On Wed, 4 Aug 2004, Steven Critchfield wrote:
On Wed, 2004-08-04 at 17:56, Dan Mahoney, System Admin wrote:
Hey all. I have a color-capable console (color ls works, and I can run
any color-smart program like naim and bitchX), but for some reason the
color in the console for asterisk, whether
a) enable me to connect that server to the public phone net
Yes
b) allow me to connect an ISDN phone to the server and use it as a
No. An ISDN phone is always an ISDN phone.
But with the right card (and magic, e.g. proper termination) you can
connect an ISDN phone to Asterisk.
1) I would think pfifo would be a better choice than sfq for your voip
qdisc. Something like:
$TC qdisc add dev $DEV parent 1:10 handle 10: pfifo limit 10
2) Marking packets worked better for me. I could never get it to work
any other way. (Hey, I'm not arguing. I'm jealous.)
3) Shouldn't
Dan Mahoney, System Admin [EMAIL PROTECTED] wrote:
Hey all. I have a color-capable console (color ls works, and I can run
any color-smart program like naim and bitchX), but for some reason the
color in the console for asterisk, whether started with -c or
safe_asterisk, isn't working for me.
Hi Steve,
Sorry to hijack the thread, but I'm confused, when I add 'tr' to the end of
my dial strings to enable the transferring of that call internally, it
breaks asterisk's dial plan totally. Calling any extension that has the tr
gives this error:
Aug 4 18:25:33 WARNING[17422]:
Hi all,
Hi Marco,
I was thinking about integrating an old PBX with Asterisk and I was wondering
some possible configurations.
You didn't mention the number of lines your PBX uses, but think of a third scenario:
Install an asterisk with twice the number of BRI/PRI-Ports your current PBX has.
My Bad, forgot the timeout on the dial string...
But I still cant transfer calls at all. Any ideas?
jd
-Original Message-
From: John Howard [mailto:[EMAIL PROTECTED]
Sent: 05 August 2004 09:18
To: '[EMAIL PROTECTED]'; '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] RE: No incoming
On Thu, 5 Aug 2004, Stefan Märkle wrote:
Install an asterisk with twice the number of BRI/PRI-Ports your current
PBX has. Connect half of them to your carrier, the other ones to your
old PBX (Some sort of proxy scenario, isn't it?).
Pro: - You don't have to change a single configuration
Thanks Tobias,
I was having a homer 'doh!' moment there!
Can anyone suggest anything as to why I still can't get call transferring
working with the Zyxel 2000w wireless phones? They have no transfer or hold
buttons, just a standard keypad. The phone is supposed to provide these
features using
I am unable to make quadbri board to work in NT mode, in TE mode it works
OK. I have set 5 jumpers to NT mode. Dmesg shows everything OK:
qozap: Junghanns.NET quadBRI card configured at mem 0xd084e000 IRQ 9 HZ
100 CardID 1
qozap: S/T ports: 4 [ TE TE TE NT ]
qozap: 1 multiBRI card(s) in this box,
On Thu, 5 Aug 2004, John Howard wrote:
Can anyone suggest anything as to why I still can't get call
transferring working with the Zyxel 2000w wireless phones?
Unfortunately not, but take a look into the ZyXEL 2000W threads in this
forum, for example
Hi,
I am unable to make quadbri board to work in NT mode, in TE mode it works
OK. I have set 5 jumpers to NT mode.
I have pretty much the same setup here and am able to connect to an AVM Fritz!X USB
(1xBRI, 4xAnalogue) and a TelesFON ISDN phone.
Are you use the power module which is an
I do not incect power to the board.
As far as I understand the power supply needed when you connect bri card
to phone without external power supply.
But my both devices use external power supply. small Lucent PBX uses
external power supply of cource.
I am unable to make quadbri board to work in
As far as I understand the power supply needed when you connect bri card
to phone without external power supply.
