if internet connection is down asterisk stops working correctly?
¿How could I solve that?
SIP locks if it tries to do DNS queries and doesn't get an answer.
Try using a local caching DNS server.
HTH
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Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39
,
--
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945
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, PeerStatus, Hangup.
More info:
http://www.voip-info.org/wiki-Asterisk+manager+API
http://www.voip-info.org/wiki/view/asterisk+manager+events
HTH,
--
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945
hints?
Thanks in advance,
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Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945
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hints?
Thanks in advance,
--
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945
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for
Asteris. I'm not affiliated with them in any way, but on my machines it works
perfectly.
HTH,
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Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945
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:
http://lists.digium.com/pipermail/asterisk-users/2008-May/211000.html
No, I'm using Asterisk 1.2.25-bristuffed. I'll be trying Russell's suggestion.
Thanks,
--
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945
, and in this situation if I make a call
from B to A, suddenly peers in server A are able to call peers in machine B.
Can anyone give me directions?
Thanks in advance,
--
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945
.
Best regards,
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Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945
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his data.
HTH,
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Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945
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in Realtime:
Set the priority as -1.
Set the app as the hint.
I have a small question: other than a phone (ie. SIP/something), what else can
I use as app? Can I handle the change via some custom code?
TIA,
--
Dott. Andrea Spadaccini
Multimedia Technologies Institute s.r.l
, in
order to display it in a web page.
How could I do it using the hint mechanism?
Thanks again,
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Multimedia Technologies Institute s.r.l.
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Hello everybody,
is it possible to execute some dialplan code when an extension changes its
state?
Say that SIP/204 becomes busy. I know that using the hint priority some phones
can show its state. Am I able to call custom code in the same way?
Thanks in advance,
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Dr. Andrea Spadaccini
[queue-caller]
exten = s,1,Set(TEST=a)
exten = s,n,Queue(various args)
The error was that I didn't make the variable inheritable, prepending it with
one or two underscores.
Now it works, thanks!
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Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it
it be implemented? I see that
uniqueid changes for each call in the scenario that I described, so I'm a bit
stuck.
I'm using asterisk 1.2.2x (I know that I should migrate.. This is the last
release of our product that uses 1.2).
Thanks in advance,
--
Dr. Andrea Spadaccini
Multimedia Technologies
, and I'm just back
home), but I think that the concept is right.
Is this the same kind of setup that you have? Does it work for you?
Can you post some simplified working code?
Thanks in advance! :)
--
Dott. Andrea Spadaccini
Multimedia Technologies Institute s.r.l
Asterisk easily if you want.
HTH,
--
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945
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Asterisk easily if you want.
that's exactly the kind of elaborate scheme I was hoping to avoid.
Sorry, I guess that the correct solution depends on your needs :)
I needed that scheme, I built it and I'm happy with that.
Have a nice day,
--
Dr. Andrea Spadaccini
Multimedia Technologies Institute
reading the source code, but I didn't find a solution.
Thanks in advance,
--
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945
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that was specified in the request.
HTH,
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Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945
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Hello everybody,
I know that I can define variables in the [global] context of extensions.conf.
How can I do the same thing in other conf files, like features.conf?
Thanks in advance,
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Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95
to
finish the execution.
Should I file a bug?
Thanks in advance,
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Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945
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as the
asterisk user (originate a call to the application System).
Maybe it could be good to limit in manager.conf which commands can be executed
by each user.
In this way, we could give more granular permissions, at the price of a
slightly more complex manager.conf syntax.
HND,
--
Dr. Andrea Spadaccini
.
Don't worry, it's a normal message. :) It's only annoying.
Bye,
--
Dott. Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
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Ciao Matthew,
I would be very surprised if chan_modem actually works... I don't think
I've *ever* seen it setup before.
Well.. So there's no hope to make that modem work with Asterisk, right?
Thanks,
--
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x
instructions?
Thanks in advance,
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Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945
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()
exten = s,203,Dial(${trunk_3}/${ARG1})
exten = s,204,Hangup()
Which asterisk version are you using?
