hi guys
what is a class 5 soft phone, i did a search on google, didn;t find, please
let me know if any one knows.
cheers
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Buy the AMPHENOL 50pins male connector alone or with a pre soldered cable
and do what you want with.
Or buy a RJ11 pannel from the usual Telco accessories resellers.
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
Hi all,
What do you think about having a single extension in the dialplan that matches
everything and then delegates the next action to an external application
through AGI? I mean something like this:
exten = _.,1,AGI,catchall.agi,${EXTEN}
Then, catchall.agi could EXEC dialplan macros in
On Wed, 11 Jan 2006 [EMAIL PROTECTED] wrote:
Ok it solved my problem (immediate=yes in capi.conf) !!!
Here is the console log
***
CONNECT_IND ID=002 #0x201f LEN=0047
...
***
What is the meaning of
In article [EMAIL PROTECTED],
Carlos Alperin [EMAIL PROTECTED] wrote:
That is right for zaptel. But you still has to do modprobe wctdm on rc.local
before to load asterisk.
Any way to fix this?
Add wctdm to the list in MODULES and RMODULES in /etc/rc.d/init.d/zaptel
While you're at it,
Hi,
this is the scenario:
One * is placed in a central location with more than enough up/down
bandwidth. One * is placed behind a DSL 3000/384. Both * are linked via
IAX trunking. Everything is fine until the upload channel of the remote
site is filled with a download, then heavy voice
On 01/14/06 11:09 Pisac said the following:
But, it's not working anymore in Asterisk 1.2.1
when I test this with
noop(${CALLERIDNUM::3})
I get full callerid, not just first 3 numbers like it use to be on 1.0.9
i believe the syntax is ${CALLERIDNUM:3} and not as you're using it with
double
exten = _X.,1,AGI,catchall.agi,${EXTEN}
should do it for u
Hi all,
What do you think about having a single extension in the dialplan that
matches everything and then delegates the next action to an external
application through AGI? I mean something like this:
exten =
what is your type for sip entities? user, peer or friend?
When I leave a new message for a user I can't see any db queries?
ok, I'll check to use res_mysql.conf.
Thx
--- Ursprüngliche Nachricht ---
Von: Saul Diaz [EMAIL PROTECTED]
An: Asterisk Users Mailing List - Non-Commercial
exten = _X.,1,AGI,catchall.agi,${EXTEN}
should do it for u
Hi,
since I also have some applications that starts with *, like
[app-clir]
exten = _*67.,1,SetCallerPres(prohib)
exten = _*67.,2,Goto(${EXTEN:3},1)
I thought I could use _. instead of _X., that would match only numbers.
However,
On 14 Jan 2006, at 10:47, Koopmann, Jan-Peter wrote:Hi,this is the scenario:One * is placed in a central location with more than enough up/downbandwidth. One * is placed behind a DSL 3000/384. Both * are linked viaIAX trunking. Everything is fine until the upload channel of the remotesite is
the problem appears no matter where I terminate the call (IAX or
Zap), and I don't have that problem on a 1.0.7 connected to the same
PRI lines and IAX servers , what I have to check ? looked in confif
files but appears to be the same (indications, modules loaded, iax,
zapata, the
No,
${CALLERIDNUM}:3 erase first 3 digits
${CALLERIDNUM}::3 returns first 3 digits
${CALLERIDNUM}:3:3 should erase first 3 digits and return next 3 digits
So,
if
${CALLERIDNUM}=0123456789
Then
${CALLERIDNUM}:3 returns 3456789
${CALLERDINUM}::3 returns 012
${CALLERIDNUM}:3:3 returns 345
But this
You are using incorrect syntax. Notice where the close bracket is placed,
using your examples:
${CALLERIDNUM:3} erase first 3 digits
${CALLERIDNUM::3} returns first 3 digits
${CALLERIDNUM:3:3} should erase first 3 digits and return next 3 digits
Pisac wrote on Saturday, 14 January 2006 5:10
You need a FXS to FXO converter between ATA and the GSM box. You can get one from www.broad-tel.com
On 1/13/06, Ronald Voermans [EMAIL PROTECTED] wrote:
Hi All,
I have a GSM box, which needs to connect to a analogue phone line. I've plugged the GSM box to a Grandstream ATA (386). This ATA has
It's IVR system with complicated extensions.conf and ivr.conf (included
in extensions.conf), and I don't know which part of it to send here,
because IVR hangup in different places in different times (so it's seems
to me independent from what is written in extensions/ivr.conf, but I
belive it could
I have an x100p which suffers from echo (no surprise there apparantly
:), and a few of the things I've read about tuning out echo say the
first thing to get right is the tx and rx gain, and for that you need a
few different types of test lines from the telco (Telstra, Optus,
whatever). Does
Sorry, I use correct syntax in dialplan, but here in e-mail I maked this
mistake.
