on here about voiceconduits. The
general gist was they are not yet open for public business.
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received (both
directly and through a transfer) and look at them. How can we do this ?
How are you performing the transfer?
Have you tried the following?
show application ResetCDR
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you're seeing.
At any rate, without log, error, or console messages there's not alot we
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ts_threshold=0&comments_offset=0&comments_sort_mode=commentDate_desc&comments_maxComments=10&comments_parentId=931#threadId1168
It might be channel specific, I do not know.
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sound like you are an objective third
party, next time try to remember to set your email address properly.
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these sites:
http://www.voip-info.org/
http://www.asteriskdocs.org/
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om beginning to end.
Not found it yet? Read it again. All of it.
Still looking? OK, I'll give you a hint, all that stuff below my signature.
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allerid. Check the wiki for examples, I don't know the syntax. (search
for ex-girlfriend blocking)
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asterisk just to be in sync with the libraries. (then rebuild)
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To
any US did that arrives into that
context.
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Oswaldo Arratia wrote:
Has anyone figured out how to make a Sipura to dial an extension
automatically as soon as you pick the the handset?
Go to google and type: sipura hotline
Read the first three links.
Test.
Send us a note telling what worked for you.
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http
Rejected connect attempt from
> 66.234.228.170, request
> '[EMAIL PROTECTED]' does not exist
So, where is this voicepulse_connect_context context?
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; misdials an extension
include => vp_context
This way, you define both contexts, and include the extensions that were
defined in [vp_context] into [main_vp_context].
I don't know if this will resolve your other problem, but I believe this
is the dialplan you we
other readers: I've not had an error like this, so I'm not sure
exactly what the protocol is. Should the original poster post next to
asterisk-dev, or to mantis?
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he phone number that is calling you)?
You will need hardware that is compatible with your areas telephone
network. (Stating that as it is likely different from the US network.)
If digital voice circuits(in any form) are available in your area,
you'll likely be happier using them than POTS l
t, can you explain this
registrar db concept you're talking about?
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card company a call and have them lean on sixtel.
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Since international numbers vary in length, you'll want to build a
pattern that will match a rule(that you'll specify) for dialing
international numbers of any length.
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red to see if it happens again.
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h
aptel devices can be set in immediate mode in your
zaptel.conf file.
For IP hardphones, look through their dialplans for a hotline mode. For
example, I am fairly certain that my SPA2000 could be configured this way.
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to see what exactly
happened, but stopping and restarting asterisk resolved it immediately.
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using
them for anything real important.
There doesn't seem to be a business in the market that one could rely on.
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ber search
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Andrew Thompson wrote:
Greetings,
Sorry, wrong list :-D
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To
pport
* Nufone actually took their contact number off their website for a
while, I guess hoping I wouldn't be able to contact them. They never
once responded to any of the telephone calls I made to them. I just
signed in and see that they didn't ever refund my payment and close my
acc
y on the phone would take away my objections to the
entire process of '#'-transfers :)
I just have one problem with that. I have _never_ (that I know of) seen
a phone with A-D on it!
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OK - I should know this... How does someone call in and pick up there
messages remotely?
Jon
Hint: Post a new thread properly, and those people who had stopped
reading this thread might see it.
Hint2: GIYF, asterisk check voicemail
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http
with the ReplyTo munging that is done on
lists.digium.com lists?
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convention for referring to the two distinct
modes of a device?
Can two entries with the same name exist at the same time, if one is a
friend and the other is a user? Mainly, I'm asking, are the results here
undefined, or is this a recognized use?
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would have expected
it to have a subject line with at least password or reminder.
2) - Speaking of passwords, you might want to change yours now that
we all know what it is!
Hmm... wasn't paying close enough attention.
Done.
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Stefan Gofferje wrote:
That's brilliant! And so easy...
Works exactly as supposed. You should put it onto the wiki under section
"tips & tricks".
Regards,
Stefan
PS: Chris, your boss might like this also!
Wow, is it too late to put in for royalties? ;)
I'm glad it w
Stefan Gofferje wrote:
Andrew Thompson schrieb:
Stefan Gofferje wrote:
[private_huntgroup_day]
exten => s,1,Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED],15,rt)
exten => s,2,Wait(1)
exten =>
s,3,Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED]&Local/[EMAIL PROT
Twice in the last week or so, I've received a message similar to the
attached.