IIRC, priority jumping (ie. going to n+101) was disabled by default in some
1.2.x version. You should rely on DIALSTATUS. See Dial() page in voip-info.org.
HTH,
--
Dr. Andrea Spadaccini
Multimedia
of NVFaxDetect, as in my 1.4 test machine there isn't,
right now, any PCI card.
Please test it and report the results.
HTH,
--
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945
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? If not, what is your approach to gain
setting in mISDN?
Thanks in advance,
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Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945
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Ciao Florian,
I'm trying to setup Asterisk on debian etch (with the debian packages)
with a Fritz!Card PCI ISDN card and chan_capi.
Why don't you use mISDN drivers?
Bye,
--
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945
understand in depth, I now
understand clearly from reading your book.
Any plans for a sequel ? I'll order 10 copies in advance :-)
AFAIK, the second edition will be out in August, covering Asterisk 1.4.
Bye,
--
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l
this message through a search engine: the
announce = XX option in queues.conf allows me to solve my problem.
I can set it dynamically through the fourth parameter of the Queue()
application.
--
Dott. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l
Hello *,
do queues allow me to set an announce like the A() option of the Dial() cmd?
The announce that I've found is a message that is heard by the caller. I'd like
to send a message to the member of the queue that picks up the call.
Thanks in advance,
--
Dott. Andrea Spadaccini
Multimedia
class.
Can someone explain me what's going on?
Thanks in advance,
--
Dott. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
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an
hangup and hangup conditions must be handled (i.e.
Vicidial/Astguiclient).
Just my two cents,
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Multimedia Technologies Institute s.r.l.
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Ciao Andrew,
On Tuesday 06 February 2007 7:02 am, Andrea Spadaccini wrote:
This would surely be more intuitive and would require less dialplan
programming when there are more than one point where one might get
an hangup and hangup conditions must be handled (i.e.
Vicidial/Astguiclient
manager.
HTH,
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Multimedia Technologies Institute s.r.l.
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this is a IAX problem.
Can anybody help me?
Asterisk version 1.2.13.
TIA,
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Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
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installation, and then everything ran smoothly.
HTH,
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Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
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to prevent
it from showing auth data in the logs.
If you aren't a C programmer, I can write for you a small patch that
will get the job done.
Bye,
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Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
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you want.
See http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL for more
details.
HTH,
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Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
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of deployments?
Thanks in advance,
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Multimedia Technologies Institute s.r.l.
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install those functions?
I'm using Asterisk 1.2.10.
Thanks in advance,
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Multimedia Technologies Institute s.r.l.
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the error, but might explain why you get a working
CDR.
Try to issue the db get and db put CLI commands, to see if AstDB is
working.
HTH,
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Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
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+MYSQL .
HTH,
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Multimedia Technologies Institute s.r.l.
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Hello everyone,
did anyone attempt to use Asterisk with UMTS third generation mobiles?
I'd like to route some calls to an UMTS phones, but I don't know if
this can be done/is likely to be done/will be available someday/is
impossible.
Thanks in advance,
--
Andrea Spadaccini
Multimedia
Well, how does Asterisk interact with those devices? Is there a
chan_gsm_pci?
Thanks,
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Multimedia Technologies Institute s.r.l.
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Ciao Noc,
checking whether the C compiler (gcc -O6 ) works... no
In my gcc version (3.4.4), there's no -O6 switch.
Try removing from your CFLAGS the -O6 switch, or replacing it with a
more conservative -O2.
HTH,
--
Andrea Spadaccini
Multimedia Technologies Institute s.r.l
the phone to the net WITHOUT doing anything
2. Write the configuration file
3. issue the encode.sh script as stated in the user guide
But the phone still displays no IP (it's a BT-102)...
Where's the mistake?
Thanks in advance,
--
Andrea Spadaccini
Multimedia Technologies Institute s.r.l
the phone to the net WITHOUT doing anything
2. Write the configuration file
3. issue the encode.sh script as stated in the user guide
But the phone still displays no IP (it's a BT-102)...