In dialplan I'm using ${CALLERIDNUM::3}
- Original Message -
From: Trevor G. Hammonds [EMAIL PROTECTED]
You are using incorrect syntax. Notice where the close bracket is
placed,
using your examples:
I maked mistake in my previous e-mail, but in my dialplan I didn't make
this mistake. So, my intention in previous e-mail was to write:
${CALLERIDNUM:3} erase first 3 digits
${CALLERIDNUM::3} returns first 3 digits
${CALLERIDNUM:3:3} should erase first 3 digits and return next 3 digits
So,
if
Alex wrote:
exten = _X.,1,AGI,catchall.agi,${EXTEN}
should do it for u
Hi,
since I also have some applications that starts with *, like
[app-clir]
exten = _*67.,1,SetCallerPres(prohib)
exten = _*67.,2,Goto(${EXTEN:3},1)
I thought I could use _. instead of _X., that would match only
Alex wrote:
exten = _X.,1,AGI,catchall.agi,${EXTEN}
should do it for u
Hi,
since I also have some applications that starts with *, like
[app-clir]
exten = _*67.,1,SetCallerPres(prohib)
exten = _*67.,2,Goto(${EXTEN:3},1)
I thought I could use _. instead of _X., that would match only
Hi,
does anyone have test numbers in Germany that would allow me to tune my
rxgain/txgain settings? I know there are numbers provided by other
providers in UK e.g. but have yet failed to find a number in Germany
(esp. by Deutsche Telekom).
Kind regards,
JP
this is the scenario:
One * is placed in a central location with more than enough up/down
bandwidth. One * is placed behind a DSL 3000/384. Both * are linked via
IAX trunking. Everything is fine until the upload channel of the remote
site is filled with a download, then heavy voice
On Sat, 2006-01-14 at 14:49 +0100, Pisac wrote:
I maked mistake in my previous e-mail, but in my dialplan I didn't make
this mistake. So, my intention in previous e-mail was to write:
${CALLERIDNUM:3} erase first 3 digits
${CALLERIDNUM::3} returns first 3 digits
${CALLERIDNUM:3:3} should
On Fri, 13 Jan 2006, Christian Peter wrote:
Hello List,
I'm trying to get a PrimuX Card (www.primuxisdn.de) working. The
Manufacturer says that chan_capi (the older one) used to work.
Now I'm trying with chan_capi-cm and have got the following problems:
Outgoing calls
does anyone have test numbers in Germany that would allow me to tune my
rxgain/txgain settings? I know there are numbers provided by other
providers in UK e.g. but have yet failed to find a number in Germany
(esp. by Deutsche Telekom).
Have you tried calling their repair number and asking
Rich Adamson ha scritto:
the problem appears no matter where I terminate the call (IAX or
Zap), and I don't have that problem on a 1.0.7 connected to the same
PRI lines and IAX servers , what I have to check ? looked in confif
files but appears to be the same (indications, modules loaded,
On Sat, 2006-01-14 at 08:25 -0600, Rich Adamson wrote:
does anyone have test numbers in Germany that would allow me to tune my
rxgain/txgain settings? I know there are numbers provided by other
providers in UK e.g. but have yet failed to find a number in Germany
(esp. by Deutsche Telekom).
Pisac wrote:
Sorry, I use correct syntax in dialplan, but here in e-mail I maked this
mistake.
In dialplan I'm using ${CALLERIDNUM::3}
Just for grins, have you tried
${CALLERIDNUM:0:3}
I have always found it better to explicitly specify what to do, rather than relying on a function's
Anyone have Mediatrix Unit Manager Express with the 1204 definitions handy?
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
http://www.techdatapros.com
On Saturday, January 14, 2006 3:36 PM trixter aka Bret McDanel wrote:
and some employees know what you are talking about and others (most?)
dont. The brightest people are usually working on problems so who
does that leave to answer the phones?
