Hi Mohamed,
See at [EMAIL PROTECTED] and http://app-rpt.qrvc.com
Best Regards,
F6HQZ
Francois BERGERET
France___
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Hi,
Check if you have a ground loop.
If yes, this is probably the cause of this hum.
Open the loop.
Best Regards,
Francois BERGERET
France
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Yes, Jay and Philip, you are right, but you can also have hums if ground
cables for chassis protection against electrical hazards are making loops,
if certain of them are in parallel and if they have different length between
the equipments to protect. Many audio stages (unbalanced side, not
Hi men,
What happens after restarted xinetd ?
Only one Eth access again or suddently the two ?
Francois
-Message d'origine-
De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] la part de Jerry Geis
Envoyé : lundi 28 avril 2008 20:33
A : asterisk-users@lists.digium.com
Objet : Re:
BERGERET,
F6HQZ
www.hamwlan.net
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Hi the list,
I am using Kirk DECT/SIP 600V3 every day.
This system run very very well behind an Asterisk, with transfert feature,
caller ID display and so...
Seen as an IP-Phone running a separate SIP account for each handset.
Consider the 600V3 server as a mediagateway converting DECT to SIP.
I
Hi Bilal,
Have you done ./configure in the zaptel directory before to do make
menuselect ?
Best Regards,
Francois BERGERET,
France.
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Oops !
Not seen that you have already done ./configure because I have not read
your message until the end.
Sorry !
What was the output after that ?
Francois
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asterisk-users
I must take more cofee !
Sory to have replied to the wrong thread.
Francois
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] la part de F6HQZ
Envoye : samedi 23 juin 2007 22:38
A : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] la part de F6HQZ
Envoye : samedi 23 juin 2007 22:38
A : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] TAS test equipment manuals
Oops !
Not seen that you have already done
Hi the list,
600v3 with last firmware works fine with Asterisk and SIP.
I use it every days with success, no issue.
I recommend it and think it's more reliable than WiFi for a great number of
handsets or industrial deployment with multicells.
Best Regards,
Francois BERGERET,
France
-Message
Hi men,
Resolved for one of my customers by upgrading Asterisk/Libpri/Zaptel.
I don't remember what wer the versions, sorry.
Check and advise us the results, please.
Best Regards,
Francois
No virus found in this outgoing message.
Checked by AVG - www.avg.com
Version: 8.0.233 / Virus Database:
Hi Men,
A little old now, but certainly one of the biggest worldwide Asterisk's network
:
The AUF : Agence Universitaire de la Francophonie (in french, of course).
http://wiki.auf.org/wikiteki/Asterisk
http://wiki.auf.org/wikiteki/Projet/VoIP
Best Regards,
Francois BERGERET
France
Ce message
Hi,
Excellent ! For me, Polycom have the best audio.
Just behind, I like also Aastra.
Best Regards,
Francois
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] la part de Eric Jacksch
Envoyé : lundi 12 novembre 2007 15:39
A : Asterisk Users Mailing List -
Hi,
I suspect that you are transcoding, meaning that the call is comming in a
specific codec format, and the second phone uses another codec. So, when you
do your tranfert, Asterisk is in the middle and is coding from the original
to your phone with two different codecs. If you are passing from
Regards,
Francois BERGERET
F6HQZ
France
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] la part de Gustavo
Cordeiro
Envoyé : vendredi 23 novembre 2007 12:20
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] TDM808B 8 port
Hi,
Your extension 100 doesn't exist in the context where you have your PickUp
instruction.
You must include the context containing your phones into the context used by
your PickUp instruction or the reverse, or precise the context to use with
PickUp by adding it into the instruction line :
Hi men,
And what happens without APIC/ACPI ? I hate them ! Any IRQ sharing issue ?
Francois
-Message d'origine-
From Loic Didelot
...SNIP...
cat /proc/interrupts
CPU0 CPU1
0: 83 0 IO-APIC-edge timer
1: 2 0
Hi Francois,
Here is Francois too. :-)
Why to not ask to your Digium card provider ?