So does the Fritz!X, but it doesn't supply power to the ISDN bus... (Probably uses the
external power to drive the ringer on the analogue extensions)
HTH
- Original Message -
From: Chris Shaw [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 04, 2004 7:38 PM
Subject: Re: [Asterisk-Users] FCC Rules VoIP Must Be Tappable
- Original Message -
From: Florin Andrei [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
pretty pointless and may even bring scrutiny. feds are looking for
encrypted data. thats somthing the terrorists even know. they are
supposedly communicating without any encryption so as to not raise any
flags.
All in favor of IAX with native encrypted tunneling say Aye :-)
Now I'm likely
try removing your s,102 statement
- Original Message -
From:
Adam
Bezanson
To: [EMAIL PROTECTED]
Sent: Thursday, August 05, 2004 12:11
AM
Subject: [Asterisk-Users] Using answering
machine in my phone
Is this supported? I have a very
simple setup
On Wed, 4 Aug 2004 13:34:02 +0200 (CEST)
Peter Svensson [EMAIL PROTECTED] wrote:
On Tue, 3 Aug 2004 [EMAIL PROTECTED] wrote:
There is a device called a parlay made by a crowd
called
voxtream which will route the ISDN calls based on the
DID
and/or the callerid, before the call is
Title: Mensaje
Push
send after you number,
srsergio
-Mensaje original-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de James
DuttonEnviado el: jueves, 05 de agosto de 2004 12:28Para:
[EMAIL PROTECTED]Asunto: [Asterisk-Users] Call Transfer
Problems with Grandstream
Hi,
could you post your capi.conf..
Regards
Felix
I would set the MSN's to 855285 and 859609They do not
usually include the area code.
[local]
exten = _9XX.,1,Dial,CAPI/855285:bBYEXTENSION:1
exten = _9XX.,2,Congestion
exten = _9XX.,3,Hangup
;
; CAPI config
;
;
[general]
Sorry for the HTML-Messages, I have simply forgotten to change it before
sending.
Hello,
we had a running configruation where asterisk passed the phone
number and the ddi to the pstn (i.e. 595-431)
Now only the rootnumber arrives: 5950
I do not
- Original Message -
From:
James Dutton
To: [EMAIL PROTECTED]
Sent: Thursday, August 05, 2004 6:27
AM
Subject: [Asterisk-Users] Call Transfer
Problems with Grandstream Budgetone 100 Phone
I have two
Grandstream Budgetone 100 phones connected to my
On Wednesday 04 August 2004 05:26, lists-jmhunter wrote:
As seen on my post at:
http://www.sveasoft.com/modules/phpBB2/viewtopic.php?p=28112#28112
This works very well... It does NOT work with stable 4.0! sveasoft
will be issuing a bug fix for this (4.1) in the near future.
I've been using
On Wednesday 04 August 2004 05:26, lists-jmhunter wrote:
# lower the MTU to decrease latency
#$IP link set dev $DEV mtu $MTU
Just a note -- you're not lowering your MTU to 1492 to reduce latency (the
default is 1500), you are reducing it because you're running PPPoE and those
8 bytes are the
Thanks for that Tony, it all works perfectly now.
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Josh Roberson [EMAIL PROTECTED] wrote:
Simon, i was having the exact same problem, the only solution I found,
was to remove the secret, then it worked great.. I thought I must have
been
On Wednesday 04 August 2004 17:29, Steve Szmidt wrote:
I'll end up with ADSL too so would you be willing to send me a copy of the
config work you did?
My rc.tc script is posted in the QoS with sveasoft thread, the archives should
have it by the time you get this.
Regards,
Andrew
On Wednesday 04 August 2004 18:16, Mitchel Constantin wrote:
This may not be the point of your message, but just a side note, I
believe that echocancel and echotraining are turned off when a fax is
detected automatically regardless of whether or not they are enabled.
Yes this is true; you'll
On Thursday 05 August 2004 06:56, Andrew Kohlsmith wrote:
#!/bin/bash
... well that got bitched up sufficiently...
http://www.mixdown.ca/~andrew/dump/rc.tc is a copy of the script -- kmail is
trying to be smart and substituting soft line breaks for hard ones... ugh.
