repend 41 ---> Outbound route for "41|."
---> Appropriate trunk dial "."
The only matter is that I have NO clue on where to append this code for
outgoing calls from these specific extensions.
If anyone has a simpler idea (yeah, mine is pretty twisted %) ) or knows
how to
That makes PERFECT sense and also makes me aware that I need to review
asterisk theory :-P
I'll put it under test and let you know how it works.
Thanks a lot!
Marco
Rodrigo Gonzalez ha scritto:
> Create different contexts and assign them to the extensions
>
> [trunk1]
> e
What do you think?
Bye
Marco
Alan Lord ha scritto:
Hi there,
in case anyone is interested, I've just taken ownership of a small home
network (3 handsets) of the brand new Siemens DECT PSTN/VOIP phone.
It works great with Asterisk. Here's my overview and review
Lately I've been offering these stuff to a customer; a valid solution is
the one provided by Jabra with their DECT headset (see
http://www.jabra.com/Sites/Jabra/UK-UK/products/Pages/JabraGN9330.aspx )
and a "electronic lifter" as the one for the Snom phones here (
http://www.snom.com/en/adapter
Alan Lord wrote:
If you only have one analogue line why not just get a simple x100p card?
When you use OSLEC with them they work great here in the UK. I bought my
card from a USA based eBay seller. Total cost for card and shipping was
about £17.00
Respectfully, I don't agree. I've purchase
Personally, I love the "debian way", but I must admit that when it gets
to Asterisk, I prefer to use a RedHat-based distro like CentOS, first of
all for the proven reliability, then for the widely used rpm packaging
system and last because there are many distro CentOS-based that provide
a stabl
er says "no"!
Thanks,
Marco
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Thanks Michael,
that's a *huge* thing you're telling me, in the wiki details for the
PCI-X bus I've read about retrocompatibility, but I just wanted to be
100% sure. I can go on and order my server, now!
Thanks again
Marco
ps. This proves also the complete unaccuracy of
dred 7960s
using Asterisk and chan_h323 """, so my question is:
Asterisk supports this interoperability ?
I have done some test with Vocal to make calls from IP
cisco Phone via sip/h323 translator to connect with
Netmeeting or other h.323 end points...
If this interopar
dred 7960s
using Asterisk and chan_h323 """, so my question is:
Asterisk supports this interoperability ?
I have done some test with Vocal to make calls from IP
cisco Phone via sip/h323 translator to connect with
Netmeeting or other h.323 end points...
If this interopar
gnize liboh323wrap.so
Someone has installed and using with success this oh323
package from inaccess networks ???
thanks in advance,
Marco
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gnize liboh323wrap.so
Someone has installed and using with success this oh323
package from inaccess networks ???
thanks in advance,
Marco
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this ???
Please answer me...this is a big problem
Thanks Marco.
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on refused
Do I have to use kphone as a client to call the local asterisk server?
How do I do that? Can't I use 'CLI> dial' ? I was hoping to send a
soundfile.
Thanks for any clarifications,
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in this list in the past, but I would
like to know if someone has experience on this and could share their
opinion, tricks and/or statistical results on failure/success rate when
faxing. I think that this could be useful to other people have to realize
a system like that one depicted.
Thank you in
I can do to better
understand what's the cause of this?
Thank you and best regards,
Marco Signorini.
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rks for me, except, if I
remember well, some little problem on reload (but stopping and starting
again works fine).
Best regards,
Marco Signorini.
==
INGEGNI Tech S.r.l.
http://www.ingegnitech.com
> Anyone by chance got an Init script for /etc/init.d/dahdi on a SLES 10
Hi all,
i' using asterisk with chan_oh323 to test the SIP/H.323 interoperability. All working
well...
Now i would like to know if is it possible that * doesn't control the RTP traffic.
I would tht the end point control the RTP flow...
Any idea, is it possible???
Thanks
Ma
Hi all,
i would like to know if it is possible to bridging the rtp traffic over Asterisk...
I would like that the RTP flow is not controlled by * but by the endpoint.
Is it possible??? Any suggestion to do this???
Thanks
Marco
Ë^®+$Rǫ²f¢)à+-Ë^®+$Rǫ²X¬¶Çb+¦r¡¶ÚþX¬¶Çb+¦r¿¨¥©
Your solution is Asterisk Manager Interface
http://www.voip-info.org/wiki-Asterisk+manager+API
On Tue, Mar 18, 2008 at 6:24 AM, Janu Mukherjee <[EMAIL PROTECTED]>
wrote:
> Hi,
>
> I have our software with SIP running on it.I configured asterisk server as
> proxy. How do I implement the call scre
I would recommend you Asterisk for Voice and Video and XMPP for Chat.
