Hi folks,
Am new to asterisk pbx systems.
I am trying to figure out what to do, I'll list and folks feel free to
give feedback and advice.
MAIN purpose for usage:
1.exposure to setup an asterisk box
2.get home phone service via VOIP/internet connection.
tasks so far
On Mon, 20 Oct 2008, Mike wrote:
Hi folks,
Am new to asterisk pbx systems.
I am trying to figure out what to do, I'll list and folks feel free to
give feedback and advice.
MAIN purpose for usage:
1.exposure to setup an asterisk box
2.get home phone service via
On Mon, Oct 20, 2008 at 8:42 AM, Mike [EMAIL PROTECTED] wrote:
Am new to asterisk pbx systems.
Hi Mike,
Welcome to the wonderful world of asterisk!
I am trying to figure out what to do, I'll list and folks feel free to
give feedback and advice.
You don't mention if you have read one of the
What country are you in? This is a truly global marketplace and mailing
list. We have people from the UK, Ireland, Oztrailia, New Zealand,
Bolivia, Russia, China, India, Argentina, etc. All over the world, really.
Saying what country you need the DID/DDI in will narrow it down somewhat.
I am
On Mon, Oct 20, 2008 at 04:28:30PM +1300, CSB wrote:
I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having
an annoying issue with the FXO ports. As soon as I plug either one into the
phone line it's as though the line is disconnected i.e. get disconnected
tone when trying
On Oct 19, 2008, at 5:35 PM, Edgar Guadamuz wrote:
I enabled the subscribe_event in the ais.conf and restarted aisexec.
After that I restarted asterisk and the only warning I got in
console was
Oct 11 6:38:04.340485 [CLM ] nodeget: trying to find node
If I disable the
Hi,
I had this problem once before.
It was related to running asterisk as a non root user.
I think if you run asterisk as root your problems will
go away.
OR
you can change the permissions on the /var/run
directory to 777 and the
problem might be solved too.
Regards,
Antoine Megalla.
On Mon, Oct 20, 2008 at 05:36:55AM -0700, Antoine Megalla wrote:
Hi,
I had this problem once before.
It was related to running asterisk as a non root user.
I think if you run asterisk as root your problems will
go away.
OR
you can change the permissions on the /var/run
directory to 777
Dear All,
I am trying to setup an ISDN line from local telco on a digium card. The
problem I am facing is that I am not getting any caller id from the
telco. They say that they have enabled caller id.
Please help me out.
My zapata.conf
A clearer explanation of your problem, including examples and output, is
needed.
Anael DIAZ wrote:
Hi!
I have some problem in my asterisk 1.4.2, I've installed it on centOS 5.2
and this didn't accept voip QoS and can't route the packets having voip
QoS.
So I should change voip packets to
Dear Everybody,
I have to store variable from ${CALLINGPRES} and get birth date of our
client and get back to him his birth prediction as numerology
(numerology digit value is between 1-9). I have also mentioned below
example here suppose client's birth date is 27-01-2000 then
Tzafrir Cohen wrote:
On Fri, Oct 17, 2008 at 05:04:32PM -0400, Gleim, Jason wrote:
I am using Asterisk and an X101P card as a glorified answering
machine.
We have a residential PSTN line with about six phones connected to it.
Like an answering machine, I want Asterisk answer
Hi all,
I'm trying to get a digium B410P to work with asterisk 1.6.0.1 (but i
have the same problems with asterisk 1.6.0)
Official digium documentation cover up to asterisk 1.4.x and suggest to
use zaphfc... but as i understood zap is completely gone in 1.6
I tried with misdn (1.1.8, 1.1.7.2
On Mon, Oct 20, 2008 at 12:25 AM, Eric ManxPower Wieling
[EMAIL PROTECTED] wrote:
ast_request: No channel type registered for ''SIP'
Notice the extra ' in the message.
