We have a problem with Asterisk not locking voicemail.conf for update.
It appears to not protect the file even against itself. It certainly
doesn't use flock() to protect it against others.
This is a problem for several reasons. First, of course, people can be
hand-editing the file to add or
On 2005-11-17, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
4. What is a good company to contract with for emergency support?
Digium?
Find a local consultant. There are quite a few around...
We're in a rural area of Kansas. Probably not that many around,
actually. What we're talking about
On 2005-11-17, C F [EMAIL PROTECTED] wrote:
2. What do we need to do for our data network to make VOIP reliable?
QoS, basic traffic prioritization on the switch, vlan, ???
If you are not running a bandwidth hungy network, then you might be
able to work with just one vlan, if you don't
I work for a company that is nearing the end-of-life on its existing
Nortel Meridian switch and is considering Asterisk. We have
approximately 200 existing extensions, and probably 150 out of those 200
are using basic analog phones and would stay that way. The rest would
have VOIP phones at the
On 2005-06-28, hank [EMAIL PROTECTED] wrote:
how do they know if your calling your tax dude or something? what do they do
monitor the calls or something?
They probably don't, but they could take me to court if they ever find
that. It would also be trivial for them to google for their name,
On 2005-06-28, Rich Adamson [EMAIL PROTECTED] wrote:
It's probably a $2 decision. Just pick one or two and try them.
There are a fair number of people on this list (including myself) that
stay current with multiple itsp's. Every itsp is going to have a problem
now and then, so keeping a
On 2005-06-28, r00t [EMAIL PROTECTED] wrote:
I'll second voipjet for outbound only. While many reported problems to
VoipJet bothers me for two reasons. First, their terms of service are
absolutely insane. Users are specifically forbidden to place calls
regarding medical or financial matters
On 2005-06-28, Wiley Siler [EMAIL PROTECTED] wrote:
So far my experience with TOS has been that most of them are pretty odd.
Not THAT odd :-)
No one wants the liability of a stock trade gone foul or a call to the
doctor that gets disconnected. Essentially, those things in the TOS are
just a
On 2005-06-26, Adam Megacz [EMAIL PROTECTED] wrote:
Rich Adamson [EMAIL PROTECTED] writes:
I've had pretty good luck with www.teliax.com
I like them too, except for support. I have THREE tickets open with
them that are ten days old and haven't received even a cursory we're
looking into it
On 2005-06-27, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
http://www.nufone.net. I've been using them for the past 18 months with zero
technical hassle. Jerjer and Shido6 hang out on IRC. Nufone is not a hand
holding VOIP provider. You are expected to have some clue. This has turned
On 2005-06-27, Michael Di Martino [EMAIL PROTECTED] wrote:
If this list spent at least half the time on helping other asterisk
admins as it does on
trivial things like LiveVoips bankruptcy it just might be a great list.
As it stands now this list is kind of useless. Most request for
Hi,
I have a Sipura SPA-3000 for access to my standard analog PSTN line. I
have the SPA-3000 answering and then directing all calls into Asterisk.
This setup is working fine for everything except voicemail. Most, about
2/3 or so, of messages left come across very quiet when the voicemail is
Hi,
I am dialing out from my Asterisk system to the PSTN via Livevoip. I'm
having trouble with jitter, though on my end things are fine. The
people I'm speaking to are the ones to complain.
I'm on a fairly standard DSL line, everything looks OK on this end, and
I'm not really sure what could
On 2005-04-08, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Can you tell me, which SIP providers that *work with asterisk* are
providing unlimited inbound/outbound calling for $20? Iconnecthere is 30,
broadvoice is 30, etc..., and will allow you to specify your outbound cid?
Actually, I'm
On 2005-04-08, NVC List Manager [EMAIL PROTECTED] wrote:
On Friday 08 April 2005 14:20, Kerry Garrison wrote:
The reason they charge more is for you to have more volume. Yes, yes, I know
it says unlimited. But there is no such thing. It's a marketing ploy to get
users. They estimate how much
On 2005-04-06, Antoine Delaporte [EMAIL PROTECTED] wrote:
Also, any dangers/performance issues from an isp point of view for
running ztdummy?
Few month ago I've run my Asterisk, under an UML.
I was very disapointed, my Sipura3k don't stay registred much more than
12 hours
I'm not really
Hi,
I recently encountered an odd situation: the network cable to my
SPA-841 got unplugged while it was in the midst of a call. I got it
re-plugged in about 30 seconds, and the phone rebooted. The phone
showed no evidence of the previous call in progress and worked like
normal.
Asterisk, on
On 2005-04-05, Rusty Shackleford [EMAIL PROTECTED] wrote:
What could I do so that Asterisk would automatically
terminate a call in these situations?