Asterisk in parallel with Jabberd2 (XMPP server) may feet your requirements,
and if you use a XMPP MSN Transport Gateway you can do even more.
On Mon, Mar 17, 2008 at 5:50 PM, Carlos Carvalhar <
[EMAIL PROTECTED]> wrote:
> H
May be I'm wrong but:*
timeout - the maximum time, in seconds, the call will wait in the queue.
When this time expires, the next extension, by priority, will be executed.
By default the timeout is set to 300 seconds.
So you clearly have two ways to feed your database with your statistics:
If (ag
ng as DTMF Caller ID type, but still not working.
Let us know what kind of problem you have, maybe I can help you out.
Marco
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Doug
Enviada em: quarta-feira, 18 de junho de 2008 15:57
Para: Asterisk Users Ma
I got one of this boards and I got it successfully replaced by Avanzada7
(Digium official reseller) immediately.
On Nov 24, 2007 6:46 AM, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Actually if you rule out all the clone tormenta cards (nothing wrong..
> but very dated design... I wouldnt buy
Does this number (you are dialing) has been ported from a different Telco?
When you dial from the other city and you get "service not available" you
may be dialing from a different Telco that either has no route aggreement
for the dialed network, or the number portability database (of Out of city
e instances of Asterisk that are
started due to the bug ( http://bugs.digium.com/view.php?id=8086, solved in
asterisk 1.2.13) are causing all the troubles, making this multiple
instances try to access same asterisk channel (leading us to Avoiding
deadlock messages) ?
I mean applying the patch might so
Best regards,
Marco Mouta
On Dec 10, 2007 12:24 PM, Kovář Jan <[EMAIL PROTECTED]> wrote:
> Hello.
>
> I am going through the documentation and trying to find if asterisk can
> help me in my case. It is quite difficult to find answer because I do not
> know the exact question.
&
le that is loaded with my
modules.confthat I needed to copy from the backup
/usr/lib/asterisk/modules and give
the right permissions.
Am I missing something?
best regards,
Marco Mouta
--
Esta mensagem (incluindo quaisquer anexos) pode conter informação
confidencial para uso exclusivo do d
ion|]priority)
probability := INTEGER in the range 1 to 100
best regards,
Marco Mouta
On Dec 12, 2007 8:08 AM, Eric Delaporte <[EMAIL PROTECTED]> wrote:
> Hi @ all,
>
>
>
> i set a server to a costumer of mine with a TE207P for use with 2 E1
> Lines.
>
> I set them to
Post:
Asterisk CLI : sip show peers
Asterisk CLI : zap show channels
Asterisk CLI: zap show status
As well as your extensions.conf
Are you able to ping you GSM gateway? is connected via SIP or Telephony
interface card?
Best regards,
Mouta
On Dec 18, 2007 10:47 AM, Lolu Gbenga <[EMAIL PROTECTE
In
http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-phones-home
is said Kerry Garrison that:
Both trixbox and FreePBX have phone-home mechanisms in them.
So does FreePBX phones home too?
On Dec 17, 2007 4:27 AM, Than Taro <[EMAIL PROTECTED]> wrote:
> As I pointed out here l
Thanks Tzafrir!
I really appreciate Free PBX.
Keep on going your good job.
Best regards,
Mouta
On Dec 18, 2007 11:59 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Tue, Dec 18, 2007 at 11:38:03AM +0000, Marco Mouta wrote:
> > In
> >
> http://www.trixbox.org/fo
What do you mean with record a call on hangup? If the calling party ends the
call you want to keep recorded file?
On Dec 18, 2007 6:27 PM, Jamshed Zaidi <[EMAIL PROTECTED]> wrote:
> Hello everyone out there, I am having a problem in call recording with php
> agi library. I have already recorded v
nds). What i want is During 30 seconds if user
> does hangup his/her call then message should be recorded
> otherwise(after timeout) message is discarded. Is there any thing that
> will help me...???
>
> currently I am doing the same thing on pressing 1 with php agi script
> and its wo
% -> Max time for Outage during one month is 0,432 minutes
If any of you around the world is aware of this values for VoIP SLAs I
would be thankful to exchange and discuss this info.
Thanks in advance.
Best regards,
Marco Mouta
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ng goes fine but now i'd like to try this wireless solution in order
to be reached by a phone call i.e. when i'm on a train using my laptop
(where i would have asterisk running).