-A.
Good day all
IS there a way to personalise the voicemail message when you leave a
message?
Thanks
Altus
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To UNSUBSCRIBE or update options visit:
hi,
we are using a voip h323 switch. the switch sends all caals to our
Gatekeeper (gnugk).
gnugk musst send all calls to asterisk and asterisk must do his choice
(sip endpoint or out to PSTN)
Making calls to our h323 switch works fine over asterisk. what must i
configure to get inboung h323
On Thursday 05 August 2004 04:41, Peter Svensson wrote:
Note that you may only be able to reach full integration for external
calls. Not all pbx:es can be convinced to hand off some extensions to
Asterisk etc. The level of integration you can get depends on your pbx.
The Norstar Meridian and
This is what I've done:
Take asterisk, libpri and zaptel with cvs
After that :
cd /usr/src/zaptel
make clean; make, make install
cd ../libpri
make clean
make
make install
cd ../asterisk
make clean
make
make install
make samples
And I ve started and stopped asterisk in order to see if it run.
ok, this is an example for reaeching h323 endpoints.
i want to do this:
ISDN phone (cals 069)- phone switch (pstn to h323) gnukg -
asterisk - SIP endpoint (extension 102 / 069)
the call reaches out gatekeeper (openh323) an from there i want to send
the call to asterisk (on the same
Can you try with g711 to see if all is going ok?
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Asterisk .
Enviado el: Jueves, 05 de Agosto de 2004 10:10 a.m.
Para: [EMAIL PROTECTED]
Asunto: RE: [Asterisk-Users] H323 Call Dropping
Hello,
--- Sebastian
Give each user a voice box then use 1 of the vm broadcast patches in
the bug tracker so that 1 to all in a perticular goup can be done.
On Thu, 05 Aug 2004 15:57:36 +0200, Altus Snyman [EMAIL PROTECTED] wrote:
Good day all
I got my voicemail message working,thanks but now,keep in mind I'm
Hello,
--- Sebastian Nocetti [EMAIL PROTECTED] wrote:
Dial(h323/h323:[EMAIL PROTECTED])
I think problem is in this line...
Dial(h323/[EMAIL PROTECTED])
That's not the correct way?
Thanks for the reply.
I tried both, also without mentioning the gatekeeper_ip in the dialplan. All the 3
On Thu, 2004-08-05 at 09:57, Altus Snyman wrote:
Good day all
I got my voicemail message working,thanks but now,keep in mind I'm using
SIP
We have,for example 4 people in our admin department.Each user has its
own voicemail so that when their extension is dialed directly and not
answered
Doesnt make any difference.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Thursday, August 05, 2004
6:01 AM
To:
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
Using answering machine in my phone
try removing your s,102 statement
i've only used chan_h323 which suggests u download the tars and
extract them in /root. Takes a while to build but I did manage to get
it working
On Thu, 5 Aug 2004 13:47:29 +0200, Asmine Ouloube
[EMAIL PROTECTED] wrote:
This is what I've done:
Take asterisk, libpri and zaptel with cvs
After
Try to install a newer cvs version of Asterisk CVS Head.
There were some bug fixes on 07/28/2004 for more than one file.
I personally use bristuff 0.1.0RC2k, I had made update with cvs update -A.
It was going very well (08/03/2004) besides one file
asterisk/res/res_features.c.
You have to correct
Hi Thomas,
I have configured asterisk to interact with my PSTN (UK BT Line) using a
x100p zaptel card. I can make calls outbound using 9 as a prefix, I can
receive calls inbound over the PSTN and I can of course make calls between
the 2 zyxel 2000w's using the extension 2001 and 2002. When I
Altus Snyman wrote:
At the moment,if you don't get a answer for 10 seconds it goes to
voicemail saying something likethe user ate extension 101 is
unavailable please leave a message...