Example for Zaptel (fortunately same for Dahdi) :
via Linux console and dmesg :
Registered codec translator 'DTE Encoder' with 92 transcoders
(srcs=000c, dsts=0101)
Registered codec translator 'DTE
Hi Dhaval,
Echo depends of the far end not directly Asterisk or Digium cards.
If the remote telephone or PBX return your your voice, you will ear your echo.
If the remote don't return you your audio signal, no echo.
The passedthrough circuit along the complete path can also return you echo but
exten = _8XX,1,Dial(${SIPPROVIDER}/${EXTEN:1},,G(fax-tx^send^1))
This dial command line call a FAX number through a SIP provider and, when
answered, give the hand to the macro who has in charge to realy send the
fax.
Good luck !
Best Regards,
Francois
-Message d'origine-
De
feature.
Mine is closed...
Friendly yours,
Francois
F6HQZ
Ce message sortant est certifie sans virus connu.
Analyse effectuee par AVG - www.avg.fr
Version: 8.5.409 / Base de donnees virale: 270.13.89/2359 - Date: 09/12/09
06:37:00
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Hi Men,
I believe that .T is anything + a Time out of (probably) 3 sec. before to
dial the complete called number.
Best Regards,
Francois
destination-pattern .T
What does destination-pattern .T mean? I'm not familiar with what
.T would match. I would suggest using a more specific pattern
Hi Michael,
It does what it is announced/supposed to do.
I have checked and know well all the Portech GSM/SIP family.
But, be carefull, because under the same reference you can buy/receive
different hardware versions :
- 2, 3 or 4 GSM frequencies bands
- Siemens or Simcom GSM modules
So, the
Hi Francois,
here is Francois too ;-)
Check that :
[fax-outbound-calls]
exten = _X.,1,Dial(${ACROPOLIS}/${EXTEN},,G(fax-tx^send^1))
[fax-tx]
exten = send,1,NoOp( SENDING FAX )
exten = send,n,Set(FaxTxDir=/var/spool/fax/tx/)
exten = send,n,Set(FAXFILEPDF=fax-${FAXCOUNT}-tx.pdf)
exten =
(echo Entete FAX : ${ARG6} - ${ARG4} pages - Rate:${ARG5}
- CID:${ARG7}, Resolution : ${ARG8}|/bin/mailx -s
FAX de : ${ARG6} - CID : ${ARG7} -a ${ARG3} -r ${ARG2} ${ARG1})
exten = s,n,NoOp( SENT )
exten = s,n,System(rm ${ARG3})
-Message d'origine-
De : F6HQZ [mailto:f6hq
Ah ! It's a jamming of Digium FFA user manual, ideas and tests from my
customers and myself.
From Digium's side you can/must acces to this WEB page :
https://www.digium.com/en/supportcenter/documentation/viewdocs/FAX
I love to check Digium's solutions and to know how to use them.
So, I have
Hi Aditya,
Are you installing under ROOT ?
Best Regards,
Francois
-Message d'origine-
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]de la part de Aditya Kumar
Envoye : vendredi 25 decembre 2009 07:50
A :
Hi Carlos,
It's simply not possible due to a firmware limitation when general SIP and not
Aastra proprietary mode (not enougth memory capacity).
Don't lack your time by searching a non exisiting solution.
Best Regards,
Francois
-Message d'origine-
De :
Hi Daniel,
Are you using a demo/beta version of Skype for Asterisk ?
If yes, this status is normal, the demo/beta program is terminated from a
while.
I am using the real commercial (not free) and not getting that message.
Best Regards,
Francois
-Message d'origine-
De :
Hi men,
I am sure this is the demo version, not the correct actual licensed one.
Fro mthe CLI, enter that :
fax show version
My Asterisk reply that :
Fax For Asterisk Components:
Applications: 1.6.1_1.0.15
Digium Fax T.38 Driver: 1.6.1_1.0.11 (optimized for c3_2_32)
Hi,
It seems that you perharps have an IRQ sharing issue or a motherboard or its
BIOS incompatibility.
Check again by desabling APIC/ACPI features.
You can do that by editing the file /boot/grub/grub.conf.