A portion of the attachment that's attached is not in English. Is this
my mail server failing, or someones who's on the list?
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quick and painless, or an insurmountable task.
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falling down to your 810920 voicemail?
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+servers
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iming device(zaprtc/ztdummy/zaptel).
You have a SIP problem. Please post us your [general] section and the
section for one of your sip phones.
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l-free numbers now, and I don't know if it will stop...
Is there any hope for me?
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other than ringing
your house for a few rings before ringing your house AND the cell. Even
then, the cell provider's 'out of the service area' message would answer
the call.
That's where you move on to building/adding the logic for: "Press 1 to
accept this call"
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n it until then. This would
be a way for you to buy some time between tries. If I'm just making this
up, use cron to straighten it out.
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().
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Adam Robins wrote:
Works beautifully! Thanks.
Could you post here, or to the wiki, or just back to myself the
configuration you're using to implement this?
Thanks.
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does your machine fall over
and vomit all over itself?
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aximum charge for
dialing 700 numbers or something like that. But it still ought to be a
local call(very, very cheap or free) next door to my neighbors.
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A
and that it was
really cheep, she just kind of shrugged, said she didn't think people we
knew would call it.
PS: If you can offer me a DID off of Level3's 910-817 block for a
reasonable price, contact me offlist.
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ht
remember looking over some of their phone bills, but
I can't recall if I saw usage charges on the RCFs.
Hope this helps.
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e relavent telephone numbers, or use a
database lookup tool. The last command ran here would be:
Goto(, 1)
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way to hook into
Dial that early.
I looked at show application dial when I read the question earlier today.
There is a hook for playing a message to the recipient of the Dial
before patching them together, but I didn't see anything for the other
way around.
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me up with: http://www.rcfp.org/taping/
* I take no responsibility for their content.
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John Middleton wrote:
Anyone got any experiences of these with *, and also costings?
Someone mentioned them on the list several months ago, but I don't think
anyone mentioned actually using it.
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oading the voicemail
configuration from a database, and using an AGI that can read that
database to authenticate and process your call forwarding.
An upside to this might be the ability to allow users to change their
own password(which I'm not sure they can do with voicemail.conf).
--
)
Customize accordingly...
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came back that I would
have gotten an email about it. I've sent the request back, so we'll see
what happens.
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n
to pick a NXX.
(I don't want to click the 910 area code to find out they giving me a
Wilmington or Raleigh local DID that's just useless to me.)
Thanks!
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ess digits work ok.
It sounds like you don't have a pattern to match dialing more than 7 digits.
Post your mayores context from extensions.conf.
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Asteri
with
another party?
To my understanding in both cases the phone1 is busy so why return
different priority code???
Take a look at DIALSTATUS at:
http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
Also,
Do you get a call-waiting beep when you're on the phone with the
original par
symptom of the interference you spoke of?
What filters have you used?
Thanks.
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know what kind of device I need.
Thanks in advance
look up: cellsocket
There are other similar devices, but the names now slip my mind. What
type of network is your cell phone on? (cdma, gsm, tdma, etc)
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penned to be wondering about the Stargate...
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guration, or your problem. Error messages are
helpful, too!
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ersion, ability do identify, find, and download appropriate versions of
software, etc.
This would be a good place to start:
http://www.google.com/search?q=site%3Avoip-info.org+h323+oh323
BTW, how can I get PWLib!
You could type PWLib into your favorite search engine...
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like each ring is like a seperate call as far as the
callerid history goes. Anyone have any ideas?
Michael Lyszczek
What does that have to do with voiceconduit?
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device
you would need to plug a standard phone line from your local LEC into
asterisk.
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To
Greetings,
Does someone have the link to reset your password on bugs.digium.com?
I can't seem to find one.
Thanks.
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ired. :-)
He said he was posting a large file, it may have been larger than what
the list allows.
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he name of one.
Find a site, upload it there, post your message with info and point us
at the link.
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, but any help
would be much appreciated thanks everyone!
-Chris
MeetMe?
http://www.voip-info.org/ <-- see the wiki
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ecure? The entire datastream, the
authentication(username/passwords), and/or the voice traffic itself?
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ed to
make the rest of the features work. Is there an application I can use to
listen on a line for flashes, digits, callerid, and did-type info?
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n't get the time to rewrite it, I am sure I'll need to
add six more fields within the next few months.
