, but this is my tip the main idea is that you
need to catch de DID and make a GoTo for the context you want.
Best regards,
Marco Mouta
On 4/2/06, Rich Adamson [EMAIL PROTECTED] wrote:
Steve Gladden wrote:
What version of asterisk? (been lots of changes happening to the sip
code over the last
Is it a 2.6 kernel? Did you included CRC_CCITT and RTC support when
you made the make menuconfig?
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de nik600
Enviada: sábado, 1 de Abril de 2006 10:28
Para: [EMAIL PROTECTED]; Asterisk Users Mailing List
Hi all,
I've my Server running well, then sometimes Sjphones looses registry
and it only works well again if i restart the pc running sjphone.
Has any one experience this?
Best regards,
Marco Mouta
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system does this PC have and is it up to date with security and bug patches.
Thanks
Marco Mouta wrote:
Hi all,
I've my Server running well, then sometimes Sjphones looses registry
and it only works well again if i restart the pc running sjphone.
Has any one experience this?
Best
the
users:)
Best regards,
Marco Mouta
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Hi all,
I've created this test.call file and it is not running outgoing call files:
i've made mv test.call /var/spool/asterisk/outgoing and nothing happens
Channel: SIP/200
MaxRetries: 3
RetryTime: 40
WaitTime: 25
Context: from-internal
Extension: 200
Priority: 1
My asterisk is running with
call on ZAP/[EMAIL PROTECTED] for
[EMAIL PROTECTED]:1 (Retry 2)
-- Attempting call on ZAP/[EMAIL PROTECTED] for
[EMAIL PROTECTED]:1 (Retry 1)
Do I need to define context to outbound calls through my ZAP ?
Thanks in advance,
Marco Mouta
On 3/28/06, Tomislav Vojvodic [EMAIL PROTECTED] wrote:
I
it's working , the problem was:
Channel: ZAP/g1X
I changed to ZAP/g1/X
And it's working fine!
Thank you all
On 3/28/06, Marco Mouta [EMAIL PROTECTED] wrote:
Thank you for your fast reply!!!
It's working on for SIP:)
I've tried to my zapata and doesn't make the call, i get
In fact i've never done it. And i don't have any Cisco Phone...
If i find something i will report it here :)
Best regards,
Marco Mouta
On 3/28/06, Tomislav Vojvodic [EMAIL PROTECTED] wrote:
heh.. I just noticed that ;)
Heh, do you know maybe how to update time/date on all Cisco 7905 phones
I'm a bit newbie, could you tell me how to i apply the patch?
Thanks in advance
Marco Mouta
On 3/27/06, Benoit Panizzon [EMAIL PROTECTED] wrote:
On Friday 24 March 2006 16:05, Benoit Panizzon wrote:
Hi all
Apparently there is a patch for those 1.2.4/5 MeetMe Freezes:
http
it helps.
Best regards,
Marco Mouta
On 3/22/06, Charles Marcus [EMAIL PROTECTED] wrote:
C F wrote:
Polycoms are not the best if you want a phone that works behind NAT.
Do you mean in general? Or only if you are trying to interconnect
multiple offices?
Are Polycoms fine for just one office
prilocaldialplan=local
nationalprefix=0
internationalprefix=00
faxdetect=both
callwaiting=yes
echocancel=yes
immediate=no
overlapdial=yes
echocancelwhenbridged=yes
echotraining=400
callerid=asreceived
group=1
context=from-pstn
channel = 1-15,17-31
Thanks in advance, Marco
,
Marco Mouta
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entered 0 (string) which is empty()
$agi-stream_file(dir-nomatch);
} // else, we timed out
Probably it's because i'm newbie, but is it correct 3 equals? ($digits
=== 0)) ?
Best regards,
Marco Mouta
regards,
Marco Mouta
On 3/10/06, Olle E Johansson [EMAIL PROTECTED] wrote:
Friends in the Asterisk.org community,
There is a lot of cool stuff going on in Asterisk development, things
that will change Asterisk and
make it work better in your organisation, make it easier to sell in
your area
suggestions?
Best regards,
Marco Mouta
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from users with
problems ( in fact i didn't find any sucessfull feedback).
I'm a bit afraid of doing all the tutorial and get in troubles with my
stable asterisks
Any one has tried it?
Best regards,
Marco Mouta
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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 out,
this to a real Linux System.
Does any one could help me understanding what is going on?