[snip]
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20VoiceMailMain
0 Mailbox options
Sorry for the lack of enough info, I was starting to think it just wouldn't
work and wanted to get confirmation or not.
As I mentioned in my first post, I do hear the answering machine start to
play the greeting then hangs up in the same spot during the greeting. This
is before the machine plays
if i understand right:
the asterisk must register at the gnugk. the prefix, asterisk registers
with, tells gnugk wich calls should be routed to asterisk.
if prefix in h323.conf ist like:
prefix=069
gnugk sends all numbers, beginning with 069xx to asterisk?
the asterisk shows in
On Thursday 05 August 2004 00:11, Adam Bezanson wrote:
Is this supported? I have a very simple setup where I have 2 X100P cards
and a TDM10B.
Should work without any problems whatsoever.
If I let it ring the TDM10B interface answers the call and the greeting
message of the answering machine
?
- Original Message -
From: Altus Snyman [EMAIL PROTECTED]
To: asterisk [EMAIL PROTECTED]
Sent: Thursday, August 05, 2004 7:25 AM
Subject: [Asterisk-Users] personal voicemail
Good day all
IS there a way to personalise the voicemail message when you leave a
message?
Thanks
Altus
I have a project I am trying to implement in Asterisk that I cannot seem to
find a solution for. I have 2 IAX calls, 1 is incoming and is on hold, and
the second one is an outgoing call created using a call file. Is there any
way to bridge these calls together, so that instead of both of them
On Wed, 2004-08-04 at 10:32, Simon Ward wrote:
Hi everyone,
I'm having some problem trying to set up an IAX connection between two *
servers.
The scenario is :
serverA has an X100p card and will direct all calls from the X100p over
IAX to a specific extension on serverB which is at the
Hi, I have been reading the mailing list archives and google results but havent
been able to find quite the information i am looking for. Perhaps someone on
the list can give me a hand.
Our company has several sites. Two of which we regularly make calls between over
the PSTN. Both sites currently
Unfortunately it doesn't help. If I specify actionid then I'll recevie
Response: success or error with that actionid, but events will still
flow without it. Any other ideas?
Thanks,
Michael
On Wednesday 04 August 2004 07:15 pm, Nicolas Gudino wrote:
Hello,
On Wed, 2004-08-04 at 18:56,
Calling all Definity admins,
Got a few questions about Definity - Asterisk interoperability.
1) What are the options for integration? Can I hand off extensions from the
Definity and vice versa?
2) Anybody have any working configs they would like to post?
I've found and read the legacy
On Thu, 5 Aug 2004, Andrew Newton wrote:
Our company has several sites. Two of which we regularly make calls between over
the PSTN. Both sites currently have thier own PBX systems but they are not
linked by any means other than the PSTN (and we pay by the min for calls
between sites)
Both
I'm having difficulty getting a PRI working between asterisk and a Merlin
Legend. I have been attempting to get it going as esf, b8zs 5ess (although
I have tried other switchtypes...). I found the following info in the Legend
docs:
If the switch type is 5ESS, the protocol MUST be set for ATT
Per that url you have to get a support contract to change the phone from
skinny to sip and you got the runaround trying to contact Cisco to get
access to the sip images.
Gary
Which hurdles are you talking about specifically? These phones work
great with asterisk (as long as you install
When I load wct1xxp and zaptel and do a ztcfg -v all channels come up, but
I get a pulsing orange light. (normal ?) On the Definity side, the 464 has
red lights pulsing on 2 and 4.
Hi-
Better post your zaptel.conf and zapata.conf. I'm not a Definity expert,
but something's wrong. Maybe, for
Can you please say what problems you're having?
Does the board come up correctly and display a green LED?
Are there errors on the console
Etc...etc
Thanks
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
-Original
Hello group,
I wanted to try out the asterisk iax protocol between two asterisk
machines but have several problems with it.
My scenario looks like follows. I am using asterisk 0.9.0 on both machines.
