Hi All,
I was trying to test to send a fax to an international number.
Here's the setup:
FAX -- HT486 -- SIP PROXY -- GATEWAY -- PSTN -- FAX
Unfortunately I haven't been able to do it, I read somewhere that fax uses
G711 only, is this true? because our gateway provider uses only G729. does
Hi,
Has anyone implemented astpp? I'm configuring one right now and I have a
problem on the pricelist.
I followed the steps here
http://www.astpp.org/index.php?n=ASTPP.Installation and created tables
using http://www.astpp.org/index.php?n=ASTPP.Structure, but i didn't see
there a query on
Hi Sir,
My problem is when I click on pricelist, i have an error there's
something wrong on the pricelist database.
When I looked at the database and search for a table called pricelist
there's nothing there. I foolowed the querires on the the structure but
also found any query that creates
Hi All,
Can't explain what happened, last night i was setting the voicemail
configuration, and it worked properly:
-- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/1000100-08219db0,
@VM-1000) in new stack
-- SIP/1000100-08219db0 Playing 'vm-login' (language 'en')
i can hear the
Hi,
For now i just turned off acpi. and it works now.
just dont know what's the connection of that though
:-)
i will try to do the things you guys suggested also
when i get the chance, thanks for you help!
regards,
nhadie
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sun, Mar 30, 2008 at
Hi,
Is it possible for me to detect fax on a sip trunk?
my provider has a fax service that can send/receive
fax.
is it possible that i use a that trunk as a telefax?
meaning i will try to detect if it's a fax, if it is i
will forward it to an extension that can handle fax if
not will forward it
Hi All,
I just started playing around with asterisk realtime,
added some extensions and started making test call,
sometimes i can call the extension sometimes i can't.
below are errors i see on the CLI, has anyone
encountered this before?
[settings]
sippeers = mysql,sipdb,sip_customer
sipusers
Hi All,
I'm trying to configure a ringgroup, which will ring the extension in the
group one by one. this is what i tried on my extension.conf
[macro-dial-ringgroup]
exten = s,1,Dial(SIP/${ARG1},15)
exten = s,n,NoOp( Dial Status: ${DIALSTATUS})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten
Hi All,
I'm tryng to test different scenarios for followme for different users:
[localext]
exten = 101,1,Set(FM = ALWAYS);
exten = 101,n,Macro(dial-ext|SIP/${EXTEN}|vm-1|moh-101|fm-101);
exten = 101,n,Hangup
exten = 102,1,Set(FM = NEVER);
exten =
Hi,
Would just like to ask about cdr, i have an asterisk and i would like to bill
only outbound calls not extension to extension, when i'm looking at the CDR, i
can't figure out which fields i need to filter all outbound calls only.
e.g if i dial 00. or 9XX (for local pstn calls) those
hi all,
we recently bought a clone box, motherboard with ICH7R raid controller (which i
thought was a hardware raid controller). but recently i learned that those
things are called FRAID( Fake RAID) which is basically a software raid also. so
i decide to just use Software RAID (using CentOS
Hi,
I recently installed asterisk, i used sterisk-1.4.20.1, i i set directrtpsetup
to yes, no whow would i know if the rtp/media is not passing to asterisk. any
tool or can u just sniff?
regards,
ron
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Hi,
I have this dialpan to call international:
exten =gt; _00.,1,SET(TIMEOUT(absolute)=300)
exten =gt; _00.,n,Dial(SIP/[EMAIL PROTECTED])
exten =gt; _00.,n,NoCDR()
exten =gt; _00.,n,Hangup
Is there a way to check if there is only 1 minute remaining on the absolute
timeout?
also an additional
Hi,
Hope anyone can help me on DUNDi. I got 2 asterisk servers. configs below.
tried this on the cli:
*CLI dundi lookup [EMAIL PROTECTED] bypass
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
*CLI dundi lookup [EMAIL PROTECTED] bypass
DUNDi lookup returned no results.
DUNDi
Hi,
Would just like to know if it's possible to be able to call a macro at the same
time.
i use a macro to dial local extension to another extension.
exten = 100,Macro(dial-ext|SIP/100)
exten = 101,Macro(dial-ext|SIP/101)
but now i would like to use it on a simple ringgroup where it will
/100)
Set(DIALGROUP(test,add)=Local/101)
Dial(${DIALGROUP(test)})
ronald ramos wrote:
Hi,
Would just like to know if it's possible to be able to call a macro at
the same time.
i use a macro to dial local extension to another extension.
exten = 100,Macro(dial-ext|SIP/100)
exten = 101,Macro
Hi,
I'm not sure if this is the proper way to approach it but i can't figure out
how to setup dundi.
what i did is, i try to determine which server a user is registered, by calling
an agi to query the realtime db and capture the regserver of the user.
e.g.
exten =
Hi,
how would i know what codec is being utilized? currently i have set allow=ilbc
disallow=all.
i unset all codecs on x-lite except ilbc.
i tried to make a call and look at the channel i see these. does this mean it
is using ulaw? how about writetranscode? does that mean there is no
Hi,
I have setup 2 asterisk talking a single mysql cluster. I'm also using
realtime db. I've setup sip peering between the two asterisk servers.
[asterisk-1]
insecure=port,invite
type=peer
host=201.202.203.204
context=from-asterisk-1
[asterisk-2]
insecure=port,invite
type=peer
Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2 asterisk.
I copied the config from DUNDI enterprise SIP with no password. Only thing i
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv
PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 6:23 PM
Ron,
What does the peers section in dundi.conf look like?
On Tue, Aug 26, 2008 at 3:00 AM, ronald ramos [EMAIL PROTECTED]
wrote:
Would like to try setting up
of the registrations and any restarts on the asterisk process
it may take some time for phones to re-register.
On Tue, Aug 26, 2008 at 2:10 PM, ronald ramos [EMAIL PROTECTED]
wrote:
Hi Bruce,
my apologies, but the error was because of the key.
i just run keys init on the CLI and it works,
question
],nopartial
priv = dundi-static,0,SIP,[EMAIL PROTECTED],nopartial
On Wed, Aug 27, 2008 at 3:56 AM, ronald ramos [EMAIL PROTECTED]
wrote:
Hi Again,
Is there a way i can detect whether a user has been added into the
regcontext?
Currently i'm seeing this and just gives a fast busy.
[Aug 27 16:44:46
hi,
when a user register on my asterisk i can see it adding Noop for that
extension, but after awhile i won't see it anymore:
what are the reasons for it being removed on the dynamic context?
one thing i found when i unregister it's removed.
dialplan show myregcontext
[ Context
hi,
is it possible to store the IP address of the caller in the CDR? how about the
end date/time? thank you.
regards,
ron
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AstriCon 2008 - September 22 - 25 Phoenix,
Hi All,
Any solution on how I can implement prepaid billing on asterisk?
But not the calling card type, just a simple Custome rwill buy credit,
consume then buy again.
Also, is there a solution for that when you combine asterisk with ser?
Regards,
Ronald
Thank you all! I will check on those.
Regards,
Ronald
JP Carballo wrote:
Ronald Ramos wrote:
Hi All,
Any solution on how I can implement prepaid billing on asterisk?
But not the calling card type, just a simple Custome rwill buy
credit, consume then buy again.
Also, is there a solution
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