Best regards,
Marco Mouta
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entering two diferent days on my conference room... Also I don't think
it is a good choice to contact Administrator to change my Meetme
password.
Best regards,
Marco Mouta
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connection 4Mbit.
I hope this Excellent mailing list could help me on giving me some
Feedback and or advices/tips.
Best regards,
Marco Mouta
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,
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]
http://asteriskathome.soundforge.net . It is a great
resource for beginers. Also
? Am I doing something wrong?
Best regards,
Marco Mouta
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= 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 configuration?
Thanks in advance,
Marco
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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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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in advance, regards,
Marco.
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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,
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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-- 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 fully tested
regards,
Marco Mouta
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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
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.tgz
end,
Any idea/solution ?
Thanks.
Marco.
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Any idea/solution ?
Thanks.
Marco.
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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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,
Is this 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 dial's
channels while
testing Asterisk,
Anyone with experience, sample configs or idea, please contribute.
Thanks.
Marco.
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ringing the line, i
need something like Voicemailexists , but for SIP peers.
any solution ?
Thanks.
Marco.
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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 use
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,
As anyone tried integrating App_Directory with any Text2Speech mechanism
like festival ?
Marco.
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-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:
Marco Balmer wrote
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).
Server1 acts as a SIP
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 week
Marco
server2*CLI sip show users
Username Secret Accountcode Def.Context
ACL NAT
server2*CLI realtime load sipusers name 301
Column Name Column Value
id 6
% in 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 trust /proc
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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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
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
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
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 pointing
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
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
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
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 059465494
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
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 will
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
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
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
Nick Barnes ha scritto:
I've only ever seen when the signalling is wrong. For example if the line is
in PTMP mode when it should be in PTP or vice-versa.
this is the zapata.conf:
group = 1
context=default
signalling = bri_net_ptmp
channel = 1-2
So, you're using NT mode PTMP signalling.
?
Will the computer be sufficient, or would you take a more powerful one?
I can always take this computer and later take a more powerful one, but I
would lose all the money of the setup, so I would like to take the good one
since the beginning.
Thank you, ciao
Marco
Emanuele Pucciarelli wrote:
Are you sharing the IRQ? (check /proc/interrupts)
hi, what do you think? this is a bit too much low level for me.
ciop:~# cat /proc/interrupts
CPU0
0: 780877027IO-APIC-edge timer
7: 2IO-APIC-edge parport0
9: 1
Hi, i've downloaded/compiled/installed the bristuffed asterisk
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a
and i'm using it with an hfc card. It runs on a debian 3.1 sarge machine
with kernel 2.6.11. Asterisk works well if i configure the card using
isdn4linux.
I'm having problems dialing out (not
Hi,
Anyone has a working example of VoiceXML with asterisk ? i was looking
around voip-info and the internet, and couldnt find more then proof of
concept documents.
Also, does anyone knows how FWD does their VoiceXML (411) service ?
Thanks for any info
Marco
? anyone has them working
with any type of modem ? (aopen or bestdata).
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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Use externnotify (see
http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf) with a script
to send sms.
Some time ago I used a perl script called sendSms found in Internet.
Bye.
Marco
-Messaggio originale-
Da: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di
Julius
Do you have an 's' extention in the default context ?
Marco.
Dimitris Kounalakis wrote:
Hello,
I am trying to configure asterisk 1.0.7pre to get incoming calls from an
ISDN line using an AVM fritz PCI 2.0 with Chan_capi 0.3.5. My problem is
that the context is not recognised in the /etc/asterisk
In data Tue, 8 Mar 2005 18:25:38 + (GMT), hai scritto:
[chan_zap.so]Mar 8 17:53:06 WARNING[2447]: loader.c:258 ast_load_resource:
/usr/lib/asterisk/modules/chan_zap.so: undefined symbol:
ast_retrieve_call_to_death
Mar 8 17:53:06 WARNING[2447]: loader.c:391 load_modules: Loading module
and hangups the call but the other side still be connected..
I also see on asterisk cli this message:
chan_sip.c:787 retrans_pkt: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno 45854
(Non-critical Response)
Does someone experience the same problem?
Can someone help me?
Thanks.
Marco
I have some problems starting asterisk with a hfc card using zaphfc:
[chan_zap.so]Mar 8 17:53:06 WARNING[2447]: loader.c:258 ast_load_resource:
/usr/lib/asterisk/modules/chan_zap.so: undefined symbol:
ast_retrieve_call_to_death
Mar 8 17:53:06 WARNING[2447]: loader.c:391 load_modules: Loading
Hi all
where can I find infos aboutthis VXML
intepreterfor asterisk?