SER1 - asterisk1 - IAX - asterisk2 - SER2
Both SER and asterisk run on a machine with a public IP
Robert,
I'm actually looking into the same thing as you.
I have (and have had) them talking to each other OK for some time.
The Wiki describes how to do that.
The basic setup will allow calls from asterisk - Definity OK.
I have some outgoing calls routed to asterisk via ARS on the
On 05/08/04 15:24 +0100, Tom Lawrence wrote:
snip
0Kernel panic: fatal exception in interrupt
i have had to rebuild the kernel to get the modules in but they seemed to go
in ok after that. If I run ztcfg I can see both of the cards working. Could
it be something to do with the IRQ numbers
H323/${EXTEN} since your using a gatekeeper
Jeremy McNamara
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Can you please say what problems you're having?
Does the board come up correctly and display a green LED?
Are there errors on the console
The legend looks OK, When the PRI is idle, it too looks OK. When a call is
routed from asterisk to the LEGEND, * can no longer find the d-channel and
the
There has not been any comment on this thread does this mean there are
PROBLEMS??
mohammad mirzaee wrote:
HI ALL;
Is there anybody who use app_radius(astersik radius module)???
is it stable?
Regards
mohammad
--
~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~
Clayton Gray,
Has anyone used the Voip Billing System from http://advancedvoip.com/ ?
They seem to also offer a billing solution for Interconnections. I'm
curious if anyone has some experience using their software?
Thanks,
- Darren
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Asterisk-Users mailing list
Given:
Queue(foo|tHnr||bar)
where queue foo includes something like IAX2/gw/18005551212
should # transfer work on the remote phone?
A read of app_queue.c looks like it ought to work, but all
I get is dtmf sent to the caller.
(Incidently, I'd really prefer to be able to hit eg * during
the
Hi,
I am trying to link up a comdial PBX to Asterisk using T1 tieline EM. I
have it working for comdial to asterisk but not the other way. Comdial does
not listen for any DTMF before answering the ZAP channel and requires codes
before allowing asterisk to call an outside line or inside extension.
Seth Remington wrote:
In voicemail.conf set up your Admin mailbox:
101 = 101,Admin Mailbox,,,delete=1
This will allow you to record your Admin prompt but the delete=1 will
auto delete the message from this mailbox. Then in extensions.conf:
exten = 2,1,Voicemail(101102103104)
Change the extension
Hi,
I've just initiated
a new bounty for the above;
http://www.voip-info.org/wiki-Asterisk+bounty+callingcard+to+MySQL
Any takers or any
contributors please respond to me privately. I do not know exactly how the
bounty process works, but I can coordinate on this ?
SW
On Thu, 5 Aug 2004, Sathya Weerasooriya wrote:
Hi,
I've just initiated a new bounty for the above;
http://www.voip-info.org/wiki-Asterisk+bounty+callingcard+to+MySQL
Any takers or any contributors please respond to me privately. I do not know
exactly how the bounty process works, but I
SW, let me know if you find anyone. I'd be willing to
contribute some funds to make it work properly. :-)
--Richard Cook[EMAIL PROTECTED]Tel:
705-497-9320
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sathya
WeerasooriyaSent: Thursday, August 05, 2004 2:04 PMTo:
[EMAIL
On Thu, 2004-08-05 at 13:57, Wayne wrote:
Seth Remington wrote:
In voicemail.conf set up your Admin mailbox:
101 = 101,Admin Mailbox,,,delete=1
This will allow you to record your Admin prompt but the delete=1 will
auto delete the message from this mailbox. Then in extensions.conf:
I tried to get it working, even opened several tickets with them, but
eventually hired a programmer friend to build something similar for me...
