I need to be able to send an sms alert to one's mobile/cell phone. For
instance, when I receive a voicemail message in my inbox, I also want to
be able to get a message on my cell phone alerting me of this e-mail. How
possible is this? And if it is, what do I need to do to get the service up
and
I need to be able to send an sms alert to one's mobile/cell phone. For
instance, when I receive a voicemail message in my inbox, I also want to
be able to get a message on my cell phone alerting me of this e-mail. How
possible is this? And if it is, what do I need to do to get the service up
and
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
Good day all
We have a snom 220 that for some reason keeps on giving this message
Got SIP response 486 Busy Here back from 192.168.21.222
even though there is no active calls to it and there are 2 accounts set
on the phone?
Please Help and advice
Thanks
Altus
quote who=C F
Why not, just use the email address given before from you email
client.
So, you can generate an SMS message on your Cell phone and send it
to your, say, hotmail account? Or are you talking about using
an embeded email client on the phone to create an email. Not using
SMS at
quote who=Eric Wieling
Robert Hajime Lanning wrote:
um, backwards. E-Mail to SMS. I have not seen the other way
around.
Both Cingular and Verizon supports both.
I have not tried this, nor have I seen any documentation mentioning
it. Do you or anyone else have a pointer for the info?
Rich Adamson a écrit :
would like to know if some of you have tested asterisk connected to an
EADS 6550 analogique PBX (also know as Nexpan50).
Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no
other card, each of them have their own IRQ) all ports connected to the
EADS.
Title: Panasonic KX-TD1232
Hello,
I'm trying to connect an Asterisk server with the Panasonic KX-TD1232 Phone System. Is this possible? Which hardware do I need and which Asterisk configuration files should I adjust?
Dennie
Another option is to send sms by mail.
1-) You subscribe to an sms provider who can allow you to do mail2sms;
2-) You send sms message under the form [EMAIL PROTECTED] ;
3-) SMS provider receives SMS from you and will send it through its gateway;
Hope this helps
Quoting Marco Ziglioli [EMAIL
Hello,
Have a weird problem when using asterisk (1.0.6). There are certain
numbers I cannot dial when using asterisk with my broadvoice account.
No problems with inbound. With outbound calls, I can call some numbers
(for example broadvoice customer support number) and unsuccessfully with
some.
We do have that service here using [EMAIL PROTECTED] The
way it works is that I can do mail2sms and sms2mail.
What I would like to do is to have my * box send an sms to a cellphone,
that is, to say [EMAIL PROTECTED] where 0485.. is my cellphone number
and it.co.ug my sms provider/domain. This sms
On Fri, 2005-03-11 at 20:18, Kanishka Somaratne wrote:
Hi
Is there a billing system that i can view all the call taken by SIP
clients in asterisk
http://www.voip-info.org/tiki-index.php?page=Asterisk+CDR+Areski+GUI
__
I am looking at the possibility integrating Asterisk with our
current Mitel 200sx. If this is possible what physical connection
is made between the Mitel box and * box? Then can a user choose
if a call is go out VoIP or not?
I'm more familiar with the SX2000 family rather than the 200 series,
Have a look at this site
http://www.bayhamsystems.com/asterisk.html
It was easy to install and the example works fine. No word on
commercial pricing yet but you can test in the meantime.
Ian
Julius Kidubuka wrote:
I need to be able to send an sms alert to one's mobile/cell phone. For
instance,
On Mon, 14 Mar 2005, Dennie Verstrepen wrote:
I'm trying to connect an Asterisk server with the Panasonic KX-TD1232
Phone System. Is this possible? Which hardware do I need and which
Asterisk configuration files should I adjust?
Yes, it is possible. How it is done depends on what interfaces
Hi
after upgrade from R2 to 1.0.6 , my dtmf not working and i cant dial . can
any 1.0.6 user help me why i cant do that.
Bashir
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To
Hi Peter,
I'm looking at a similar set up using a GHX1232 but I can't find a
single refence or docmentation for a GHX1232 anywhere though, and I'm a
bit of a newbie to this game. Do you know if it would take a similar
approach to integrate asterisk into that system?
Thanks
John
-
Peter
Hello!
Looks like h323 compiling is FAQ, but I didn't found an answer...
The same problem with 0.6.5 and 0.7.1:
gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE
-I/var/local/files/asterisk-1.0.5/include -I../wrapper -g -c -o
On Sun, 13 Mar 2005 21:49:52 +0200, Dimitris Kounalakis
[EMAIL PROTECTED] wrote:
Hello *Martijn,
Thank you for your response.