Add acpi=off noapic quiet at the end of the line starting with kernel, and
reboot the
) HAM radio call sign and position (and
some extra info as time, temperature and so...).
You can read more about Asterisk and HAM radio here :
http://www.zapatatelephony.org/
and here :
http://app-rpt.qrvc.com/
Best 73's from F6HQZ Francois (France)
Le 25/02/2010 19:45, Chris Kairalla a écrit
Title: Message
Hi
Mike,
You
must continue - for zaptel only- to "make linux26", as it is described in
the companion file "README.Linux26" in the Zaptel folder
(/usr/src/zaptel).
Read
the text from this file, as suggested inits
title:
To
build for Linux 2.6, first you must be sure
Buy a TDM2400P card with several quadFXO modules : 24 ports max :-)
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de roswel ajf
Envoyé : vendredi 27 janvier 2006 23:17
À : asterisk-users@lists.digium.com
Objet : SPAM: [Asterisk-Users] fxo/fxs cards with
Hi Harry,
How many IRQ do you have ?
Be carefull for power supply is it is several TDM2460E (all FXS ports) !
It is better to use a seconf power supply...
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
[EMAIL
A PICMG card equiped with Pentium M CPU permits to reduce dragsticaly the
power consumption.
As this, you can use more power from your PSU for the interface cards.
But, for several TDM2460E/B cards with a heavy traffic charge (many
simultaneous rings), I believe that it could be better to use a
Hi Dave and the list,
I was at this exhibition near all the first day.
Next time, we must organise a meeting for handshaking and discussions for
Asterisk lovers during a next exhibition ?
Just by saying hello and what's happening to the list ?
It could be cool to meet us in real world ;-)
Have
Argh ! Failed meeting with you ! Sorry !
Sure, Asterisk must be more present to this kind of exhibition.
What could be the next french popular show at low price booth ?
May be Mark could be there during it if we ask him ?
Any idea of scenario or presentation ?
-Message d'origine-
De :
Interested, of course, but may be we can do that for a nearer exhibition
this year ?
Francois BERGERET.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Dave Cotton
Envoyé : vendredi 3 février 2006 13:54
À : Asterisk Users Mailing List - Non-Commercial
Hi the list,
I can confirm you that I have not noticed any echo issue in this
configuration (analog phones on quadFXS modules AND analog lines on quadFXO
modules) at the same place and Asterisk when some echo issues occured with
IP-Phones.
TDM2400E is an excellent choice :-)
Best Regards,
Hello,
I have an IAX2 trunk like this running well with IAX2 and SIP users mixed at
each side.
Runing like a charm :-)
Don't forget to add username definition from this example.
To avoid too much load for your CPUs with transcoding, tempt to have only
the same CODEC choice for all phones and
Hi,
I have good results with the new TDM2400P serie (with the hardware echocan,
of course).
May be you must check one TDM2401E to see if it's ok for you...
Good luck.
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la
Title: Message
Hi,
I
believe that you have inverted fxo_ks and fxs_ks into your zapata.cong file
"signaling=" declaration...
Invert
and redo the tests.
Good
Luck !
Francois BERGERET,
France.
-Message d'origine-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part
Hello Cosmin,
This is extract from my zapata.conf :
busydetect=yes
busycount=3
busypattern=500,500
Check how is your local busy pattern for more efficiency.
Good luck !
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
Hello the list,
Be carefull to have this rule available at begining of your rules list,
because shorewall use the first one matching and stop to check the
following. If you have another with a range including this UDP 4569 DNAT
before your new one (as UDP 1024 to 65535 for example), it could
Hi Chan,
1/ be sure to have correctly inputed your country zone
2/ disable the fax recognition in zapata.conf
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de chan (Alpha
Trilogies Networls)
Envoyé : lundi 27
Hi gentlemen :-)
I am searching a radio base GSM or DECT with high power for long range, and
the terminal units (handy).
This equipment must be connected to a T1 port from an Asterisk.
The number of simultaneous channels must be 7 to 10.
Do you know a manufacturer with nice equipments at
Hi Pascal !