I'm sure with the proper programming and time you could get it working, but
it won't have nearly the same level of integration as a it would with a
CISCO
I canhelp on this work for free. no
problem
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Richard
CookEnviado el: Jueves, 05 de Agosto de 2004 03:17
p.m.Para: [EMAIL PROTECTED]Asunto: RE:
[Asterisk-Users] new bounty for modifying calling card application to
mysql
SW, let
Today is the first time I've had any show stopping errors with asterisk
since I started using it over 8 months ago... While I wait on the telco to
check the circuit I thought I would post this here. Not sure yet but I
thinks my 4-port PRI card is gone!!!
I made a call and right when the
Hi mohammad,
app_radius and cdr_radius apps are working pretty well :)
If you have any questions please go to the project web site:
http://appradius.minitelecom.org
or contact us at: [EMAIL PROTECTED]
Best regards,
Lubo
Clayton wrote:
There has not been any comment on this thread does this mean
I have the setup you described working somewhat. You *will*
need a T1 crossover(at least I did). You also have to be careful about what
the Definity is doing. After programming my Definity my CSU
liked to stay in the loop state even though I put the thing in service.
To compound the matters it
Hello !
I am a new user with Asterisk software. I managed to compile
asterisk on my Redhat 9 box, and also to compile openh323 and pwlib software. I
configured Asterisk to load the h323 module:
[chan_h323.so] = (The NuFone Network's Open H.323
Channel Driver)
== Parsing
Actually, this may not be doing what you think. It remembers all of the
options from earlier channels. This would work better if it was the
first channel you defined.
Ah thanks forgot about that. Still no difference though :(
What happens if the *only* thing on the port is the answering
The box has a T100P card hooked up to a csu on the Definity with a
patch cable.
A. You don't need a CSU if located close to each other.
B. Patch Cable? If using CSU, straight thru cable, if no
CSU cross-over cable.
C. Make sure to know your pin outs on the CSU vs pin outs T100P
Lucent/Avaya/ATT
Howdie:
I've been reading some old threads and still have a couple of questions
about applying the AGI_BACKGROUND script inside a Conference. Perhaps
someone can save me a bit of fidd'lin.
Am I right in assuming that the MEETME_AGI_BACKGROUND script **WILL WORK**
on SIP conferenced channels
On Thu, 2004-08-05 at 15:21, Adam Bezanson wrote:
Me too. Something's indeed strange. I'll probably just end up using the
Voicemail system in Asterisk then.
FWIW, the voicemail system in Asterisk is pretty nice. Just one of the
reasons I like the system as a whole.
On Thu, 2004-08-05 at 15:04, [EMAIL PROTECTED] wrote:
span=1,1,0,esf,b8zs
#bchan=1-23 (Uncomment if you are doing PRI)
#dchan=24 (Uncomment if you are doing PRI)
fxsls=1-12 (Remove if you are doing PRI)
em=13-24 (Remove if you are doing PRI)
loadzone = us
defaultzone=us
Does CID work when
Advanced VOIP Billing Software is not reliable as we some bitter experience with this
software. We used to lose about 10% of the CDRs and the Abilling Software is user
un-friendly. The User Interface is very cludgy and does not follow any of the Windows
or industry standards.
The Menus are
Is it possible to send the CallerId to IconnectHere with Asterisk
when making outbound calls?
I read somewhere that it doesn't work.
I have set up everything to send the correct CallerId info to IconnectHere
but I get a 442-887-926267 caller id.
In [globals]
ICONNECT1=1713...(my number)
Seth Remington wrote:
mailbox (the 101 Admin mailbox in the above example). I'm not sure what
your setup is but most channel types support checking multiple mailboxes
to send a MWI. So you could alert the user if there was a message
waiting in their personal mailbox or the group one.
I would like to invite and encourage everyone involved with Asterisk to
attend AstriCon September 22 - 24 at the Atlanta Marriott Century Center.
This will be an unprecendented opportunity to build relationships with the
people in the Asterisk community. Mark Spencer, all of Digium, and all
In article [EMAIL PROTECTED],
jeff quade [EMAIL PROTECTED] wrote:
Howdie:
I've been reading some old threads and still have a couple of questions
about applying the AGI_BACKGROUND script inside a Conference. Perhaps
someone can save me a bit of fidd'lin.