*That was my opinion too, it looses the context due to a bug, and can anyone
confirm it also?
But I have no output from the command Show channels, and it happens so
On Sun, Mar 13, 2005 at 07:22:58PM -0600, Anton Krall wrote:
So far nobody has answered this post... Anybody has seen this error before?
Could you use a more verbose logging?
IIRC, the technology is the channel type, e.g: sip, zap, iax.
Somewhere something is getting either an empty channel
Hi to all,
I have a problem with this wildcard and one E1 line.
The server is a Asus P4P800S with the 865PE chipset,
and 512MB RAM. The kernel version is 2.4.26.
I have donwload and build the latest CVS version of
zaptel, libpri and asterisk following the ufficial instructions of
Hi there
Just a quick question, I will be building some servers in a lab utilizing
Digium E1 cards. I would like if possible to avoid the expense of installing
an e1/ISDN30 in my lab. I have two questions really, first does anybody know
of an effective simulation tool I can use to replicate a
Hi to all,
I have a problem with this wildcard and one E1 line.
The server is a Asus P4P800S with the 865PE chipset, and 512MB RAM. The
kernel version is 2.4.26.
I have donwload and build the latest CVS version of zaptel, libpri and
asterisk following the ufficial instructions of digium.
I have
Hi
I am having alot of difficulty connecting to SIP providers (I am trying 3)
and can't seem to find anything similar in the wiki or on the lists.I
can receive inbound calls fine however on placing an outbound call the
calling phone never gets 'connected' but 2 way audio is passed for about
Master Abi [EMAIL PROTECTED] wrote:
Not having a backlit display is bad design.
Actually it is a feature issue, not a design issue. :)
Doug
--
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com
Hello!!
Please help me with next problem...
While traying to read voicemail system plays all service messages
and then hang upthe line...
console display next:
Mar 14 14:16:02 WARNING[135271424]: file.c:1004 ast_waitstream_full: Wait
failed (No such file or directory)
Asterisk runing on
Hi
Good Morning,
I am having alot of difficulty connecting to SIP providers (I am trying 3)
and can't seem to find anything similar in the wiki or on the lists.I
can receive inbound calls fine however on placing an outbound call the
calling phone never gets 'connected' but 2 way
hello list,
i searched for nearly a week for a solution to this problem, as there is:
analog fax machine -» grandstream ata -» asterisk -» sip trunk from provider
-» provider gateway to pstn -» analog/isdn fax machine on pstn
everything worked out fine until my provider decided to implement
Hi, Dennie,
Yes, it is possible. Specially that model has DTMF Inband signaling,
in other words you can send dtmf tones to asterisk when using it as a
voicemail so that it knows what extension did not answer the call and
can thus be directed to the right voicemail.
You need to play with the
On Mon, 14 Mar 2005, John Brennan wrote:
I'm looking at a similar set up using a GHX1232 but I can't find a
single refence or docmentation for a GHX1232 anywhere though, and I'm a
bit of a newbie to this game. Do you know if it would take a similar
approach to integrate asterisk into that
On Mon, 14 Mar 2005, Brett, Gary wrote:
Just a quick question, I will be building some servers in a lab utilizing
Digium E1 cards. I would like if possible to avoid the expense of installing
an e1/ISDN30 in my lab. I have two questions really, first does anybody know
of an effective
Thats cool Peter, thanks,
Has anyone on the list ever heard of a Goldstar GHX-1232? It seems to be
a bit of a dinosaur. I'm hoping it might be a rebranded device though
and someone might be able to point me in the right direction for
documentation?
Thanks in advance.
On March 14, 2005 06:50 am, pixer wrote:
3: 0 XT-PIC t4xxp
Without loading the module the LED glows in red colour, but the moment we
load module, it goes off. (No red or green).
We ran zttool and tried to run a loop test, but zttool simply hung with the
message 'Looping
On March 14, 2005 06:43 am, Brett, Gary wrote:
Just a quick question, I will be building some servers in a lab utilizing
Digium E1 cards. I would like if possible to avoid the expense of
installing an e1/ISDN30 in my lab. I have two questions really, first does
anybody know of an effective
check your entensions.conf file /etc/asterisk/extensions.conf
. ${ETEN:${TRUNKMSD}})
we had same problem this was the fix
- Original Message -
From: Matthew Boehm [EMAIL PROTECTED]
To: Asterisk Users asterisk-users@lists.digium.com
Sent: Sunday, March 13, 2005 1:23 PM
Subject:
Hi!