Actually I have not called them yet and I
it shouldnt be a problem from ILBC to g711u/a , but for g729 you need
a licence, otherwise no transcoding can ocurr. However does not seems
to be your problem, since the call should be hanged up, and you just
dont receive audio. That seems to me more like a problem with RTP not
finding a right
I have this setup:
(PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones
and a few of VoIP phones directly connected to Asterisk.
Calling a number (just one!) - an automatic responder (IVR) -
from VoIP phones works, from analog phones doesn't work:
NOANSWER after a few
Hi,
is there a way to 'manage' result of a call file (NOANSWER, BUSY, max
attempts, etc) put under /var/spool/asterisk/outgoing?
Thanks
Mimmus
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You could just route the call files through their own context, and using
some smart scripting, just write the output of DIALSTATUS to a file ..
thus it would only write to a file somewhere, when something happends in
that context.. keeping it separate from the rest of your dialplan.
Hi,
is
So,
if
${CALLERIDNUM}=0123456789
Then
${CALLERIDNUM:3} returns 3456789
${CALLERDINUM::3} returns 012
${CALLERIDNUM:3:3} returns 345
But this do not work anymore in 1.2.1, and if I do not found
solution for this I will downgrade to 1.0.9
Have you tried ${CALLERID(number)::3} ? I have a
Hi Dov. I cannot make guarantees because i did not test it, but you
could try applying this small patch i have made for you ;)
--- apps/app_voicemail.c.orig 2006-01-14 09:46:36.745550865 -0600
+++ apps/app_voicemail.c2006-01-14 09:53:28.635280667 -0600
@@ -1755,8 +1755,13 @@
I'm using Asterisk 1.2.1-BRIstuffed-0.3.0-PRE-1f
Your answer was helpfull, it's working now like it used before.
But I'm dissapointed with all this minor needless problematic
changes which needlessly spending my time. I will realy double rethink
in the future about upgrading any tuned system
I did exactly that.
If I don't load before modprobe wctxxp doesn't work properly.
Thanks,
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joel Vandal
Sent: Friday, January 13, 2006 11:19 PM
To: Asterisk Users Mailing List -
Yes, you are right, it's working.
Thanks.
- Original Message -
From: Steve Ringwald [EMAIL PROTECTED]
Pisac wrote:
Sorry, I use correct syntax in dialplan, but here in e-mail I maked
this
mistake.
In dialplan I'm using ${CALLERIDNUM::3}
Just for grins, have you tried
Hi there,
as far as I know (and some staff of German Telekom, who are really knowing
about they are talking) there is not such a public number you could call
for such a purpose. German Telekom has special owkrplaces Prüfplätze from
where such kind of tones can be supplied to a specific
Thank you for your response.
I found also this:
If the call is not answered, and the standard extension failed with
priority 1 exists in the same context, control will jump there.
in thw wiki: http://www.voip-info.org/wiki-Asterisk+auto-dial+out
M.
-Original Message-
From: [EMAIL
I upgraded from 1.0.9 to 1.2.1.
In 1.0.9 everything worked perfect.
Now, I call in my IVR, and after navigating in menus when I get dialtone
for dialing extension, Sound is choppy and I get bunch of messagess:
-- Silence suppression is disabled (option_silence_suppression=0
chan-timingfd=30)
--
Ok,
Now that I get zaptel working, is this loading something as safe_asterisk as
module?
Thanks,
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joel Vandal
Sent: Friday, January 13, 2006 11:19 PM
To: Asterisk Users Mailing List -
Yes,
This is the winner
It works. Really I'm loading wct1xxp, but still I don't know why also is
loading the wcusb with wct1xxp.
Thanks for the tip.
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Mountifield
Sent: Saturday,
I know there are numbers
provided by other providers in UK
If you happen to know of any, please feel free to post them. :-)
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons
___
Thanks .Find My replies in between your lines "Please note that recent IOS has SIP NAT traversal turned on by default.I believe that it only supports internal UA / external server.Since you also want the opposite, you should probably turn it off:no ip nat service sip tcp port 5060
On 1/14/06, Pisac [EMAIL PROTECTED] wrote:
I upgraded from 1.0.9 to 1.2.1.
In 1.0.9 everything worked perfect.
