anything. If I log
Did you test the echo delay?
will 64ms be suffitient?
You can easily test the delay by recording the transmit and receive path
to a sound file and using some sound editing software see how big the
delay is.
That is how I did it when I worked for a Telco in Switzerland on theis
TDM switch they
There's idiots that tell people about free, and cheaper-than-free
deals all the time. Here's just one such idiot:
http://www.fatwallet.com/forums/messageview.php?start=0catid=24threadid=524641
BTW, the idea is to get all that you want for yourself first, THEN
tell everyone about the deal.
Hello Friends,
I am experiencing a problem. The rtp packets which detect dtmf from inband
are being dropped. I have tried a priority ip address which allows voip packets
first but it didnt work out. Asterisk is dropping only dtmf packets. I am using
Sip protocol. Is there any way in asterisk
James Harper wrote:
virtually all software echo cancelers cannot get double echo removed
completly. It can get the first one but not the second one. There
are
instances where you get a 2nd echo, so ... Asterisk is no exception
from this afaik nothing software only based is.
If you
Guys! meant no harm, no one is stupid, it's just the fact that such deals are possible but not with the conflict of an open source hardware ( there is no open source hardware ).
Appologies!
Truely/
Joe
From: Joseph Tanner [EMAIL PROTECTED]Reply-To: Asterisk Users Mailing List - Non-Commercial
Krystian Filiks wrote:
Did you test the echo delay?
will 64ms be suffitient?
You can easily test the delay by recording the transmit and receive
path to a sound file and using some sound editing software see how big
the delay is.
That is how I did it when I worked for a Telco in Switzerland
Doug Lytle wrote:
Krystian Filiks wrote:
Did you test the echo delay?
will 64ms be suffitient?
You can easily test the delay by recording the transmit and receive
path to a sound file and using some sound editing software see how
big the delay is.
That is how I did it when I worked for a
Krystian Filiks wrote:
If they could hear their own voices than I would not invest in echo
cancelling and for this is the far end responcible so I would take
contact with the service suppliers and ask them if echo canceling is
included.
These are standard analog (Centrex) lines.
Echo
Joseph Rothstein a écrit :
I can't seem to change the default registration for IAX clients:
Feb 6 12:22:52 NOTICE[7883]: chan_iax2.c:5673 update_registry: Restricting
registration for peer 'virbiage' to 60 seconds (requested 3600)
Feb 6 12:23:03 NOTICE[7883]: chan_iax2.c:5673 update_registry:
HelloWe recently moved to Asterisk 1.2.4 (from 1.0.x) and our 10 Uniden UIP200 have stopped working ever since.We can make a call with the UIP200 to any other extensions, but it can not receive a call. In fact the UIP200 always appears offline:
It does show up in asterisk a few seconds after the
Hi All,
anyone in Spain is using a ONO PRI? In that case are you experiencing any
problems with asterisk
and ONO? Wich are your zaptel parameters?
Thanks
Xavier Gil
__
LLama Gratis a cualquier PC del Mundo.
Llamadas a fijos y
First, about the Jabber library: I'm using Asterisk Perl and the
Jabber module for Perl.
About dinmically loading the jabberid list, welll that's the problem I
had and now I'm developing that. I thought about (and it's what I'm
doing) generate a little database in XML in which you would put
If you wnt to do it quick, I've seen this in another post of this
list, and I think is good:
exten = s,1,System(/bin/echo -n -e '${CALLERIDNAME}
${CALLERIDNUM}'| nc -w 1 192.168.1.16 10629)
then you have tyo be monitoring that port and capture the information,
you can do that in VB.
2006/2/6,
Hi Alexander, Thanks for the quick response. Actually I tried out this. I tried like - Action: Originate Channel: SIP/111 Application: MeetMe Data: |qdwx ActionID: ffe56637 But actually, it invites 111 and when 111 accepts the call, it will ask for conference number and places 111 into
Hey asteriskers!