France is not more difficult than other country.
This is one of my channels behind France Telecom :
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
Hi everybody !
I never use any prefix number to dial out.
I prefer to do like any standard residential subscriber, not to force
somebody to think : Oh no ! I have forgottent to input the 9 - or 0 -
before to dial out !.
Directly inputing the real number is more natural.
Adding a prefix is an old
Hello again,
They are many Asterisk servers outside of the US that use a different
national plan...
Here, in France, we are using _0Z for fixed national telephones
lines, including _06 for national mobiles, _08 special
(often higher) price calls, _00Z. For international
Hi Christian,
Increase a variable in the menu loop, or exactly in the t and i
extensions like this :
exten = s,1,Wait(3)
exten = s,n,Answer()
exten = s,n,Set(LoopStep=1)
exten = s,n,Set(TIMEOUT(digit)=3)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Wait(1)
exten =
Hi Alexander and the list,
Have you well checked your E1 cable ?
Sometime, you must use a crossed E1 cable (not an Ethernet one)...
Check also without the crc check.
How is your zapata.conf file ?
Have you checked with a loop (crossed E1 cable) between two spans (one in TE
the second in NT, of
Autocorrection mode :
pri_cpe / pri_net rather than TE / NT ;-)
-Message d'origine-
De : Francois BERGERET [mailto:[EMAIL PROTECTED] De la part de
'[EMAIL PROTECTED]'
Envoyé : jeudi 3 mai 2007 21:03
À : 'Asterisk Users Mailing List - Non-Commercial Discussion'
Objet : RE :
Hi Gavin,
A second Asterisk server replacing the provider (best way), or doing a loop
between two different ISDN ports on a same card (worst way) must help you.
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
Hi Zeeshan,
Ethernet Network (or Switch) congestion ?
QoS not realy effective ?
Too high CPU load in Asterisk the server ?
Who knows...
You must check during a default.
Good kuck !
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De
Hi Farook and the list,
You have may be forgotten to input that in the misdn.conf file :
nationalprefix=0
internationalprefix=00
dialplan=0
localdialplan=0
cpndialplan=0
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la
Hello Aslay,
In some country, this feature is a paid option from the TELCO side.
In France the analog lines have not this feature enabled in standard, only
the digital lines .
Are you sure that it's actualy available in your case ?
Best Regards,
Francois BERGERET,
France.
-Message
Check without the echocan module (remove it) if any 'crackle is listen
again.
If yes, the echocan is not faulty.
If yes, check another echocan module temporary.
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Ed
Hi men,
I have already encountered some issue like this with few switches (very
known great brand) which doesn't like VoIP traffic !
Check by drectly connected the VoIP equipment - if you can - with temporary
long Ethernet cables bypassing the tested switch to see what happens in this
case.
You
Hi David and the list,
It's normal ;-)
Near all European BRI operators cut off the line between calls. So, you must
trieve the correct parameter avoiding to survey the line as for mISDN :
pmp_l1_check=no
I use mISDN without any issue with B410P.
I hope this help.
Best Regards,
Francois
Have you taken care of any eventual IRQ sharing ?
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Edoardo Serra
Envoyé : samedi 24 mars 2007 20:27
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] Re: RE : SIP/IAX peers UNREACHABLE and audio
Hi !
Prefer to have only one card with how many ports you want.
Always better for IRQ flow.
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Farooq Ahmed
Envoyé : lundi 26 mars 2007 09:11
À :
Hi Pierre and the list,
I have the habit to do like this after having compiled Zaptel and Libpri :
cd /usr/src/
wget http://www.misdn.org/downloads/mISDN.tar.gz
wget http://www.misdn.org/downloads/mISDNuser.tar.gz
tar xzf mISDN.tar.gz
tar xzf mISDNuser.tar.gz
cd mISDN-1_1_1
make
Hi the list,
Think Kirk solution ;-)
www.kirktelecom.com
This is an DECT/GAP infrastructure solution, and the bases can be seen as
something like SIP/DECT gateways.
Each wireless phone is like a separate IP phone from Asterisk side.