Do i need a particular driver to do so?
How can i make asterisk "loo
B connection in asterisk?
Basically what i need is really near to chan_mobile but i don't need
Bluetooth connection...only USB.
Thanks again
Marco M.
p.s. am i writing in the right place or should i write to another
asterisk-related mailing-list?
__
Dear all,
I've created a digium certified asterisk professional - dCAP linkedin
group for anyone, dCAP, interested:
http://www.linkedin.com/e/gis/60298/39AE1350DBF3
Best regards,
Marco Mouta
dCAP
November 2006
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confide
the right direction?
Below is my configuration. The sofpthone is registered as 1060.
Thanks in advance.
Marco Signorini.
pjsip.conf:
[transport-tls]
type=transport
protocol=tls
bind=0.0.0.0
cert_file=/etc/asterisk/sslcert.pem
method=tlsv1
[1060]
type=endpoint
transport=transport-tls
context=from-i
where the SIPML5 seems not able
to connect to the asterisk box.
Thank you and best regards,
Marco Signorini.
On 06/12/2014 03:21 AM, Steve Ng wrote:
I am using Asterisk v12.3.
As far as DTLS, I understand that applying the following Javascript
will temporarily fix for SIPML5 to Asteri
risk 11, but there is the same
problem.
I've already read all the information about canreinvite and directmedia
Can anybody help me?
Thanks a lot
Marco
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Hello,
I set up a new server for Asterisk with 11 cert 6 on it. I am migrating from a
previous server. I have replicated all the configurations, modules and setup
that I know of. However, when I tested an outbound call, it didn’t work.
Checking the asterisk message log yielded nothing. Any idea
t38_udptl=yes
t38_udptl_ec=fec
t38_udptl_maxdatagram=400
[trunk-patton]
type=auth
auth_type=userpass
password=X
username=X
=
Thanks
Marco
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To
It seems to me, that your problem is that X100P is not detecting that
the caller has hangup through PSTN.
I really got lots of problems with disconnect detection, and currently
i only get it working on asterisk @home 1.5 , it doesn't work well on
later releases.
The main changes i've made in zapa
help me with this.
Best regards,
Marco Mouta
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Hi i'm developing a solution with ASterisk, but in fact i don't know
which ATA SIP device should buy.
Could you give me some advices?
Marco Mouta
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To U
-- Forwarded message --
From: Marco Mouta <[EMAIL PROTECTED]>
Date: Feb 15, 2006 1:58 PM
Subject: Which ATA device do you recommend?
To: [EMAIL PROTECTED]
Hello,
I'm developing a Voip Solution for a client, which ATA SIP do you
recommend? there are some ATA devices f
help me with this.
Best regards,
Marco Mouta
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Hi all,
I'm going to buy E1 digium110P ,any one knows how i can get faxtomail
working for three different channels?
I mean:
channel1-->[EMAIL PROTECTED]
channel2-->[EMAIL PROTECTED]
for 1 channel is not dificult [EMAIL PROTECTED] and NVfaxdetect
Using [EMAIL PROTECTED] 2.5 faxToPDFmail works, b
nk in advance, regards,
Marco.
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.
I have then to restart the phone to reactivate the Buddy Watch function.
Is there anybody that can help me with this problem? Is it a problem of the PBX
or a problem of the phone?
Thanks in advance, regards,
Marco.
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I solved that problem for Polycom phones with the patch at:
http://bugs.digium.com/view.php?id=6047
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Hi Douglas,
I'm using Asterisk-1.2.4
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es, am I wrong?
Best regards,
Marco Mouta
On 1/18/06, yrving rivas <[EMAIL PROTECTED]> wrote:
>
> Ok, thanks, it works for me.
>
> Regards,
>
> Yrving
>
> Dovid Bender <[EMAIL PROTECTED]> escribió:
> If you are new I would reccomend using [EMAIL PROTECTED]
>
Is this a bug? Am I doing something wrong?
Best regards,
Marco Mouta
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[macro-ricezionefax]
exten => s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten => s,2,rxfax(${FAXFILE})
exten => s,102,Goto(2)
Is this a problem of spandsp (I'm using spandsp-0.0.2pre25)
or is there an error in my configu
Could it be Call Waiting Deactived?
On 3/7/06, Rolf Brusletto <[EMAIL PROTECTED]> wrote:
> All - I've been muddling around with this for a few days now.. and I'm
> trying to figure out why I am not receiving more than one phone call on
> each polycom 501 phone. I can make more than one phone call
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a on a Debian 3.1
The system is connected with an HFC card directly to the telco line
card is in TE mode
and signalling used is bri_cpe_ptmp
I am able to dial out some "numbers" and some not.