Am I right in assuming that the
Did iConnectHere ever fix their inbound DTMF problem? Is it useable with *
again?
- Original Message -
From: ERwin Hernandez [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, August 05, 2004 12:41 PM
Subject: [Asterisk-Users] iConnectHere and CallerId
Is it possible to send the
Hello,
I am connected to my SIP provider through the DSL router with NAT.
I do have a dynamic ip address. My router ip is on 192.168.1.1, the * box
is on 192.168.1.254. To overcome the NAT problems I forward all incoming
packets to my router and I use the externip,localnet,nat * options.
- Original Message -
From: Bartosz Wegrzyn [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, August 05, 2004 1:21 PM
Subject: [Asterisk-Users] NAT problems
Hello,
I am connected to my SIP provider through the DSL router with NAT.
I do have a dynamic ip address. My router ip is
Is it possible to set the attach= setting on a per user or per context
basis? We want to give our users the choice of no email notfiication, email
notification with no attachment, or notification with attachment.
Thanks,
Gary
___
Asterisk-Users
Yes in the voicemail.conf
Set the default to no and follow the instructions in the voicemail.conf file
i.e (not sure if this is exact
X = ,Joe Foo,[EMAIL PROTECTED],,||attach=yes
This mailbox protected from junk email by
On Thu, 5 Aug 2004 [EMAIL PROTECTED] wrote:
Date: Thu, 5 Aug 2004 14:56:06 -0400 (EDT)
From: Andrew McRory [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] PRI Errors... Ouch
Reply-To: [EMAIL PROTECTED]
Today is the first time I've had any show stopping errors with
I didnot use that magic name, because lately there was a big discution
regarding my post. I will try to be as precise as as I can now.
This is my sip conf file
[general]
externip=my DDNS Domain name
bindaddr = 0.0.0.0
port=5060
localnet=192.168.1.0/255.255.255.0
disallow=all
allow=ulaw
- Original Message -
From: Bartosz Wegrzyn [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, August 05, 2004 2:00 PM
Subject: Re: [Asterisk-Users] NAT problems
I didnot use that magic name, because lately there was a big discution
regarding my post. I will try to be as precise
After discussion I try the following without result:
DIALTEMPLATE
TEMPLATE MATCH=#... Timeout=5 User=Phone /
TEMPLATE MATCH=* Timeout=5 User=Phone /
/DIALTEMPLATE
Then I try, without result too:
DIALTEMPLATE
TEMPLATE MATCH=*# Timeout=5 User=Phone/
we are trying to use asterisk for converting SIP to H.323 calls.
asterisk (0.9.1) runs on the same linux (Redhat 8) box of our gatekeeper
(gnugk version 2.0.8).
the calls are going out through a cisco gateway.
when I make a call from a SIP phone to a PSTN number reachable through the
cisco
Hello list.
I've been running a Cisco VOIP WAN for the last two years using CallManager V3.2. The
remote 1760 routers are set up as H323 gateways. I've got a VG200 that is my PSTN
gateway with a full T1 voice trunk to the local CO. I'm also running Unity v3.x for
voicemail.
We are
Thanks for the helpful information.
I must say that although I was using the port forwarding I had the
nat=yes set on. I did that due to the fact that my asterisk didnot work
without this setting turned on. I dont know why. Also, I was told by
other people that this must be on even with port
Looks like it worked for 5 minutes.
After one of the test calls it stoped working.
Then I was keep trying and trying and the console was showing still
the magic IP number after the SIP/. First time it changed for broadvoice
IP it started to work back again.
Here I am lost, because I dont know if
- Original Message -
From: Bartosz Wegrzyn [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, August 05, 2004 3:18 PM
Subject: Re: [Asterisk-Users] NAT problems
Looks like it worked for 5 minutes.
After one of the test calls it stoped working.
Then I was keep trying and trying
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