That is either an error in the error message or you have a an extra ' in
your Dial line. Something like Dial('SIP/
On Mon, Oct 20, 2008 at 12:23 AM, Juan Rodríguez [EMAIL PROTECTED] wrote:
The second call its OK, so the problem it is just with the Dial(SIP/102), so
try:
originate SIP/102 application Dial SIP/102
and
originate SIP/101 application Dial SIP/102
and
originate SIP/102 application Dial
I do not think NAT is the problem, NAT normally gives you problems like one
way audio or no registration.
Try calling the SIP/102 on other extension:
;TEST
exten = 1002,1,Dial(SIP,102|20)
exten = 1002,n,Hangup()
instead of:
exten = 102,1,Dial...
But this is a very strange error... Check if
On Mon, Oct 20, 2008 at 10:37 AM, Juan Rodríguez [EMAIL PROTECTED] wrote:
I do not think NAT is the problem, NAT normally gives you problems like one
way audio or no registration.
Try calling the SIP/102 on other extension:
;TEST
exten = 1002,1,Dial(SIP,102|20)
exten = 1002,n,Hangup()
Any updates? It still seems to happen, though not as often as it used to.
We're using Polycom 320 phones, if that makes a difference, though we did do
it with X-Lite as well.
On Sat, Oct 11, 2008 at 3:03 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
Thanks, Steve,
That's what I am unsure of. I
Tried using GROUP()?
When a call comes in or goes out:
Exten = XXX,1,Set(GROUP(bdwi_out_1)=outgoing/incoming);
Exten = XXX,n,GotoIf($[${GROUP_COUNT(outgoing/[EMAIL PROTECTED])}] 1?fail)
Exten = XXX,n,Dial(...)
Exten = XXX(fail),1,Congestion();
Exten = XXX(fail),n,Hangup();
Obviously choose
some body can help me with astrisk server . i have problemes withthe message is notice [5483] : chan_iax2.c: 5325 register_verify : no registration
for peer 'x' from (xx.xx.xx.xx.)can you explan me wath's the master thank's __Do You
I tried using GROUP(), here's a snippet from the first post.
;Took out the Set(GROUP()) because I moved it elsewhere to try and fix
it added all the phones when Asterisk calls agents on a Queue.
[frombandwidth]
;exten = _+1.,1,Set(GROUP()=SIPGROUP)
exten =
My experience with GotoIf, what follows the ? has to be part of the extension
itself.
In your example:
Exten = _1NXXNXX(100) would be the intended target.
Maybe that's just 1.4 specific, I'll admit I haven't read this entire thread.
Also, use specific groups:
Set(GROUP(SIP)=SIPGROUP)
Eric Chamberlain schrieb:
Is there a way to override the fromdomain specified in the sip.conf
and instead set the value from the dialplan?
If we use:
Set(CALLERID(num)[EMAIL PROTECTED]
The SIP From header turns into:
[EMAIL PROTECTED]@10.10.10.10
Maybe you could abuse
Incoming calls ring SIP users who have |Ttr in their dial plan, but outgoing
calls are done through a macro as follows:
[macro-diallink2voip]
exten = s,1,Dial(SIP/[EMAIL PROTECTED],120)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-ANSWER,1,Hangup
exten = s-CONGESTION,1,Dial(SIP/[EMAIL
Something very similar had happened to our Polycom's. Somehow a qualify=yes
for all those peers seemed to solve it.
Try it if it's not enabled already.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill
Michaelson
Sent: October 18, 2008 3:16 PM
To:
Dear All,
I'm looking for someone who has implemented OpenR2 in Thailand
successfully. Any settings, advice, caveats etc. are welcome.
Best regards,
Peter Lindqvist
www.voxion.net
___
-- Bandwidth and Colocation Provided by
I'm trying to set the callerid(name) to Office for all calls from the
main office.
exten =s,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} =
0${REGEX(21245711*)} ] ? Office:${CALLERID(name)} )})
The main office callerid's are all 212 457 11xx. But this statement
seems to match everything,
The GotoIf works, because it does failover sometimes, just not all the
time, I followed instructions from here:
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf
And it seems to work in other areas that I use it in a similar way. I
only have the Set(GROUP()) when we are making outgoing calls on
On Mon, 2008-10-20 at 14:10 -0400, sean darcy wrote:
exten =s,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} =
0${REGEX(21245711*)} ] ? Office:${CALLERID(name)} )})
[snip]
What I'd expect is a callerid(num) of 2124571123 to generate an if test
of [02124571123 == 021245711*] or TRUE.