First of all , (apart from solving your problem) you really should get
rid of the whole [demo] context from extensions.conf, and place your
stuff in your own context (i.e. [local]) (just for convenience and
security). Getting back to the problem - as I see it you want to dial
out through Cisco
The cross-over cable is what I do between by Asterisk and my Lucent
PBX's. Works great!!
Peter Svensson wrote:
On Mon, 14 Mar 2005, Brett, Gary wrote:
Just a quick question, I will be building some servers in a lab utilizing
Digium E1 cards. I would like if possible to avoid the expense of
Just in my dial plan. I am not using any real Lease cost routing package, as
a matter of fact I am developing one but it's not ready yet.
W
- Original Message -
From: Robert Augustyn [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
For those that are interested, I was just out on the Cisco site and
noticed that they had released firmware 7.4 as of March 11th for the
7940/7960 phones.
Doug
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Hi,
I followed the instructions on http://www.asterisk.org/index.php?menu=download.
I picked the latest version using CVS.
Things went fine until I cd zaptel ; make clean ; make install.
I then get an error when compiling zaptel.c
/usr/src/linux/include/linux/kernel.h:75: error: parse error before
Yes, I've seen it already, but it's not really as user-friendly as
sjphone.
In firefly, you cannot even paste the phone number in.
Any other ideas?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Anton Krall
Sent: Montag, 14. März 2005 02:21
[EMAIL PROTECTED] wrote:
I followed the instructions on
http://www.asterisk.org/index.php?menu=download.
I picked the latest version using CVS.
Things went fine until I cd zaptel ; make clean ; make install.
I then get an error when compiling zaptel.c
Title: Message
Too
hard to say. My problem is with a Channel bank, if it made it any better then
it's very little.
-Original Message-From: Stuart Hirst
[mailto:[EMAIL PROTECTED] Sent: Wednesday, March 09, 2005
2:51 PMTo: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Robert Hajime Lanning wrote:
quote who=Eric Wieling
Robert Hajime Lanning wrote:
um, backwards. E-Mail to SMS. I have not seen the other way
around.
Both Cingular and Verizon supports both.
I have not tried this, nor have I seen any documentation mentioning
it. Do you or anyone else have a
Roman Zhovtulya wrote:
Yes, I've seen it already, but it's not really as user-friendly as
sjphone.
In firefly, you cannot even paste the phone number in.
Any other ideas?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Anton Krall
Sent: Montag, 14.
I had this issue. I configured my sip phones to use rfc2293(?) instead of inband. Note:the rfc number is incorrect but I don't feel like looking up the correct one right now. Just look in sip.conf example and it will tell you the right number.
On Mon, 2005-03-14 at 04:51, Bashir Ullah -
WARNING[9013]: pbx.c:1554 pbx_extension_helper: No application
'VoiceMailMain2' for extension (local, 225, 1)
I see now that VoiceMailMain2 has been depreciated
/VoiceMail is now replaced by VoiceMail2 in the CVS, so voicemail2 will
be obsolete soon. The old voicemail is not included in the
On Monday 14 March 2005 08:50, Altus Snyman wrote:
Good day all
We have a snom 220 that for some reason keeps on giving this message
Got SIP response 486 Busy Here back from 192.168.21.222
even though there is no active calls to it and there are 2 accounts set
on the phone?
Either someone
Hello!
I'd like to Flash() a modem line (BRI) with Asterisk. It is a
passive ISDN-card connected to a hardware PBX. I use ISDN4Linux.
I recognised that unfortunately the Flash() application flashes
Zap devices only. Now I am wondering how I could flash Modem/ttyI0.
The source code chan_modem.c
On Mon, March 14, 2005 17:06, Andres said:
You might want to try the steps provided above yourself Peter. Because
even if we have a context that leads to never never land at the top of
sip.conf, I am still able to make free calls. A sip debug clearly
Welcome to the wonderful world of
I need to be able to send an sms alert to one's mobile/cell phone. For
instance, when I receive a voicemail message in my inbox, I also want to
be able to get a message on my cell phone alerting me of this e-mail. How
possible is this?
This is probably cheating: I have a free email account
Title: Message
I have done
some research on the discussions that have occured on this list about
DS3s with Asterisk.
It seems to be
dead and I have not found any active work on the
project.
I know that a
full DS3 may have some technical limitations with why they may not work with
I have works Panasonic TD1232 with asterisk!