I know that may look weird, but it's happening:
We have an * server running in a wireless(cellular) operator for IVR services, we bill them per minute, but there is a remarkable difference between our CDR records and their billing system.
* server have a Sangoma, and 3 PRI_ISDN
YAC is a nice popup application (for Windows) to eat alerts just like
the one below.
http://sunflowerhead.com/software/yac/
Peter
On 06/02/06, Facundo Ameal [EMAIL PROTECTED] wrote:
If you wnt to do it quick, I've seen this in another post of this
list, and I think is good:
exten =
hi,
Depends, but if it is a standard H.323 yes. If it is 'Avaya H.323' NO.
I am not sure if * support it 'as is', but you probably need to run
H.323 as a Terminal endpoint.
To test, just connect with OpenPhone.
'Avaya H.323' is basically RTP/RTCP as normal and H.323 towards their
PABX
AFAIK asterisk does not drop the packets, it just turns them into
silence if it detects a dtmf.
On 2/6/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello Friends,
I am experiencing a problem. The rtp packets which detect dtmf from inband
are being dropped. I have tried a priority ip
Jose,
Whatever man! we're cool.
Truely/
Joe
From: Joseph Tanner [EMAIL PROTECTED]Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comTo: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comSubject: Re:
On 10/21/05, Sean Cook [EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Sherwood McGowan
Sent: Friday, October 21, 2005 3:45 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
Hi all,
Im going to configure a middle asterisk
installation. Ill use a TE210P to connect a T1 channel bank and a PRI E1
line.
Im thinking on using a SuperMicro P8SCT Mother
Board that has a 1x 64-bit 133MHz PCI-X 3.3V.
In TE210P documentation Ive read:
The TE210P is a 32-bit 33MHz
Hi all,
I would like to eliminate about 150 lines in log /var/log/messages) every
time a call is placed/received
If I type, on the asterisk console,
set verbose 0
the lines in the log disappear, but it appears to me too drastic as a
method
The lines shown in the log don't appear (at least
Hi,
Is there any detailed guide/tutorial source online on queues?
Zach
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Yes. The wiki and voip-info.org
--- Zach A [EMAIL PROTECTED] wrote:
Hi,
Is there any detailed guide/tutorial source online
on queues?
Zach
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To
On Feb 6, 2006, at 5:08 AM, ammar Ali wrote:
Jose,
There are No open source IP phones, I was only joking, I assumed you
should know what an open source is.
The AG-168V is an open sourced ATA. Although the idea that Walmart
would give something (useful) away for free, was funny to me.
You may configure level (verbosity) of login in logger.conf file.
In file head some help.
Hi,
Simone
-Messaggio originale-
Da: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di
[EMAIL PROTECTED]
Inviato: lunedì 6 febbraio 2006 17.45
A: Asterisk Users Mailing List -
Phone Dev wrote:
Can SuperMicro slot (that is a 133Mhz slot) be used for TE210P card ?
Yes. Digium PCI cards work in a PCI slot capable of 133MHz, but the
cards operate at 33MHz and will slow down the PCI bus to that speed.
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Tzafrir Cohen wrote:
do the zaptel tarball (1.2.2 and 1.2.3) miss the file .version? Without
it version.h will be generated with an empty version number and some bad
things will happen, IIRC.
It was a bug in the release script; the script has been fixed for future
releases.
Hello all,
I've got Asterisk and a TE205P. One port on the TE205P talks over
E1 ISDN PRI to the outside world (thorugh BT). The other port talks to
an Avaya switch, also over E1 ISDN PRI. All is working well, except
that when people try to dial out from the switch through Asterisk (with
A customer of mine wants an IVR where the first 3 choices are
1 English
2 Spanish
3 French
I can build the IVR but how do I get the system prompts to then speak
the selected langauge. For example, a caller has selected Spanish and so
is routed to the Spanish part of the IVR. At some point he
Hi,
As many of you may know, we are undertaking several tests in order to test
the interoperability between several PBX IP from different vendors. Until
now, we were trusting that the VoIP IP PBX were good enough to be
interconnected directly, however, one of the vendors have presented the
SBC
Mark Phillips a écrit :
A customer of mine wants an IVR where the first 3 choices are
1 English
2 Spanish
3 French
I can build the IVR but how do I get the system prompts to then speak
the selected langauge. For example, a caller has selected Spanish and
so is routed to the Spanish part of
I have trouble getting ${EXTEN} or ${DID}
when receiving incoming callfrom TE110P in Japan.