In particular it seems that i can't call mobiles and special telco
nu
Matteo Brancaleoni ha scritto:
I am able to dial out some "numbers" and some not.
In particular it seems that i can't call mobiles and special telco
numbers like the information call center, emergency numbers,...
try with:
pridialplan=unknown
prilocaldialplan=unknown
it works.
thanks
I'm using autodial in conjuction with TxFax to send faxes on demand.
An home made application generates the call file and puts it in the
outgoing spool, the file is like this:
Channel:Zap/g1/1232314324
MaxRetries:0
RetryTime:60
WaitTime:20
Context:faxout
Extension:s
SetVar:FAX_FILE=/shared/awfa
Manuel Casal ha scritto:
I made the "make menuconfig" and "make dep" in the kernel sources.
i do not remember well how i solved that problem but i'm sure that "make
dep" will issue you a warning and stop.
run "make" to start the kernel build process and then stop it after few
seconds. it wil
Manuel Casal ha scritto:
make[1]: Entering directory `/usr/src/linux-2.6.8-24.16-obj/i386/smp'
make[1]: *** No rule to make target `modules'. Stop.
make[1]: Leaving directory `/usr/src/linux-2.6.8-24.16-obj/i386/smp'
make: *** [linux26] Error 2
linux:/usr/src/asterisk/bristuff-0.2.0-RC8g/zaptel
Roger Schreiter ha scritto:
Hi,
package tiff-v3.5.7 contains the currently recommended version
of libtiff in order to run spandsp (fax support for asterisk).
i had no problems receiving faxes with version 3.7.2.
on the other hand i have big problems in sending multipage faxes. only
the first
Remco Barende ha scritto:
Do you see anything on the console even if you dial a number that isn't
answered?
i see this for a non existant number:
Attempting call on Zap/g1/12345 for [EMAIL PROTECTED]:1 (Retry 1)
i guess it prints out for every call originated by a call file.
asterisk -cvv
Marco Parmeggiani ha scritto:
on the other hand i have big problems in sending multipage faxes. only
the first page goes through.
uhm, no, neither the first page is received. i was optimistic.
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Can someone explain me what's going on and why the receiver of this fax
guives up saying communication error?
Slow carrier up
Slow carrier down
Slow carrier up
<<< CSI: 40 20 20 20 20 20 20 20 34 39 34 35 36 34 39 35 30 20 39 33 2b
CSI without final frame tag
Remote fax gave CSI as: "+39 059465
Roger Schreiter ha scritto:
Marco Parmeggiani wrote:
> ...
i had no problems receiving faxes with version 3.7.2.
on the other hand i have big problems in sending multipage faxes. only
Hi,
where did you get that version?
On libtiff.org, 3.6.1 is the most recent one.
you're poi
I tried with several iax softphones:
iaxcomm
idefix
iaxphone
and i have a problems that i do not have with SIP clients.
A calls B, B phone starts ringing, asterisk says that call has been
accepted, that is ringing but it is not yet answered. If B "picks up",
asterisk says that call has been an
Richard Cook ha scritto:
Hello,
Has anyone had issues with faxes showing up squished in the TIFF file?
Any ideas what could be causing it?
there's a faq on the spandsp site.
the problem is not with spandsp. it's with the image visualization
program. (i.e. irfanview 3.97 (win32) has the
change
the IRQ, but it will always move them together, anyone with some info
about my options ?
Thanks,
Marco.
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Only one PCI slot can hold the full size card like the TDM400P , the
other slot has a smaller opening on the case.
Marco.
Alexander Lopez wrote:
Can you try a different slot on the PCI bus??
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hi,
I tried setpci INTERRUPT_LEVEL (or something similar, cant remmeber
now), and also setpci seems like it changed the IRQ, lspci -v still
shows the old IRQ
Marco.
Stefan de Konink wrote:
On Sun, 25 Sep 2005, Marco Supino wrote:
I am building an asterisk pbx (1.0.9) on an IBM x306
seen by the
BUS and not by the kernel) shows me the TDM400P is on IRQ 5, why does
the kernel puts it on IRQ 7 ?
any insights much appriciated.
Marco.
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the zttest testings,
Thanks for any info.
Marco.
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Hi,
My TDM is on its own IRQ, and the x306 has only one full-size PCI slot..
so no playing with it,
what results do you get from zttest ? what IRQ is the card on ?
Marco.