Now, I call in my IVR, and after navigating in menus when I get dialtone
for dialing extension, Sound is choppy and I get bunch of messagess:
-- Silence suppression is disabled
Thanks "If what you are trying to do is a SIP -- NAT -- Internet -- Nat -- Asterisk call them I'm afraid you would need to use a SIP/RTP router"Want more clarification on your last line call them . pls explain what you meant by call them who or what should i call.Expecting
This was too fast.
It works the first time.
After I did the make config for asterisk, especting to load it as module
(But before removing the modules that I don't use = everyone but wct1xxp)
It failed to load both wct1xxp and asterisk. Only zaptel, and at least 20
times the mpg123 ( due to all
Asterisk 1.2.1-BRIstuffed-0.3.0-PRE-1f
- Original Message -
From: BJ Weschke [EMAIL PROTECTED]
Where did you download this 1.2.1 version of Asterisk from? These
messages are coming from a patch to Asterisk that should not be in any
version of the 1.2 branch.
works with Asterisk.
I'm thinking I'd need 2 to support 6-8 lines - Or suggest some other
equipment that will provide up to 8 fxo ports and connect to asterisk.
for future projects I'd also like something with 2 fxo ports and 4 - 5
fxs ports - I suppose a digium card would do fine for 2 fxo
Yes you are probably right but I don't know how to rotate the fax in
fax2mail. I was hoping someone here on the list had to do it and would
post the solution :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter
aka Bret McDanel
Sent: Friday, January
I looks like someone decided to bundle a patch that
hasn't been merged yet. Good for testing, not so
good for initial impressions.
In /etc/asterisk/asterisk.conf add or uncomment this:
[options]
;silence_suppression=yes
And see if that helps. You need a timing source for it
to work, which is
It looks like there is a timing issue between zaptel load, and asterisk
load.
If I stop both services.
[EMAIL PROTECTED] ~]# service asterisk stop
Shutting down asterisk: Asterisk ended with exit status 0
Asterisk shutdown normally.
[
The script has two functions-
1. Once a minute check to see if any MeetMe conferences are
active and list the participants of any active conferences.
2. It registers an event_handler for MeetMeLeave and processes
the output.
The script simply loops issues manager commands. If command
fails,
If I stop the asterisk service, and only left zaptel on boot.
Zaptel loads but not wct1xxp or wcusb
[EMAIL PROTECTED] ~]# lsmod
Module Size Used by
ipv6 270753 10
autofs423621 2
zaptel193540 0
crc_ccitt 6209 1
I've found something here: http://bugs.digium.com/view.php?id=5374
but I don't understand how this can be connected to my problem :-(
- Original Message -
From: Pisac [EMAIL PROTECTED]
I upgraded from 1.0.9 to 1.2.1.
In 1.0.9 everything worked perfect.
Now, I call in my IVR,
It's helped.
Thanks!
- Original Message -
From: Dan Austin [EMAIL PROTECTED]
I looks like someone decided to bundle a patch that
hasn't been merged yet. Good for testing, not so
good for initial impressions.
In /etc/asterisk/asterisk.conf add or uncomment this:
[options]
On Friday January 13 2006 10:14 pm, James Harper wrote:
The best I can do so far (which appears to be a bit of a hack) is
(:0S0), which says to add a '0' to the start of the string and dial
immediately. This gives asterisk an extension dialled of '0', which
isn't the 's' that i'd hoped for,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Code Lover
Sent: Thursday, January 12, 2006 1:39 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Server Specification
Hello,
Is the hardware specification is enough to
oh323 0.6.7, asterisk 1.0.10. The problem is that h245 tunneling is not working
for outgoing calls. I tried all combination of h245Tunnelling, h245inSetup,
fastStart options in oh323.conf but the call signaling is allways the same.
Does somebody know what could be solution to this problem. I need
On Fri, Jan 13, 2006 at 09:39:09PM -0500, Carlos Alperin wrote:
That is right for zaptel. But you still has to do modprobe wctdm on rc.local
before to load asterisk.
rc.local is run after the standard init.d scripts. Thus if you load
asterisk in an init.d script, you'd be loading the zaptel
i suffer the same double ring tone on our 1.2.1 box. 1.0.9 box ha no such
problem. i used the r to solve it but this is not a good solution though.
Best Regards
matt
__
- Original Message -
From: Simone Cittadini [EMAIL PROTECTED]
To:
After install everything on the supposedly right place, my conclusion is
that zaptel doesn't load wct1xxp module.