Does anyone have idea how to fix
this?
Is this because TE110P does not support
INS1500 as switchtype?
pri debug does not show "Called Number".
Protocol Discriminator: Q.931
(8) len=32 Call Ref:
It does show up in asterisk a few seconds after the UIP200 reboot:
-- Saved useragent Uniden SIP Phone p2 Ver BS4.70 for peer uip200
but after about 5s I will get something like:
UIP200 is now unreachable.
It appears that, for whatever reason, the packet being sent to the phones
from
There is no good help on wiki and voip-info.org, I've gone through it
already.
Zach
-Original Message-
From: Dovid Bender [mailto:[EMAIL PROTECTED]
Sent: Monday, February 06, 2006 11:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Help on
CTech wrote:
Nabeel Jafferali wrote:
Am I missing something completely obvious? Is there a way to see why
Asterisk is sending 488 (i.e. what is not acceptable?).
Did you solve the 488 error? I run into the same problem as you.
Below is my invite msg. And my 488 response is exaclty like
Hello everyone,
As I promised at eTel last week, I have finished up work on my
Asterisk Native Sounds project. Here's a little diddy from astlinux.org:
---
Asterisk Native Sounds are a collection of audio prompts for Asterisk.
They will improve quality,
What kind of help do you need then?
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Zach A
Skickat: den 6 februari 2006 18:31
Till: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Ämne: RE: [Asterisk-Users] Help on queues
There is
I need practical examples showing solutions to various solutions, e.g.
how can a caller leave a queue and go back to the main menu instead of
hanging up and redialing, or how can a queue be started for an
extension, i.e. if 3-4 callers dial 201 and 201 is busy, instead of
sending calls to voice
Were you able to acomplish this?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan
Sent: Thursday, October 27, 2005 5:31 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Zapbarge feature available?
We would like to beable to
Kevin,
I have experienced the same issue. I get worse echo with the VPM
installed than with software EC. Have had it at 2 different sites with
2 different TE411P's.
- EricOn 2/6/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Stagg Shelton wrote: I just implemented a system using a TE411P hardware
All,
I had updated to 1.2.4 right when it came out. I had been working just fine.
Today I seem to be having recuring seg faults. can explain it.
How can I find why?
Anyone else experiencing this?
I am running (2) TDM04B cards (has been working since 1.0.9)
I have a handfull of UIP200 phones
Hi,
I'm using Asterisk 1.2.3 with the asterisk-oh323 channel driver, version
0.7.3.
Pwlib is V1.8.7 an OpenH323 is V1.15.6.
Following CallFlow:
SIP-UA - OpenSER - * - CCM
OpenSER routes all calls with prefix 60 to Asterisk, where I've configured
following extension:
exten =
I have a new asterisk server (details and conf files below) with
a TDM421p (2 fxs (phone), 1 fxo (pstn)).
Noisy FXS lines. (analog
phones connected to TDM400 fxs modules)
On the fxs lines there is a low static hiss all the time. For
example, if I pickup and press any key to break the
I running Asterisk 1.1 on Mandriva 2006.
Everything works fine, can connect with softphone, send outgoing calls to VOIP
provider.
The only (and big) problem is that Asterisk refuses to authenticate incoming
calls with the message (in the log):
Failed to authenticate user XX sip:[EMAIL
hi there,
I saw a page on voip-info about the thomson ST2030 phone. There is not
so much info on there, that's why I would like to raise a question
here.
