Hello ppl,
Any idea how do I write in include lines(for contexts or include files)
in the database, in the ARA static config?
thanks in advance.
Ben.
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Rushowr wrote:
Benjamin Jacob wrote:
Hello ppl,
Wanted to know your experiences, if you've worked with Asterisk Realtime
Architecture.
Which one do you prefer, static or realtime?
I personaly think, the static architecture is a better solution, cuz, in
the realtime config, to check the dial
Tony, the VPM450 is far better than the TE411P's VPM400. One main thing is that is has full 128ms tails on all spans whereas the VPM400 shared 128ms as you used more spans, and second is the Octasic chip makes the sound real crisp and clear.
For the moment, if you need FAX tone detection, you will
Hi to all,
I read
http://www.voip-info.org/wiki/view/MS+LCS+2005+%252F+SER+%252F+Asterisk+Integration
Is it possible to use ser as a presence server instead
of LCS 2005 ?
Harry
___
D
Hi all,Currently i've made some tests with VoIP buster and everything is running ok for outbound calls.Now i've created new VoIPbuster account, and my goal is to allow VoIPbuster partners to dial into my dialplan IVRs for free.
But I always get my VoIPbuster account (currently registred with my ast
On 09/11/06 18:36 Paco Brufal said the following:
Hello,
Here in Spain there is a VoIP provider (Telefonica) that only works
if when you make an outgoing call, the SIP headers are like this:
INVITE sip:@telefonica.net SIP/2.0
But Asterisk is sending this:
INVITE sip:@sbc.ngn
hello
If I want to use asterisk to hookup to a SIP account
I just use the "register" line in sip.conf with the
extension number at the end...
But how about if I want to use a SIP trunk from a
provider which gives me 10 DID numbers with the same account?
thanx in advance
rick
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On 11 Sep 2006, at 10:29, Roy Gardner wrote:
Hi,
Our setup is:
Asterisk 1.0.7 running on Debian 2.4.27-2-386
TE110P card
ISDN 30 (UK E1 PRI)
When making outgoing calls to the PSTN using call files I get the
following problems:
1. No hangup detection - have to wait for time-out
2. No pick
Hi All
I use Grandsteam GXP2000 phones.
Is there anyway within the dialplan/indications etc to have a custom
ringtone based on who is calling the phone.
i.e if i have a call from an internal user i get one ringtone if its an
external call i get a different ringtone??
Many Thanks in Advance
Can Asterisk bind on multiple ports?
I wish I could in my sip.conf make Asterisk bind different ports per
different context, so that calls coming in udp port 5060 would fall in
one context and calls coming in port 5061 fall in other different
context. Is that possible? How can I edit my sip.con
Think of BerkeleyDB as a barebones embedded RDBMS. It carries the much the same functionality (such as ACID) as most Relational DBs but without overhead (such as SQL translation) - for many years it formed (possibly still forms - not sure of present) the core of MySQL.Steve Totaro <[EMAIL PROTECTE
Hello,
Here in Spain there is a VoIP provider (Telefonica) that only works
if when you make an outgoing call, the SIP headers are like this:
INVITE sip:@telefonica.net SIP/2.0
But Asterisk is sending this:
INVITE sip:@sbc.ngn.rima-tde.net SIP/2.0
because "sbc.ngn.rima-t
Hi,
How do the 302 redirects work in asterisk, and what is the
"promiscredir" directive doing? I am not getting the documentation on this.
I have following happening on my asterisk box:
-- Executing Dial("mISDN/1-1", "SIP/[EMAIL PROTECTED]||Tt") in
ne
Benjamin Jacob wrote:
> Hello ppl,
> Wanted to know your experiences, if you've worked with Asterisk Realtime
> Architecture.
>
> Which one do you prefer, static or realtime?
> I personaly think, the static architecture is a better solution, cuz, in
> the realtime config, to check the dialplan(n h
FRANCISCO PEREZ-LANDAETA wrote:
Hi, this message is for Steve.
Sorry for replying to the digest. It wasn't my intention.
I would appreciate if you can guide as to how make the tenor asm200
work with asterisk. I am using asterisk at home. I guess my problem is
configuring the tenor so that it is
use:exten => s,2,Read(fwdnum|audiofile-to-play|10)then you'll have the number entered in variable ${fwdnum}.Type "show application read" in asterisk console for further details...
Hope this helps...2006/9/9, James Williams <[EMAIL PROTECTED]>:
I'm currently trying to write a section into my dialpla
I guess I will give it a try. The numbers are pretty much static in the
way they are routed.