Then, that is the reason for Asterisk to fail loading.
However I change the MODULES RMODULES on the zaptel on /etc/init.d
/etc/sysconfig, it continuous same way.
Carlos Alperin
Can anyone recommend a tool that can be used on Windows XP to configure the
Mediatrix 1204?
-Kerry
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On Sat, Jan 14, 2006 at 03:55:50PM -0500, Carlos Alperin wrote:
After install everything on the supposedly right place, my conclusion is
that zaptel doesn't load wct1xxp module.
Then, that is the reason for Asterisk to fail loading.
Could you please provide:
/etc/zaptel.conf
Of course,
/etc/zaptel.conf
[EMAIL PROTECTED] etc]# cat zaptel.conf
# Autogenerated by ./genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# It must be in the module loading order
# Span 1: WCT1/0 Digium Wildcard T100P
Never mind, I found out that the latest 1204 SIP versions actually have a
web-based GUI now. The lack of documentation for this is mind boggling.
Still not working yet, but at least I am in the config system now.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I isolated problem, but I cannot find a cause. I think this is a bug!
So, there is very very simplified dialplan which working in 1.0.9 but in
1.2.1 have that unexpected hangup:
;-
exten = s,1,answer
exten = s,2,digittimeout(0)
exten =
On Sat, 14 Jan 2006 11:22:51 -0600, Tim Litwiller wrote
works with Asterisk.
I'm thinking I'd need 2 to support 6-8 lines - Or suggest some other
equipment that will provide up to 8 fxo ports and connect to asterisk.
for future projects I'd also like something with 2 fxo ports and 4 -
5
The Aastra VentureIP system used a semi proprietary, non SIP protocol. I do
not think it would integrate with Asterisk very well.
Cory J Andrews
VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
++
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
AIM - B2CORY
-
Does anyone know how to determine the echo tail size (in ms) of a particular call?
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That would be called a milliwatt generator. It likely exists in their
central office, but its typically used by their technicians to ensure
new installations meet specs and sometimes in troubleshooting. Call
your
telco repair number and see if they will give you the telephone number
for it.
Do Asterisk support Advice Of Charge (AOC) on ISDN lines?
Do any ISDN drivers (bristuff, capi, vISDN, mISDN) support AOC?
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In article [EMAIL PROTECTED],
Pisac [EMAIL PROTECTED] wrote:
I've found something here: http://bugs.digium.com/view.php?id=5374
but I don't understand how this can be connected to my problem :-(
It looks like the maintainer of the BRIstuff distribution might have
decided that patch was worth
welcome to mediatrix hell.
Aparently they are supposed to be good once you have them working.
clear skies!
-Original Message-
From: Kerry Garrison [mailto:[EMAIL PROTECTED]
Sent: Saturday, January 14, 2006 4:01 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
Im working on some code to be able
to preload the echo cancellers (and obviously dump the data too, which Ive
already done). I should have a patch ready tonight or tomorrow.
If you are interested I can attach a plot
of the coefficients against time which might tell you the sort of thing
Yes, plese do post itOn 1/15/06, James Harper [EMAIL PROTECTED] wrote:
I'm working on some code to be able
to preload the echo cancellers (and obviously dump the data too, which I've
already done). I should have a patch ready tonight or tomorrow.
If you are interested I
Heres the graph of the echo coefficients I
grabbed from a x100p card on my asterisk server. If my
interpretation is correct, it shows that most of the echo comes in at about the
28th tap, and assuming a sample rate of 8000hz,
that would be about 3.5ms.
Will that tell you the sort of
Wouldn't that large spike be the primary sound rather than the echo?On 1/15/06, James Harper [EMAIL PROTECTED]
wrote:
Here's the graph of the echo coefficients I
grabbed from a x100p card on my asterisk server. If my
interpretation is correct, it shows that most of the echo
You need the unit manager software that should have come with your box.
Your box most likely only speaks SNMP, so this is the only tool I know
that has the MIB's and setup to know how to set the MIB values. However
there are many more tweaks in manually tuning some of the MIB's through
the
To agree with you - I don't remember what the impedence is in Australia, but
it isn't 600 ohm.
PaulH
On Sunday 15 January 2006 10:16, James Harper wrote:
That would be called a milliwatt generator. It likely exists in their
central office, but its typically used by their technicians to
I'm using Sipura 3000 as well, however I will have to wait until
Monday about the Switch I'm not sure. So far it looks like Sipura is
at fault. In the mean time I would like to hear from others using the
Sipura 3000 FXO if they have the same problem.