Has anyone got hands-on experience with this phone? (with or without
extension module)
I am interested if it can be used (as SIP phone) in a
try host=dynamic in your sip peer entry hth
-Original Message-
From: Francois [mailto:[EMAIL PROTECTED]
Sent: Monday, February 06, 2006 11:45 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Will not authenticate incoming VOIP provider
calls
I running Asterisk 1.1 on
Stupid mistake!
Was looking for a nice server.
Nothing fancy, but as it was going to host *, it thought it wise, to opt
for something better than a no-name machine used for everydays desktop
jobs.
So i checked for a dell power-edge tower server:
- 3Ghz em64t cpu
- 1GB memory, ECC
- 80GB system
I get an echo when going from a SIP phone to a PRI trunk. I hear the
echo on the SIP phone. From reading some other post I think that I need
to tell me phone company to turn on echo canceling. If the echo was on
the other end than it would be my problem?
Is this right? What exactly should I
I just ran into this today, on 1.2.3 with Polycom IP 501 phones.
Message was from a potential customer looking for a PBX too... imagine
that embarrassment :)
Anyone know how to get this resolved?
Thanks,
Nathan
I had this happen today, also. I've seen it happen in the past, but
Kris,
This is very cool! Thanks for doing this. CPU power is at a much
higher premium than disk space, so it makes sense to have prompts in
multiple formats to cut down on unnecessary CPU usage. I'll trade disk
space for extra CPU muscle any day.
-MC
-Original Message-
From: [EMAIL
I've had no luck using a Zap extension as a member in a queue.
member = Zap/123444 doesn't seem to ring.
However:
member = SIP/someExt seems fine.
Thanks for all replies.
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At 07:54 PM 2/6/2006, Michael Heckner wrote:
Hi Joao,
Joao Pereira schrieb:
Hi,
As many of you may know, we are undertaking several tests in order to test
the interoperability between several PBX IP from different vendors. Until
now, we were trusting that the VoIP IP PBX were good enough to be
kurtz wrote:
I've had no luck using a Zap extension as a member in a queue.
member = Zap/123444 doesn't seem to ring.
That is not a valid member string for a queue. Zap/1 (as in channel 1)
is, but Zap/1/1234 is not. What you specified would look for channel
'123444', which I'm sure
This is very true, you must use hardware echo cancel and voice processing.
we use sangoma 104d hardware echo cancel card, it eliminates all echoes we
had.
Best Regards
Matt
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
On Mon, 6 Feb 2006, stoffell wrote:
Seems like a nice alternative to other phones, here in Europe.
(because linksys 942 is not easily available in europe, yet)
linksys 942 doesnt look very competetive anyway.
(2 10mbit ethernet ports? who is linksys kidding?)
a polycom 501 is a nicer phone
The long waited Ultimate GSM Gateway is finally out. This time we have managed
to source a new patch of brand NEW GSM Gateway at prices that is only 50% of
what the market rate. And with the SMS Function and many more...
For purchase please email gsm AT cyper-telecom.net. We accept paypal and
Single port GSM Gateway support 900 / 1800 GSM mode with external antenna.
Brand new unit and all of them will be tested before dispatch.
Extremely easy to setup and can be used out of the box without any
configuration. So should be good alternatively of phonecell or nokia pbx
etc..
Units
Sam Tam wrote:
The long waited Ultimate GSM Gateway is finally out. This time we have managed
to source a new patch of brand NEW GSM Gateway at prices that is only 50% of
what the market rate. And with the SMS Function and many more...
What part of 'non-commercial discussion' is hard for you
Hi,
I've had a bit of a
problem with one way audio, and it happens exactly when I believe it shouldn't
(and works perfectly when I would guess I could have issues.
Setup:
GrandStream
GXP2000---Linksys Router---Internet--Asterisk box (hosted
somewhere, fixed IP, no NAT)
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