I just did not know if Asterisk would choke on a conf file with a couple
thousand lines.
Thanks,
Steve
picciuX wrote:
don't know for conf size limitation (but i guess it won't be a problem
with a w
I believe the TE412P is intended to supersede the TE411P. What, if any,
are the advantages of the TE412P over the TE411P? Apart from the
disadvantage that the hardware DTMF detection doesn't work on the TE412P
(will that be fixed in the future?)
My UK supplier can supply both cards, but the TE412P
Hi,For a small call center, we would like to change default behaviour.Current setup and behaviour are :- a bristuffed 1.2.10 Asterisk server with 4 BRI ports- 5 SIP hardphones (Snom 320) with BLF for line or extension monitoring
- incoming calls ring all phones and light BLF on- when a call is comi
I always got this warning after I'm using IAX2 channels .
Sep 11 21:44:09 WARNING[30229]: chan_iax2.c:6536 socket_read: Received trunked
frame before first full voice frame
What's it mean?___
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as
Hello
I have a problem with callerid in the manager interface.
I think that Asterisk has a strange way to handle callerid, until I
found out to set the o-switch in the DIAL statement, it did not work
the way I wanted, it still doesnt, but now it works ok in one
direction.
My extension is 311, i
Sharon Lim wrote:
Hi all,
I have tried to install freepbx and a2billing application. Now see
both application is not integrated special on cdr part.
Any idea how to integrated it?Confuse!
--
Regards,
Sharon Lim
*Good memories are to be folded neatly and tucked away into the back
pocket *
You missed my point completely. His "original" post was a reply was
hijacking a very long thread (the digest thread) that he did not trim.
Just trying to teach some netiquette so he will get more help. I have
found that top vs bottom posting is not a major issue to most, but few
make a big
don't know for conf size limitation (but i guess it won't be a problem with a well-sized machine).About asking on the fly vs writing on change: if your "routing information" varies very often, on the fly should make more sense. Otherwise, it's not useful to retrieve continually same data: better to
Hi,Our setup is:Asterisk 1.0.7 running on Debian 2.4.27-2-386TE110P cardISDN 30 (UK E1 PRI)When making outgoing calls to the PSTN using call files I get the following problems:1. No hangup detection - have to wait for time-out2. No pickup detection - the dial-plan starts as soon as the line ringsAl
Hi,If you set Asterisk to ring several extensions for an incoming call, it appears that the call will be added in every phone's missed calls logs though the call was picked by one extension.In the long run, this prevent users from using missed calls features as these logs would filled with many cal
Hello ppl,
Wanted to know your experiences, if you've worked with Asterisk Realtime
Architecture.
Which one do you prefer, static or realtime?
I personaly think, the static architecture is a better solution, cuz, in
the realtime config, to check the dialplan(n hence the sql database) for
each
Rich Adamson wrote:
That's strange; how many people just responded with "that worked"?
None that I've seen! If they did, then they started with a subversion
directory, or they were responding to a different situation.
I suggest you start with a clean tarball and try it yourself. Or look in
the
I have two contexts how could I isolate context A from context B ,in
other words I want to ban context A from calling context B
In sip.conf, define phones/extensions something like this:
[1000]
type=friend
context=cust-a
[1001]
type=friend
context=cust-a
[2000]
type=friend
context=cust-b
[
You could disable dialing altogether unless they
press hash - that way they would learn about the hash key feature pretty quickly
:-)
Unfortunately I dont see an easy solution since a
dialplan covering all possibilities may be too complicated.
Cheers
Gerry
- Original Message
Hi all, I have tried to install freepbx and a2billing application. Now see both application is not integrated special on cdr part. Any idea how to integrated it?Confuse!-- Regards,
Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket *
Dear
I have two contexts how could I isolate context A from
context B ,in other words I want to ban context A from calling context B
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplor
http://www.voip-info.org/wiki/ here got alots of example but you need to find it. You can start with http://www.trixbox.org/ that install everything. Good luck!
On 9/11/06, Panagiotis Zikos <[EMAIL PROTECTED]> wrote:
Hi all, I am new in the asterisk company. I need to set up a small voip syste
Hi all, I am new in the asterisk company. I need to set up a small voip system for about 60 phones ( a small enterprise organization). The system must support voip calls (calls inside the enterprise) but must be able to send calls over isdn (24 channels). Thus the asterisk server must oper
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