Where's the mistake?
Thanks in advance,
--
Andrea Spadaccini
Multimedia Technologies Institute s.r.l
in the From: and To:
headers, while asterisk doesn't.
Can you help me?
Thanks in advance,
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Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
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.. maybe it'll be faster to
configure them in the traditional way.
Thanks a lot for your help,
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Multimedia Technologies Institute s.r.l.
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Ciao Eric,
If you had a PRI (not just a T-1) AND your telco permits you to set
it.
Is there any hope to change the caller-id on a BRI line?
Thanks,
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Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
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information.
HTH,
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Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
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Ciao Benchev,
Also register= can be done only from a .conf file.
Well, I'm experimenting right now with this, and I can tell you that
register = works even with static realtime.
HTH,
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Andrea Spadaccini
Multimedia Technologies Institute s.r.l
:
[00:0B:82:09:5D:81][000][FFFB][01000817] Send SIP message: 1 To
192.168.1.2:5060
repeated N times (with N 10).
The phone is configured correctly, with the username and password
specified in sip.conf.
Did someone encounter this issue?
Thanks in advance,
--
Andrea Spadaccini
Multimedia
, but I still see the
default number on the called phone display (a mobile).
Please help me..
Thanks in advance,
--
Andrea Spadaccini
Multimedia Technologies Institute s.r.l. - www.mediatechnologies.it
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in from instead of
callerid (overrides the callerid) when placing calls _to_ peer
(another SIP proxy). Valid only for type=peer
/quote
And again:
quote from=Asterisk sip fromuser
This is used when calling TO this peer FROM asterisk.
/quote
Thanks for your answer,
--
Andrea Spadaccini
Ciao Giorgio,
when I read MSN I thought of Microsoft messenger and I thought I used
SIP protocolI never used it, sorry for my stupid advice ::)
Don't worry, the subject is not punctual. :)
--
Andrea Spadaccini
Multimedia Technologies Institute s.r.l
phone, text is the optparam
/snip
So maybe the d option will help me.
I'll test it next week..
Thanks again,
--
Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
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.
HTH,
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Multimedia Technologies Institute s.r.l.
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as NT. The GSM gateway is connected to
the NT port of the PBX running asterisk. Is it a correct configuration?
Right now I expect an incoming call to the GSM gateway to be routed by
Asterisk, but the gateway doesn't even synchronize with the PBX.
What could be my mistake?
TIA,
--
Andrea Spadaccini
follow my dialplan rules.
Thanks to everyone,
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Multimedia Technologies Institute s.r.l.
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Hello everyone,
I'm trying to set up an Asterisk machine with a quad-port BRI
Junghanns card, and I want to use the mISDN drivers.
I'm having some trouble configuring it: do I need to use CAPI drivers?
I haven't found good links, could you please provide some info?
Thanks in advance,
--
Andrea
understand the connection between mISDN and chan_capi: are they
similar? Do I need both of them? I'm a little bit confused!
Thanks,
--
Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
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a 'sip-in ;incoming sip calls' context not
found in extensions.conf.
IMHO the comments should be stripped off by asterisk itself!!
It should be easy to modify the script, but the problem would remain.
Should it be filed as an Asterisk bug?
--
Andrea Spadaccini
Multimedia Technologies Institute
your script.
regards,
/Olle
I sent this info to the script's author.
Thanks for your help!
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Multimedia Technologies Institute s.r.l.
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I
right?
Where can I obtain more info about these metrics?
Thanks in advance,
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Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
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= mysql,asterisk
Thanks, this was the main point of my question!!!
Here you will find all information you will need, to setup the
static and real realtime:
http://www.voip-info.org/wiki-Asterisk+RealTime
Hope this helped
Yes, it helped a lot!
Thanks again,
--
Andrea Spadaccini
Multimedia
and use Real Realtime
sippeers and sipusers?
Moreover, is there an easy way to switch between Static realtime and
Real realtime mode for the realtime families?
Thanks in advance,
--
Andrea Spadaccini
Multimedia Technologies Institute s.r.l
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