In my office have TD1232 with one expanssion!
So i have 64 extenssions, i bought a TDM04B card and connect 4
extensions of TD1232 at card.
I Configure the number 8 to give me a tone of whatever 4 lines(g1)
and go to internet at other asterisk
hi list.
i have the following problem.
if i dial an ip endpoint from my ip phone and the endpoint is busy, in
my cdr i see (answered). I think there must be busy.
why is that? any hints?
thx,
thomas
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hallo,
i trying to dial with a python script via the manager interface, it works
ok but i would like to send a soud file name as a variable to the dialplan,
so that i can call a number and send it a different soundfile i choose in
my pyton script.
the problem is, that the * dials correct and
look at this thread:
http://lists.digium.com/pipermail/asterisk-users/2005-March/094509.html
On Mon, 14 Mar 2005 11:09:11 +0300 (EAT), Julius Kidubuka
[EMAIL PROTECTED] wrote:
I need to be able to send an sms alert to one's mobile/cell phone. For
instance, when I receive a voicemail message in
That looks really much closer to wht I'm looking for, but it doesn't
seem to support SIP (at least Windows version).
When I go to add an account, it gives only ICQ, Yahoo, MSN and a couple
of others.
What I would like to have is to connect it to my Asterisk via SIP.
Is there any other way/any
I am trying to get more familiar with ASTCC, but I miss some tools, but
I believe somebody has already thought about it:
1. I would like to send a standard letter to the users, as soon their
balance drops below a certain value. E.g., Dear user, you have only 1.-
left, consider to fill
Take a look at this one:
http://lists.digium.com/pipermail/asterisk-users/2005-March/094509.html
You can also enable call back in voicemail.conf
On Mon, 14 Mar 2005 11:07:49 +0300 (EAT), Julius Kidubuka
[EMAIL PROTECTED] wrote:
I need to be able to send an sms alert to one's mobile/cell phone.
I am checking on the SetGroup / CheckGroup commands, but I have some
troubles to undestand the examples.
SetGroup(moh) can be moh anything as I like? Usually moh stands for
music on hold
CheckGroup(1) checks if somebody in in group moh. Does it mean I can
only have one SetGroup(xxx) ??
I have connected the KX-1232 to asterisk with the T1 card.
Is it dissappointing though as I have not gotten any Caller id
information running over the T1.
But it does function.
Jerry
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-Original Message-
From: Julius Kidubuka [mailto:[EMAIL PROTECTED]
I need to be able to send an sms alert to one's mobile/cell phone. For
instance, when I receive a voicemail message in my inbox, I
also want to be able to get a message on my cell phone alerting me of this
e-mail.
I'm going to do a deployment of LocustWorld MeshBoxes in some of our remote
locations. Build 90 comes with Asterisk 1.0, and our plan is to use the
MeshBoxes as a WAP for non-Asterisk uses but also to add a 2nd NIC to deploy
Snom's in the remote location. This works fine (was suprisingly easy to
* Mateo Meier [EMAIL PROTECTED] [2005-02-28 09:13]:
I tryed that with capi.. but no luke. It will hang up the line anyway :-(
exten = s,1,Playback(transfer)
exten = s,2,Flash(capi/72044**:041720,18)
exten = s,3,SendDTMF(${ARG1})
exten = s,4,Hangup()
Any idears why ?
No wonder! The
Raoul Bönisch wrote:
Hello!
I'd like to Flash() a modem line (BRI) with Asterisk. It is a
passive ISDN-card connected to a hardware PBX. I use ISDN4Linux.
I recognised that unfortunately the Flash() application flashes
Zap devices only. Now I am wondering how I could flash Modem/ttyI0.
The source
Colin Anderson wrote:
I'm going to do a deployment of LocustWorld MeshBoxes in some of our remote
locations. Build 90 comes with Asterisk 1.0, and our plan is to use the
MeshBoxes as a WAP for non-Asterisk uses but also to add a 2nd NIC to deploy
Snom's in the remote location. This works fine (was
On 07:46 PM 3/13/2005, Brett, Gary wrote:
Hi there
Just a quick question, I will be building some servers in a lab utilizing
Digium E1 cards. I would like if possible to avoid the expense of installing
an e1/ISDN30 in my lab. I have two questions really, first does anybody know
of an effective
On Mar 14, 2005, at 5:20 AM, Doug Lytle wrote:
For those that are interested, I was just out on the Cisco site and
noticed that they had released firmware 7.4 as of March 11th for the
7940/7960 phones.