Thanks
Marco
Hi Foong, That's a good question you've put out there. Yes, Asterisk supports VXML andhere's how it's done; Firstly in the SIP.conf, you need to have your VXML application/browserdefined; sip.conf
?
Why these?
Can someone help me?
Regards.
Marco Ziglioli
Alascom Services S.R.L.
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will be glad to hear it.
Thanks.
Marco.
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happen on my system.
Can someone help me?
Thanks.
Marco
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Title: Messaggio
Ok!
problem solved!
tT
missed on extension used for test.
Thank
you very much for support
Marco
-Messaggio originale-Da:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di Dennis
WebbInviato: giovedì 24 febbraio 2005 19.05A: Asterisk
Users
be sufficent for one session (for
which FWD works fine with)
Any help appriciated.
Marco.
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for such a small system.
Hope it helps.
Marco
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rudolf
Ladyzhenskii
Sent: Monday, February 21, 2005 6:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Minimal hardware
You don't need the zaptel library if you aren't going to use any digium
cards.
Marco
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, February 17, 2005 8:02 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users
Go to the ITU website www.itu.org there you can buy all the specifications
you're looking for.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Daniel
Nyström
Sent: Tuesday, February 15, 2005 8:47 AM
To: asterisk-users@lists.digium.com
Subject:
such an implementation, and let me tell you that
this works really great!!!
Hope
this helps
Marco
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Rod
BaconSent: Tuesday, February 15, 2005 4:30 PMTo:
asterisk-users@lists.digium.comSubject
Have you set your DNS SRV entry for SIP correctly???
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andrew
White
Sent: Tuesday, February 15, 2005 7:04 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] newbie: help two cisco phones (sip)
Hi,
Dear Xu, my name is Marco Castillo, I'm in Guatemala, Central America, and I
have recently succesfully installed a TE110P here in Guatemala. There are
many implementations of a E1 or T1, but I think that the great majority can
be configured via the zaptel drivers. I will suggest you to buy a card
I don't know about your problem, but since you use mISDN, why not use
the specific chan_mISDN?
http://www.beronet.com/?PageID=3017
It's Free Software (GPL)
Regards
Marco Menardi
btw, if you login in their bug tracker, the home page has alink to a
document that tells you how install their boards
Remember that SIP uses DNS SRV entries, maybe one of the phones you use
efectively use the DNS SRV entry and the other not. Some VoIP phones have a
flag where you can deactivate this functionality for SIP. If not, make sure
you have in your local DNS a SRV entry for SIP.
Hope this helps.
Marco
.
Marco
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Peter
Svensson
Sent: Thursday, February 10, 2005 6:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] No dialtone in a E1
On Thu, 10 Feb 2005, Marco
=Video 097
canreinvite=no
disallow=all
;allow=ulaw
;allow=alaw
;allow=speex
allow=gsm
allow=h261
allow=h263
nat=yes
context=ip
;qualify=yes
;dtmfmode=rfc2833
Thanks for any help
Marco González
__
Do you Yahoo!?
The all-new My Yahoo! - What
packages. And modify the Makefile for the zaptel libraries to point to the
proper directories for the kernel headers (don't use usr/include/linux).
Hope this helps
Marco
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Geoff
Nordli
Sent: Thursday, February 10, 2005
from the POTS network makes a call to a
SIP client (through * and the E1) he doesn't receive the apropiate tone of
call progress. Does anyone has some ideas about this?
Ing. Marco Antonio Castillo
Chief Design Engineer
Van Der Kaaden IT Consulting
Guatemala, Guatemala C.A
(www.asterisk.org), go to resources,
there you will find good introductory material.
Marco
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of
vdasilvaSent: Wednesday, February 09, 2005 1:02
PMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users
Hi everyone!!! My Name is Marco, I'm from
Caracas,Venezuela. I'm a new Asterisk user...
I'm trying to make a video conference with sip. I have
the Eyebeam from Xten, a video sip phone. I have a
good audio conection, but nothing about the video
Now I'm trying to do the same with h323
in
the TE110P, I have no dial tone in the caller phone. Does anybody has a
idea???, Is this configurable in the zaptel.conf file???
Any help would be greatly apreciatted.
Ing. Marco Antonio Castillo
Chief Design Engineer
Van Der Kaaden IT Consulting
Guatemala, Guatemala C.A
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