But
I have a macro to dial out, similar to yours in that it fails over to Zap/Dahdi
trunks in the event our bandwidth stuff is overloaded.
I run this in a macro, and only set and check groups within that macro. I'm
confused why yours would attach to phones in any way, unless you mean phone
to
I am sure this has been discussed prior, however, I am sitting here and
being asked this very question by my superiors. They are loving what I have
done with our two Asterisk servers here; however, they keep asking me if it
is secure or not. Of course, as with anything, I suspect that on a
VPN IP phone?
Then firewall up the asterisk to disable any outside access and place the
vpn server with the asterisk in a locked cabinet .
Sure that will stop someone trying to physically listen to their call.
Or they can always use the good old landline or mobile phone and let the
government
You can tell your superiors with great confidence that 99% of the issues
that fall under this conceptual umbrella have to do with the security of
your network, not of Asterisk the application, as is true of most other
security issues of concern to them.
With regard to call tapping, that is most
Well, when it fails over to the Dahdi trunk, it doesn't dial properly,
so I think I broke the macro. I will add the Set(GROUP()) stuff inside
of that macro-trunkdial-0.3 context and see if that helps. But it's
weird that I can't dial out. Here's a bit of the full log:
DEBUG[8221] app_macro.c:
lol
On Mon, Oct 20, 2008 at 3:34 PM, Sam Tam [EMAIL PROTECTED] wrote:
VPN IP phone?
Then firewall up the asterisk to disable any outside access and place the
vpn server with the asterisk in a locked cabinet .
Sure that will stop someone trying to physically listen to their call.
Or they
Hi
I'm trying to get the status of an extension that has DND set using the
service code, or trying to disable the service codes altogether so that
I can do them in the dialplan if needed
any advice wout be appriciated
Thanks
Robb
___
-- Bandwidth and
On Mon, Oct 20, 2008 at 9:20 PM, Jared Smith [EMAIL PROTECTED] wrote:
On Mon, 2008-10-20 at 14:10 -0400, sean darcy wrote:
exten =s,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} =
0${REGEX(21245711*)} ] ? Office:${CALLERID(name)} )})
[snip]
What I'd expect is a callerid(num) of 2124571123
A.R. Nasir Qureshi wrote:
Dear All,
I am trying to setup an ISDN line from local telco on a digium card. The
problem I am facing is that I am not getting any caller id from the
telco. They say that they have enabled caller id.
Tell them they are wrong. There is no calling party number IE
Anael DIAZ wrote:
Hi!
I have some problem in my asterisk 1.4.2, I've installed it on centOS 5.2
and this didn't accept voip QoS and can't route the packets having voip
QoS.
So I should change voip packets to be routing with centOS.
I want to use iproute2 but i don't what to do after
Jared Smith wrote:
On Mon, 2008-10-20 at 14:10 -0400, sean darcy wrote:
exten =s,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} =
0${REGEX(21245711*)} ] ? Office:${CALLERID(name)} )})
[snip]
What I'd expect is a callerid(num) of 2124571123 to generate an if test
of [02124571123 ==
sean darcy wrote:
OK. So I changed the * to .. , like so:
exten =s,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} =
0${REGEX(21245711..)} ] ? Office:${CALLERID(name)} )})
which I would expect to mean 021245711 followed by two other characters.
It still matches a blank callerid(num).
