) 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
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
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 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 :
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 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 ${ARG3})
-Message d'origine-
De : F6HQZ [mailto:f6hq
Hi Michael,
It does what it is announced/supposed to do.
I have checked and know well all the Portech GSM/SIP family.
But, be carefull, because under the same reference you can buy/receive
different hardware versions :
- 2, 3 or 4 GSM frequencies bands
- Siemens or Simcom GSM modules
So, the
Hi Men,
I believe that .T is anything + a Time out of (probably) 3 sec. before to
dial the complete called number.
Best Regards,
Francois
destination-pattern .T
What does destination-pattern .T mean? I'm not familiar with what
.T would match. I would suggest using a more specific pattern
feature.
Mine is closed...
Friendly yours,
Francois
F6HQZ
Ce message sortant est certifie sans virus connu.
Analyse effectuee par AVG - www.avg.fr
Version: 8.5.409 / Base de donnees virale: 270.13.89/2359 - Date: 09/12/09
06:37:00
___
-- Bandwidth
exten = _8XX,1,Dial(${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
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
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 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 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,
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:
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 :
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,
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
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 -
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,
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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Hi Mohamed,
See at [EMAIL PROTECTED] and http://app-rpt.qrvc.com
Best Regards,
F6HQZ
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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
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 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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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
BERGERET,
F6HQZ
www.hamwlan.net
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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
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
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 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 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 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
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
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
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 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 !
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 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
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 Mark and the list,
You can switch to Industrial PC.
I mainly use PICMG 1.0 standard Single Board Computer cards and passive PCI
busses with success.
They are 5V PCI bus and maintained for ten's years as industrials want.
Perfect for IPBX with a long live or MTBF.
I prefer Pentium-M equiped SBC
Hello !
Shared IRQ ?
Very old tired CPU ?
No echocan module on the TDM2400 (echocan Zaptel solution claims more
motherboard CPU power) ?
Not enougth RAM ?
Not CPU optimized compilation with 1.2 ?
Please describe more your server and Asterisk version...
Best Regards,
Francois BERGERET,
I wish many stars in your blue sky for this new year :-)
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Sam Tam
Envoyé : dimanche 31 décembre 2006 19:19
À : 'Asterisk Users Mailing List - Non-Commercial Discussion'
Objet : RE: [asterisk-users] Happy
Hi Josué,
Have you checked the strap on the TE110P board ?
You must have it on the E1 position, not T1 (open ?, I don't remember at
this hour, sorry).
Check also without crc4.
And recheck ztcfg -vvv.
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
Hello the list,
You can use FXS and em signalling to reverse the line polarity temporary to
trigger an external door opener interface.
This is very easy.
Good Luck !
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part
Hi men,
Have a look at : www.asterisknow.org
This will be THE standard !
Best Regards,
Francois BERGERET,
France.
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Hi Marty,
I have checked/played FXS ports behind Asterisk with success and checking
now a new firmware for FXO one stage (normaly two stages). All this gateways
have the same manager unit and parameters suite, looking like Cisco models.
It's normaly easy to use if you have trained for Cisco (only
Hi the list and Marty,
Take a look to www.aliwei.com.
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Martin Joseph
Envoyé : lundi 4 décembre 2006 20:47
À : asterisk-users@lists.digium.com
Objet :
Hi the list,
You must input extensions using the 5 (may be 4 in some countries) last
digits representing your telephone number end for this BRI line in your
current ISDN calls incoming context.
Open the ISND debug mode and see what is on your asterisk console screen
when a call comes.
That's
Hello,
All the biggest gateways manufacturers do that.
Search for Aliwei, Audiocodes, Patton, etc...
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Noc Phibee
Envoyé : lundi 30 octobre 2006 20:51
À : Asterisk
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 -
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
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 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,
Hi Stephen,
+99 ms via IPSec FreeSWan
But good protection and no NAT issue.
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Stephen Bosch
Envoyé : mardi 25 juillet 2006 17:25
À : Asterisk Users Mailing List - Non-Commercial
Hi Franck,
NOACPI and the sound must be more clear.
And, of course, have you tell to /usr/src/zaptel/zconfig.h and
/usr/src/asterisk/Makefil what kind of processor you have and enabled MMX if
possible before to compile ?
Good Luck !
Francois BERGERET,
France.
-Message d'origine-
De :
Hi !
Call Digium crew.
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
[EMAIL PROTECTED]
Envoyé : mardi 4 juillet 2006 11:20
À : asterisk-users@lists.digium.com
Objet : SV: [Asterisk-Users] Running 40 active
Hello,
Ridiculous business argumentation...
By changing 2 resistors maping on the same card you can say to system that
is any response as X100P, X101P, or Clone.
No proof to good quality or if it realy run !
Take a look to voip-info.org about X100P and X101P, you will learn more
about the
Hy men, :-)
Use Industrial PICMG PC's.
Higher cost at buy, but very stable and evolutive platforms.
SBC doesn't change during a long industrial period.
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Steve
Hello,
I use and sale (as Distributor) Micronet and Aliwei gateways.
Fine and stable, without echo.
Each port is seen as a separate SIP account.
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Nikolay
Pavlov
Hello Giorgio,
I am a TDM2400 happy user. :-)
Could you show your zaptel.conf zapata.conf config files ?
Think to tell us how many modules you have and where they are plugged on the
TDM2400P.
Are the leds on the echocan modules running as a LasVegas casino (scrolling
in a circular pattern) ?
If
Hi the list !
I share Ethernet card IRQ with my TDM2400 without any trouble here, on an
old Intel motherboard and an old PII400 !
This is another proof that sharing IRQ is not necessary an issue.
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
Hello,
Check your gren module by moving it from slot to slot on the TDM400P card.
If the problem is following your module, it's the module itself the cause.
If not, and running well on other slot, it's the TDM400P itself.
Good Luck !
Best Regards,
Francois BERGERET,
France.
Hello Yusuf,
This is a normal use of zap channels : it is not possible to see if the call
is realy answered, and Asterisk say yes as soon as the call is placed.
That's all...
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
exten = 19,1,Dial(SIP/19,20,tr)
Must be :
exten = 19,1,Dial(IAX2/19,20,tr)
Because you are using IAX IP-Phones...
Best Regards,
Francois BERGERET,
France
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Olivier
Saulnier
Envoyé : mardi 2 mai 2006 16:09
Hi Louis-David,
Check without crc4
Best Regards,
Francois BERGERET.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Louis-David
Mitterrand
Envoyé : dimanche 23 avril 2006 09:39
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] TE410P card
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
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
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
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
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
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 !
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.
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
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
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 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 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
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
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
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
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
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
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