On 1/13/06, Thczv F. Thczv [EMAIL PROTECTED]
Kind of, I think the various ec algorithms train differently, but they end up with a
list of numbers which represent how much of the primary sound to replay (inverted
of course) and when, so in my case the ec needs to take
the current sample, and subtract the value of the sample 3.5ms ago
According to this page: http://www.asterisk.org/doxygen/Config_sip.html
canreinvite=yes redirects just the RTP. I was
under the impression that the entire SIP connection got redirected, therefore
losing accounting ability. Could someone clarify this?
--Mike
Rich Adamson wrote:
Since there does not seem to be anyone else complaining about the same
problem, there must be something in your config that is causing it.
Without specific copy/paste samples of what you've configured, no one
is going to be able to guess at what you are doing.
Given the
The new versions of the 1204 SIP version have a web interface now. Took me a
few minutes to figure out the login/password, but once I got in, it looked
pretty simple. My only problem is that I have outbound working fine but
inbound calls are not working.
-Kerry
-Original Message-
milliwatt generator. (Obviously its not as good as using a CO
milliwatt
as now you have to take into consideration the loss from the second
pstn line, but it is a way to get a handle on the transmission loss
values, etc.)
Would the txgain on the 2nd line also come into play? I guess
Thanks for the heads up - I didn't see anything that said it did work
with asterisk so I thought I better ask.
So if you where setting up a 6 - 8 telephone line system with 10 - 12
phones and trying to stay under $3000 for the system and phones what
would you suggest. It sounds like if I
I'm using bristuffed Asterisk with ISDN/ZAPHFC
I have VERY ugly (outgoing) sound through ISDN/HFC if echocancel=yes in
zapata.conf, but without echocancel I have bad (incoming) echo
Through PSTN/FXO sound is ok with or without echocancel.
I tried other echo cancellers (in zconfig.h) two times:
I'm using zaptel on a x100p card, so my $0.02 may be apply, but I found
that if I said echocancel=256 (256 taps), I would end up with horribly
distorted outgoing audio (started off okay, but with bad echo, and after
about 20 seconds most of the echo was gone but the audio was almost just
noise).
You need hardware echo cancel card such as sangoma 104d card to do hardware
cannel to fix the bad echo problem. Software solution is not powerful enough
at all.
Best Regards
Matt
High Performance Gigabit Clustering Appliance
You could do an AudioCodes, Mediatrix, or Vegastream 8FXO SIP gateway, and a
dozen SPA-941's and still stay under your $3K budget.
Cory J Andrews
VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
++
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
AIM - B2CORY
-
Hello everybody,
I am sorry to bring this up again if this
kind of echo issue has ever discussed.
Phone2 in below call path experiences quite
annoying echo:
Phone1 -- FXS (TDM400P) -- Asterisk
-- SIP GW -- PSTN -- Phone2
It is annoying as on phone2, we can hear
the whole words we say
Reinvite doesn't happen until after the call is picked up. After it's picked
up, new invites' are sent and the phones communicate directly. Sorry, I forget
the details. It was a few weeks ago.
Doug
-Original Message-
From: Mike Hammett [mailto:[EMAIL PROTECTED]
Hi all,
I have a Cisco 1760 gateway and and Cisco 7960 VoIP phone running via
Asterisk. Both are running g711A codecs and SIP. On inbound calls I get
a native bridge, however on outbound calls I never get a native bridge.
With other SIP gateways I do get a native bridge on the outbound call.
My
I'm guessing that you have a similar entry in your sip.conf for the
7960?? The 7960 has a setting for preferred codec. It defaults to g711
U-Law. You might try changing this setting also as the 7960 doesn't
know that you only want to speak A-Law. You will also want to make sure
that the nat
Yes the 7960 is also set only to use alaw. I was under the impression
though that nat=yes did not effect this. And if it does why does it
native bridge ok on inbound calls with the same nat=yes
On 1/15/06, Jonathan Feally [EMAIL PROTECTED] wrote:
I'm guessing that you have a similar
8 lines for 10 phones is overkillreally
PaulH
- Original Message -
From: Tim Litwiller [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, January 15, 2006 2:38 PM
Subject: Re: [Asterisk-Users] I need feed
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