Check out:
http://www.voip-info.org/wiki-Asterisk+sip+rtptimeout
Perfect. Thanks!
-- John
___
On 2005-03-31, Chuck Bunn [EMAIL PROTECTED] wrote:
I am new to Asterisk and the first thing I have noticed about Asterisk
and Pingtels open PBX's is that they are using this dinosaur method of
running forums. It is a real pain getting every message in the forum and
It is a real pain to have
On 2005-03-28, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I was just wondering how others are addressing this. You can't all be
making receptionists memorize codes, are you? :)
At my workplace (which does not use asterisk), most phones are the
standard beige desk phones from the 1970s --
On 2005-03-29, Andreas Sikkema [EMAIL PROTECTED] wrote:
In a VoIP only environment, Asterisk has to use ztdummy
to have any chance of playing back understandable audio
files (without drops, hickups etc).
That brings up a question... on my Alpha, ztdummy is of (apparently)
little use since I
On 2005-03-26, Niksa Baldun [EMAIL PROTECTED] wrote:
It is probably a SPA-3000 problem. I have tried a similar setup (not for
911, I need to hangup a phone) and it works with ISDN phones and
Swisswoice SIP phone, but not with BudgetOne, for example. The phone
just won't drop the line for some
Hi,
I have a SPA-3000 and would like to use the 911 recipe from
http://www.voip-info.org/wiki-Asterisk+tips+911. So I took the simple
recipe and modified it slightly:
exten = 911,1,ChanIsAvail(SIP/potsoutbound)
exten = 911,2,Dial(SIP/potsoutbound/911)
exten = 911,3,Hangup()
exten =
On 2005-03-26, Tim Connolly [EMAIL PROTECTED] wrote:
Might be a good idea if we didn't post such scripts.. How many
people are going to accidently call 911 while trying to test this? Use a
test number, like 611 or 411 or how about your boss's cell phone ?
Sound advice. I, of course, was
Hi,
I have a SPA-3000. My telco sends no disconnect indication
whatsoever, so I am having trouble making the PSTN line detect
disconnect.
What I want it to do is detect the very loud off-hook tone that is
generated about 20 seconds after disconnect. It's not perfect, but
it's a lot better than
On 2005-03-20, goldhorse [EMAIL PROTECTED] wrote:
I use a Handytone 486. While you have to configure dtmfmode=rfc2833 in
asterisk,
it will not work if you do not set the dtmf mode to SIP info in the
ATA itself.
So you might try different combinations fo dtmf modes in both asterisk
and
On 2005-03-20, John Goerzen [EMAIL PROTECTED] wrote:
Thanks for the tip; unfortunately, it doesn't help.
I should add that I am running Asterisk 1.0.6 on Debian sid. Any config
files, debug logs, etc. that could be helpful, I'm more than happy to
provide. Just tell me what you need.
-- John
Hi,
I have a SIP phone connecting to my asterisk server, using
dtmfmode=rfc2833. When calling from the SIP phone to internal asterisk
services, such as voicemail, it works fine.
But when I call out to the PSTN, from the SIP phone, via my X100P, the
call will be connected fine. After that,
Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these
extensions.conf lines:
exten = s,1,SetVar(SET_EMERG_FLAG=0)
exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten = s,n,SetGlobalVar(EMERGENCY=1)
exten = s,n,SetVar(SET_EMERG_FLAG=1)
exten =
Hello,
I know of someone that is thinking of spending $20,000 on a new
voicemail system because their vendor is end-of-lifing the system they
have now. I mentioned that maybe Asterisk could do what they need, at a
much lower cost. Reliability is, of course, critical -- which brings up
the topic
I've heard good things about VoipJet here, so I was going to set up an
account. Then I noticed their Terms of Service here:
https://www.voipjet.com/tos.php
Several things there are very concerning to me, and I'm interested in
what other people here think of them.
* The ToS specifically forbids
I've heard good things about VoipJet here, so I was going to set up an
account. Then I noticed their Terms of Service here:
https://www.voipjet.com/tos.php
Several things there are very concerning to me, and I'm interested in
what other people here think of them.
* The ToS specifically forbids
OK, I missed the message that started this thread, but:
On 2005-03-11, Wiley Siler [EMAIL PROTECTED] wrote:
Regardless, you seriously need to come to terms with the fact that it
WILL continue to happen.
New users will always equal people without a clue.
No, that is not so. Some new users
On 2005-03-11, Jean-Michel Hiver [EMAIL PROTECTED] wrote:
I think they are putting this in to avoid being sued because that duper
important call your were placing was cut for reason X Y Z. However it's
Right, I understand. But if that's what they mean, why don't they write
something like
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