I don't see any major changes in the release notes--mostly small bug
fixes. They fixed some
Get an M13 from adtran and split it. You could also get a Cisco AS5400
Michael Blood wrote:
I have done some research on the discussions that have occured on this
list about DS3s with Asterisk.
It seems to be dead and I have not found any active work on the project.
I know that a full DS3 may
FYI, spoke with Grandstream this morning, the GXP-2000 release has been
delayed again. Looking like April now before these hit the street.
--
Cory Andrews
Senior Partner
VOIPSupply.com
+
V: 800.398.VOIP X22
F: 716.630.1548
E: [EMAIL PROTECTED]
Hi,
I have the same problem with cvs head. (1.0.6)
See http://www.inaccessnetworks.com/projects/asterisk-oh323
And https://skylab.inaccessnetworks.com/mantis/view_all_bug_page.php
(issue 00...008)
some 'patch' files are included.
I am a newbie to linux and asterisk.
I do not want to blow my
INSERT INTO sip_buddies VALUES
(1,'621',NULL,NULL,NULL,'\Demo\,621','yes','inhouse',NULL,'rfc2833',NULL
,N
ULL,'dynamic',NULL,NULL,NULL,NULL,'[EMAIL
PROTECTED]',NULL,'yes',NULL,NULL,NULL,'1',
''
,'999',NULL,NULL,NULL,'Password','friend','621','ulaw;alaw','all',NULL,0,'',
'y
Try using commas
On Mon, 14 Mar 2005, Jerry Geis wrote:
I have connected the KX-1232 to asterisk with the T1 card.
Is it dissappointing though as I have not gotten any Caller id
information running over the T1.
But it does function.
We have callerid working with that setup (well, actually an E1). You can
Matthew Boehm wrote:
INSERT INTO sip_buddies VALUES
(1,'621',NULL,NULL,NULL,'\Demo\,621','yes','inhouse',NULL,'rfc2833',NULL
,N
ULL,'dynamic',NULL,NULL,NULL,NULL,'[EMAIL PROTECTED]',NULL,'yes',NULL,NULL,NULL,'1',
''
The H0,H1 timing may be tricky, but, If the modem is AT compliant, ATD! is
the flash command. The timing is based on S register 29.
- Original Message -
From: Raoul Bönisch [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, March 14, 2005 10:50 AM
Subject:
Hi!
http://www.voip-info.org/wiki-Asterisk+addon+rate-engine
or write you own AGI :) (see ASTCC)
Best regards,
Madhawa
On Fri, 11 Mar 2005 19:18:24 -, Kanishka Somaratne
[EMAIL PROTECTED] wrote:
Hi
Is there a billing system that i can view all the call taken by SIP clients
in
I have configured asterisk to act as a B2BUA, so
can use ser for sip proxying and forward the call to a sip
provider.
The problem now is that when I place a call to an
outgoing number I don't hear nothing up to the time the callee
responds.
The first time I configured asterisk it was
Well, I'm just leaving demo in for testing. Once I get things working
I'll be changing all that to city names most likely.
I don't want the call to it the Cisco then redirect to the Asterisk
box. If I hit extension 602 right now, it works fine. What I'm trying
to do is dial out to another real
* Stu Gotz [EMAIL PROTECTED] [2005-03-14 16:56]:
The H0,H1 timing may be tricky, but, If the modem is AT compliant, ATD! is
the flash command. The timing is based on S register 29.
Yes, that's another possibility. We're close to it. I have the
idea of using the System() application to call a
Hello,
I'm trying to run an ATA behind a NAT device, and am confused on exactly
what the qualify config option does, other than send NOTIFY packets.
Outbound calls work fine, but inbound calls do not go through. With
qualify=yes and nat=yes, my show sip peers looks like:
Hi All,
Does anyone know the amount of memory used by skype?
regards
César
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Brian McCrary wrote:
Hello,
I'm trying to run an ATA behind a NAT device, and am confused on exactly
what the qualify config option does, other than send NOTIFY packets.
Outbound calls work fine, but inbound calls do not go through. With
qualify=yes and nat=yes, my show sip peers looks like:
Hi Mike:
I've been searching for something like this script to solve my wmi
problem. I used the script and it works fine only if the user leaving a
message press the # key or when I search my voicemail I leave the Main
with the # key. If me or the user (caller) leaving the message hang up
Hi
I need to know if Asterisk supports the full features of the SIP REFER message
(i.e blind and supervised transfers).