Damian Funnell wrote:
Have you checked that the TDM400P isn't sharing an IRQ with anything
else? Don't t
Yes, i am having timeouts on registering to the LAX sip server of
broadvoice.
Marco.
Nate Kapi wrote:
I've been having a lot of problems with Broadvoice lately. Anyone else
been without service for extended periods of time this
Thank you for help
Marco
server2*CLI> sip show users
Username Secret Accountcode Def.Context
ACL NAT
server2*CLI> realtime load sipusers name 301
Column Name Column Value
Hello
On Fri, 14 Oct 2005 01:25:20 -0500, Kevin P. Fleming wrote
> Marco Balmer wrote:
> > Any ideas or hints?
> Yes. Whatever documentation told you that you could share a Realtime
> SIP peer database between two Asterisk servers was in error (or at
> least very incomplete).
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I need to add and remove Sip accounts in realtime.
What's the best way at the moment to do that?
* Add/remove the user into the sip.conf and execute asterisk -x 'sip
reload' ?
Thanks for help
Marco
Kevin P. Fleming schrieb:
>
Hi,
As anyone tried integrating App_Directory with any Text2Speech mechanism
like festival ?
Marco.
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did you notice the two dots in the IP address of ldaphost ?
Marco.
Chandan Mishra wrote:
Hi
I want to authenticate the asterisk users from the LDAP directory server
not from the sip.conf.
I tried to use the astirectory-1.2
<http://www..asterisk-ev.org/download/astirectory-1.2-0.3.
on or
tip ???
thanks
marco
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the Box will loose connection after a few seconds:
Aug 21 13:16:10 NOTICE[98310]: chan_sip.c:7653
sip_poke_noanswer: Peer '10' is now UNREACHABLE!
I would be really pleased for any help :)
Regards,
Marco
canreinvide=no
Conf = configured or conflicted?
I try a "Loop" but nothing happend.
TxA, TxB etc. are empty, too.
Can someone help me? - I really need some sample
configs, too.
Which linux distribution runs smoothest with Asterisk?
Thanks!
Marco Czudej
t;1-2
--
If you need additional infos plz tell me. :)
Thanks,
Marco
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Some people is still waiting for last Astricon materials; what about them ?
Regards.
Marco Vescovi
-Messaggio originale-
Da: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di Olle E.
Johansson
Inviato: mercoledì 26 ottobre 2005 8.42
A: Asterisk Users Mailing List - Non
Hi,
I am using Asterisk 1.0.9 with the 1.2.0 zaptel, just for the fxotune
utility, which solved my echo problems , my zttest results are low, but
no echo on ZAP lines...
Marco.
Chris Miller wrote:
Mojo with Horan & Company, LLC wrote:
The recent suggestion on the list was to not
ringing the line, i
need something like Voicemailexists , but for SIP peers.
any solution ?
Thanks.
Marco.
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channels while
testing Asterisk,
Anyone with experience, sample configs or idea, please contribute.
Thanks.
Marco.
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http
ns in
zapate.conf , but nothing helped, any solution ?
the lines are coming from SBC in San Fransisco, i asked them if i have
"disconnect supervision", and they said i do have it.
Marco.
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configurable ? i would like to get the caller id of
international callers , with all digits.
Any solution ?
Thanks.
Marco.
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Yes, didnt change anything
Marco.
Angelito Manansala wrote:
hmmm
di you try this ;hanguponpolarityswitch=yes
Cheerz!
On 11/17/05, Marco Supino <[EMAIL PROTECTED]> wrote:
Hi,
I have a long delay when detecting hangups on the TDM400P card, with 4
FXO ports,
When an incoming call dia
other end,
Any idea/solution ?
Thanks.
Marco.
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r end,
Any idea/solution ?
Thanks.
Marco.
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Hi,
I need some info from people with the x100p card (digium or clone),
please send me the output of "lspci" and "lspci -n" from your linux
machine, i am tring to find out something on my * server.
Thanks.
Marco.
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? anyone has them working
with any type of modem ? (aopen or bestdata).
Marco.
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Hello All,
I am using an E100P card on a PRI
line. I need to setup a FAX extension. Can somebody help me please?
Marco
l from h323 side to sip side all work
When a try to place a call form sip to h323 nothing happen
Does someone try this???
Any suggestion will be appreciate
Tnx
Marco
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stuff", I also suggest
you:
a) use his "bristuff" (same site as above)
b) if you have kernel 2.6, use mISDN kernel patches and chan_mISDN, that
is, seems, well supported and developed (and works, with a compilation
flag, with asterisk stable and asterisk head as well):
http://www.be
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