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Kevin Walsh wrote:
Bart de Wild [EMAIL PROTECTED] wrote:
unsubscribe
No.
LOL...
You are trapped on this list forever! mwaa ha ha...
I know, lame and useless discussion of a useless post, but I just
couldn't stand it.
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http://www.voip-info.org/wiki-Asterisk+config+sip.conf (currently hosed)
http://64.233.179.104/search?q=cache:IIOmLeG89KwJ:www.voip-info.org/wiki-Asterisk%2Bconfig%2Bsip.conf+site:voip-info.org+sip.conf
(google cache)
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[EMAIL PROTECTED] wrote:
I am getting a 6 second delay whenever i dial 9 to call someone using
PSTN, What could be causing this??
Pattern matching, perhaps?
What's your dialplan look like for the station you're calling from?
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, on a web-based forum, the Admin would lock the thread...
Please, can we call this thread closed?
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message again
And/Or if they typed a wrong extension ti will read the menu again
please let me know
Thaks
Altus
Go to the wiki: http://voip-info.org/
The answer is one or both of: t, i
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d test for if the phone is locked and if so, fail it. Same
thing for calls going toward the phone if that's required.
You would also need an extension and possibly a password to "unlock" the
phone, by setting the variable to another value.
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omes up with an easy
solution for the original poster, I'll take a look at it.
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o require coding and testing.
If a SIP adapter could be dropped in and as a side effect of the
configuration it broke sending DTMF out, only a few changes to the
dialplan would be required to get things back in order.
Anyway, it was just an idea, and he did say he was looking for ideas.
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n any case, chan_sip would be much more likely to be hackable to make
DTMF quit working.
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working. If any of them failed, the
lady that maintained the phone system would be the next person called.
That was never a happy phone call...
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you will post it on the wiki(if you're
not doing it now), or at least mail it to any other interested parties,
right?
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er channel. It plays the time from
a nuclear clock.
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lso have not noticed
any swaying.
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or whatever) is sufficient, or if you
must actually restart all of asterisk. (This could help narrow down the
issue.)
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Brian West wrote:
>>
>> Does anyone know if just installing the new asterisk will work fine
or do
>> I have to remove the 'old' install first?
>>
>> greets
>
> no
I generally tar and gzip the three folders to a backup folder before I
do up
d be interested in working
on the script, but have no use for it personally at this time. If
someone else is interested, I'd be inclined to devote some compute
cycles to it.
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[EMAIL PROTECTED] wrote:
Hello everyone!
Version 1.0.2 is now available for Asterisk, Zaptel, and libpri.
Would it be possible to get official changelogs for these releases?
(If they're out there, please point me toward them.)
Thanks.
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these keywords:
site:Linux.digium.com ipsec
site:Linux.digium.com vpn
Sugar to taste... (ie, add any other keywords that you think are helpful)
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at you're looking for.
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ginning of some kind of "dial-plan modules"
collection, where you post your macro or dial plan logic and show sample
usage.
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Andrew Thompson
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My dialout line is a copy of my working voicepulse out section:
[NuFoneOut]
exten => _1NXXNXX,1,SetCallerID(9104108307)
exten => _1NXXNXX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten => _1NXXNXX,3,Congestion
exten => _1NXXNXX,4,Hangup
> My context in iax.conf is NANPA or s
The main speaker (that I believe is Mark), is looking at a list of post
1.0 wish features. Is this list online somewhere?
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ice for me. It may not be for anyone else.
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xed" or makes a majority of it's call
from a single cell site.
Take the phone out and use it somewhere else a couple of times a month.
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0.0.tar.gz
http://xninja.net/asterisk/libpri-1.0.0.tar.gz
Note: I have no idea if this will remain available, of if my host will
turn it off! :)
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rpose?).
Hmm... Now there's some psychadelic/chanting on-hold music... :)
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beta.
If there is an IAX in, I could download diax again.
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an
CallerID and DID/DNIS/called number delivery. Even if both of those were
not feasable, or not currently supported by their phone systems, it
would be ok. We just would like to take advantage of the two existing
SHDSL pipes.
Currently there is 100-500/month spend in ld/tollfree between the two
off
.
I would prefer that there be no other intermediary involved in call setup.
I have a SPA-2000 at home, but I couldn't find anything in the setup for
doing this. If anyone believes that the SPA can do this, please provide
me with configuration info.
Thanks.
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