I'm trying to do a supervised transfer through Asterisk from a VoiceXML
application using the transfer tag and setting bridge=true (i.e transfer
name=transfer1 bridge=true
Sorry everyone, I know this has been hashed over a bunch of times but I
can't find anything that pertains to specific cracking and popping on the
FXO modules of a TDM04. This happens on inbound or outbound calls. This is
the first install I have done with a TDM card for FXO modules so please, be
Sorry everyone, I know this has been hashed over a bunch of times but I
can't find anything that pertains to specific cracking and popping on the
FXO modules of a TDM04. This happens on inbound or outbound calls. This is
the first install I have done with a TDM card for FXO modules so
FYI, you can stop using that script and start using the RealTime cache
ability.
-Matthew
Jose R. Ortiz Ubarri wrote:
Hi Mike:
I've been searching for something like this script to solve my wmi
problem. I used the script and it works fine only if the user
leaving a message press the #
Hello,
I wonder if I would have to sacrifice anything if I set NAT=yes for
all sip clients I have, regardless of whether they are behind the NAT or
not.
The idea is to have the setting that works regardless of whether the
user is behind the NAT or not, since I'm not sure what connection that
That was already with SET VERBOSE 255.
/rg
On Mar 14, 2005, at 3:32 AM, Tzafrir Cohen wrote:
On Sun, Mar 13, 2005 at 07:22:58PM -0600, Anton Krall wrote:
So far nobody has answered this post... Anybody has seen this error
before?
Could you use a more verbose logging?
IIRC, the technology is the
Is there any peformance problems/etc if I set NAT=yes for all clients?
nat=yes causes Asterisk to respond to the *public* source port and IP
address. Therefore, the only time you should ever have a problem is when
the packets should not go to that port/address, which I think is close
to never.
Thanks!
Could that mean any security problems?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Nabeel Jafferali
Sent: Montag, 14. März 2005 19:19
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Setting
Matthew Boehm wrote:
FYI, you can stop using that script and start using the RealTime cache
ability.
Thanks for the hint. But where can I find the cache information? I
search at www.voip-info.org and couldn't find anything.
Thanks!
JO
-Matthew
Jose R. Ortiz Ubarri wrote:
Hi Mike:
Roman Zhovtulya wrote:
Hello,
I wonder if I would have to sacrifice anything if I set NAT=yes for
all sip clients I have, regardless of whether they are behind the NAT or
not.
The idea is to have the setting that works regardless of whether the
user is behind the NAT or not, since I'm not sure
My school district will be building a new elementary school in 2006. We
were about to go to bid with a traditional intercom system for the
campus but I would like implement Asterisk at the campus.
My question is, do we build in a traditional intercom/paging system and
tie that into the
I think [EMAIL PROTECTED] will work well for what you want
to do. The GUI also allows you to edit the config
files. it just saves time it dosn't reduce the
functionality of Asterisk.
if you get the IM callbacks feature working I would be
interested. This would be a great feature to include
in
You can find the cache information in the sip.conf inside
/usr/src/asterisk/configs/ (or whereever you keep your source).
-Matthew
Jose R. Ortiz Ubarri wrote:
Matthew Boehm wrote:
FYI, you can stop using that script and start using the RealTime
cache ability.
Thanks for the hint. But
Has anybody tried NVFaxDetect Fax detection for sip SIP/IAX channel?
There is a new application from Newman Telecom for fax detection.
http://www.sineapps.com/news.php?rssid=575
Current Asterisk Fax detection doesn't work for me as I don't have
Digium cards; I'm using Siupra
--
#Joseph
On Mon, 2005-03-14 at 14:30 -0300, César Davi Ávila do Nascimento wrote:
Hi All,
Does anyone know the amount of memory used by skype?
Did you think about the best venue to ask this question. We are not a
skype support forum.
And BTW, TURN OFF HTML EMAIL. Exercise some thought about those who
On Mon, 2005-03-14 at 10:46 -0800, Chris Hobbs wrote:
My school district will be building a new elementary school in 2006. We
were about to go to bid with a traditional intercom system for the
campus but I would like implement Asterisk at the campus.
My question is, do we build in a
Hi,
Thanks Andreas !
On my colinux I add to dig into the colinux source code to
extract the .config file (it was missing from /root on
my install). After that, I did make-kpkg as explained
in your page. Then, the compilation error disappeared.
The whole issue was about some missing kernel include
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