You can use several bases and repeaters (only radio link, no
Hi Tobias and the list,
Yes, I have, I use and sell them to integrators ;-)
But only the 600v3 family, not the older ISND or analog versions, and the
current DECT handsets 40XX.
Any Digium interfaces run well with them as any SIP IP-Phone, of course.
The sound quality is GREAT and the
Or a new Digium TDM880B replacing the old TDM40B for only one IRQ...
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Jim Freeze
Envoyé : lundi 9 avril 2007 15:15
À : Asterisk Users Mailing List - Non-Commercial
Hello Mark and the list,
What about if you change the order of the modules, starting with FXS first
and finishing with FXO on the TDM400P slots ?
I remember to have read something like always start with FXS if FXS and FXO
modules are present on the board...
Feedback please.
Best Regards,
Eric, contact me off list and I will give you a nce exemple with a worldwide
Asterisk network ;-)
Francois BERGERET,
France.
f6hqz-m_at_hamwlan.net
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Eric
ManxPower Wieling
Envoyé : jeudi 14 septembre 2006
Hello,
RING 1 26 TIPfirst Zap channel
RING 2 27 TIPsecond Zap channel
RING 3 28 TIPthird Zap channel
etc..
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de C F
Envoyé : mardi 3
Hello Matt,
I have not seen how to add a site.
Could you help me (us) ?
Tks
Francois Bergeret,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Matt Riddell
(IT)
Envoyé : vendredi 6 octobre 2006 11:40
À : Asterisk Users Mailing List -
IpSec VPN ;-)
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de John Melody
Envoyé : lundi 30 mai 2005 10:37
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] IAX encrytion
What encryption features
Hi Roywish,
The best way is to publish here your .conf files to correct.
Good luck...
Best Regards,
Francois BERGERET,
Happy * french user :-)
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de craz sead
Envoyé : mercredi 8 juin 2005 09:45
À :
Hi guys !
Correct zaptel modules are probably missing, as wcfxo.
He must recopile zaptel on his Asterisk machine.
Best Regards,
Fracnois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Sebastian
Silva
Envoy : mercredi 15 juin 2005
Hi all the list,
I am searching how to insert few seconds of silence just before to send the
DTMF sequence via a FXO WildCard X101P to PSTN.
I remember that Hayes compatible modems knows a special character W that
do a 1 sec pause.
Is it possible to do something like this in DIAL line sequence ?
Hello the list, hello Doug,
Thank you, but I don't see any correct reply in this page.
I want to have a silent header of 1 or 2 seconds between to pick up the
line and before to start to sned the DTMF numbering, because my RTC provider
doesn't give the prompt tone or listen the DTMF before this
Eureka ! It's working now !
The solution was in my question, in fact. So stupid I am ! :-)))
Many thanks to you guys, working around this fantastic project, and specialy
to Eric Wieling for his nice help.
Have a Nice Night !
Merci camarade !
Best Regards,
Francois BERGERET,
France.
Yes, I have :-)
3 of this cards running well on my personnal *
What price for your ?
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Sandy Thomson
Envoyé : vendredi 1 juillet 2005 12:37
À :
Hi men,
If you unplug a telephone line behind X100P/X101P, you must have an alarm
about this Zaptel device on the Asterisk console.
When you plug the line, you can see alarm stop on a new line (I don't
remember the exact message).
Very nice test.
Best Regards,
Francois BERGERET,
France.
Asterisk -gc
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Stefan
Gofferje
Envoyé : samedi 2 juillet 2005 22:25
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [Asterisk-Users]
Hi the list !
I totaly agree Tzafrir.
I am an happy Debian user from a while now.
I recommand to my friends to use Testing branch to have the latest packages
(which are stable or near to be stable but usable).
Asterisk is delivered closed to be complete Asterisk
1.0.7-BRIstuffed-0.2.0-RC7k with
Hi the list,
Searching to start ztmonitor in quantitative mode rather than graphical.
I want to read the real voltage (RMS) or dBm on my analog telephone lines..
TIA
Best Regards,
Francois BERGERET,
France.