Atis Lezdins wrote:
On Mon, Oct 20, 2008 at 9:20 PM, Jared Smith [EMAIL PROTECTED] wrote:
On Mon, 2008-10-20 at 14:10 -0400, sean darcy wrote:
exten =s,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} =
0${REGEX(21245711*)} ] ? Office:${CALLERID(name)} )})
[snip]
What I'd expect is a
Now that I have a new card and my echo problems are 'mostly' solved, I
have another major issue to deal with. After about an hour or so the
card will stop detecting DTMF tones on incoming calls. dahdi_monitor
shows the following:
[EMAIL PROTECTED] wctdm24xxp]# dahdi_monitor 1 -v
Visual Audio
On Mon, 2008-10-20 at 16:17 -0700, Edwin Lam wrote:
sean darcy wrote:
OK. So I changed the * to .. , like so:
exten =s,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} =
0${REGEX(21245711..)} ] ? Office:${CALLERID(name)} )})
which I would expect to mean 021245711 followed by two
On Mon, Oct 20, 2008 at 7:38 PM, sean darcy [EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
On Mon, Oct 20, 2008 at 9:20 PM, Jared Smith [EMAIL PROTECTED] wrote:
On Mon, 2008-10-20 at 14:10 -0400, sean darcy wrote:
exten =s,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} =
0${REGEX(21245711*)} ] ?
hi,
for my multi-tenant pbx, i would like to approach prepaid like this:
when a customer dials number, i have an AGI that will determine what
country was dialed and retrieve the rate from the rate table,
once the rate is retrieved, i will get the remaining balance of that
customer nd compute
Alex Balashov schrieb:
SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do.
Well, i am not getting the correct meaning of 'defunct', but from the
last part of your suggestion i guess you value Kamailio/OpenSIPS more
than SER.
Are there some hard reasion for this.
I
No, the issue isn't my value or preference. The issue is that SER is no
longer maintained or developed and has not been for several years.
Tobias Wolf wrote:
Alex Balashov schrieb:
SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do.
Well, i am not getting the
Hi,
some comments to your prepaid approach...
If you have the oportunity to avoid share the same balance between more than
two users, please do it.
It is better to assign one record to each user, and store in it, his/her
balance.
Don't complicate a simple solution.
Another comments:
1.- The
Hi everyone,
I have encountered a hard problem that when i connect my anology phone
to channelbank ,I found that i dial a number and create the call,then ,I
hangup the call,but ,very quickly,I listen the ringing im my phone,I pick it
up ,and found it noting, anybody can tell me this
Alex Balashov wrote:
No, the issue isn't my value or preference. The issue is that SER is no
longer maintained or developed and has not been for several years.
The above statement is totally false. SER is indeed an ongoing project
which is actively maintained. If you subscribed to the
I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am
having
an annoying issue with the FXO ports. As soon as I plug either one into
the
phone line it's as though the line is disconnected i.e. get disconnected
tone when trying to dial out, line is busy when dialling in.
Err...
Hello Peter,
You can ask this better in the asterisk-r2 mailing list.
I don't know of anyone that has used OpenR2 in Thailand, but I am
interested in adding support for that variant. Contact me at this same
e-mail address or via Google talk (my e-mail address works for MSN as
well ) to discuss
I got Asterisk to work with R2 in Thailand about 3 years ago. This was
back before OpenR2, and I had to modify the C libraries directly. Of
course, outbound dialing is DTMF, not R2, however inbound is pretty
standard. This was on TA (now True) circuits.
I guess I can't tell you much about how
Hello Moises,
I will set up the asterisk-r2 mailing list.
As I have understood it Thailand is using the same version as China of R2.
Currently we are discussing with TOT (Telephone Organization of
Thailand), because I believe they are messing up on their side so we are
sorting that out.
We are using TOT (Telephone Organization of Thailand). They are very
messy on their side so we are sorting out some unrelated problems with
them right now - very slow response.
I believe you are correct when you say that outbound dialing is DTMF, I
have heard this before too.
Out of curiosity
Dear Matthew,
Thank you for your reply.
Please tell me if there is any way I can see the actual q931 message received
from the card without any translation / filtration / alteration by the software
or the driver ?
How much the switch type or other configuration variables affect the SETUP
Hi Peter,
Basically, I had to rewrite the R2 state machine to use pulsed outbound
signalling instead of compelled, and hack in support of the DTMF tone
groups instead of R2 frequencies. It was messy, but possible. The
inbound side was much cleaner, and you may be correct when you say you
There are no 100% solution but we can only do our best.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of broadband
Voice
Sent: Tuesday, October 21, 2008 4:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How
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