___
Asterisk-Users mailing list
Hi the list,
ztmonitor 3 -v start ztmonitor in graphical mode on Zaptel device #3.
What is the correct syntax for dBm or voltage ?
TIA
Best Regards,
Francois BERGERET,
France.
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Hi Rich,
Thank you for this response.
Strange... I have read something about this, but probably misunderstood :
http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html
3. Start 'ztmonitor' on the target trunk in 'quantitative' mode.
4. Dial the CO Milliwatt test line from
Thank you for our point of vue, Rich :-)
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Rich Adamson
Envoyé : mardi 5 juillet 2005 14:40
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: RE :
Title: Message
Hello,
Have
you optimized by chosing the correct CPU and see for MMX support before to
compile Zaptel and Asterisk ?
What
is your server cofiguration ?
How is
its load ?
How
many simultaneous calls ?
Etc...
All
litle details which can help to consider and understand your
Hi Asterisk's people,
You can buy Digium's card harware echo can models without the echo can
module and buy it later if necessary.
They are scalable ;-)
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
Hi,
Check Chinese IP-Phones with PA1688 chipset : IAX2/SIP/H323/MGCP/Net2Phone +
all the main codecs !
Good Luck !
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de WipeOut
Envoyé : vendredi 17 mars 2006 13:11
À : 'Asterisk
Hello,
As I have said earlier in the list, take a look at Chinese IP-Phones with
PA1688 chipset : IAX2/SIP/H323/MGCP/Net2Phone + all the main codecs !
Good Luck !
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Joe Hood
Check for :
dtmfmode=outband
Good luck !
Francois BERGERET,
France
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Chris Mason
(Lists)
Envoyé : samedi 18 mars 2006 17:43
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet :
Hello Fernando,
I have checked this card with and without hardware echocan : the hardware
echocan module does the job better than the zaptel software can do it. I
recommand this module without any doubt.
But, the echocan algorithms in zaptel are better and better and the CPUs
power grows
Of course, but if newbies are separated and together only without any
expert, who can explain them anything ?
I am actualy a subscriber for all the Digium lists. If more lists will be,
more subscribtions I will get and I will receive the same quantity of
messages ;-)
Francois BERGERET,
France.
Oops !
I have upgraded TRUNK again via SVN and all was seeming to be fine, no more
invalid IAX2 frames and able to place and receive calls.
I was happy..
But, few calls later (about 5 minutes) : INVAL frames again and no more
possibility to place or receive calls, no prompt tone, nothing !
Hi,
Jump to a TDM2402E for 6 POTS lines with hardware echocan.
Only one IRQ used, and easy future extensions by adding modules.
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Jared Davison
Envoyé : vendredi 24
Title: Message
How
many phones lines ?
-Message d'origine-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Curt
ShafferEnvoyé: vendredi 24 mars 2006 03:17À:
asterisk-users@lists.digium.comObjet: [Asterisk-Users] FXS
channel banks
Is anyone out
smime.p7m
Description: S/MIME encrypted message
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2 x TDM2460E (with hardware echocan module) or TDM2460B (wo/echocan) = 48
phones lines, no T1 cards, no channel banks level adjustments troubles,
direct Zap channels and simple switching.
Probably the best choice and price :-)
Best Regards,
Francois BERGERET,
France.
A very happy TDM2400 user
This card doesn't permit to support Mark Spencer's company and project.
This card has no hardware echocan and use only the X100M and S110M clones
modules.
This two reason are sufficient for me.
-Message d'origine-
De : Krzysztof Drewicz [mailto:[EMAIL PROTECTED]
Envoyé : lundi 27 mars
Hi,
zap show channel 5
To see channel 5 specs, and take a look at Echo Cancellation: 128 taps
unless TDM bridged, currently OFF during calls, you must have ON.
If you have hardware echocan module, as for TDM2400E, you must also read
DSP: yes if this module is active.
Best Regards,
Francois
Hi John,
If you enter show application dial when logged into the Asterisk console,
you can read that help (extract only regarding dial option) :
L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are
left. Repeat the warning every 'z' ms. The following special
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