is your Microsoft Windows server.
Brian J. Murrell
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To UNSUBSCRIBE or update options visit:
http
I am just misunderstanding your reference.
b.
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and
Answer would cause Asterisk to pick the POTS line up right away and dial
brian and joe's phones with it.
Am I missing something?
b.
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and
Asterisk does nothing. But if nobody picks up the POTS line (that
asterisk is on too) then it picks up.
I essentially want Asterisk to be an answering machine on the line.
b.
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Brian J. Murrell
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though, it just don't work the way they describe it.
Thanx,
b.
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in a. :-)
The above code made asterisk *never* pickup, so it must be possible,
I hope so. I'm going to dig into source to see. :-/
b.
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intro and recording the
conversation.
[from-pots]
exten = s,1,Dial(SIP/brianSIP/joe,30)
exten = s,2,Voicemail(u2001)
exten = s,3,Hangup
I will try this exactly and see if it works any better.
b.
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Brian J. Murrell
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that shares the POTS
line with Asterisk. Will that second pick up of the POTS line look like
a hangup on the POTS line to Asterisk while it is Dial()ing?
b.
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on the line pick up, it sends a hangup to Asterisk. Very
cool.
Thanx for all the input.
b.
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seem to work, despite my skepticism. It's wonderful.
b.
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dude. Not everyone has the $$ to outfit the whole
house with IP and IP phones right away.
b.
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Brian J. Murrell
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the Dial run it's timeout course.
Any way I can do that? Is there a way to create some kind of dummy
Technology/resource that will never answer but will always keep Dial()
happy that it might?
Thanx,
b.
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Brian J. Murrell
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the
Wildcard X100P?
b.
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Asterisk-Users mailing list
I have a macro which I call when I want to ring the house phones. So
for example, when a Zap line rings it enters the dialplan in the
inbound-pots context which is as follows:
[inbound-pots]
exten = s,1,Set(CDR(accountcode)=...)
exten = s,n,Macro(check-incoming)
exten =
On Mon, 2008-04-07 at 08:48 -0400, Steve Totaro wrote:
Just ignore it. Exit on non-zero just means Hangup.
Strange. Hangup sounds like a perfectly valid and successful operation.
I wonder why make that a non-zero exit.
Maybe set
verbose 0 will make you feel better.
~sigh~ I want to see
When I make a toll-free call using tf.voipmich.com DTMF doesn't work.
According to this post:
http://www.trixbox.org/forums/trixbox-forums/help/enum-strangeness it's
because voipmich needs dtmfmode set to info.
How do I specify this for a single SIP peer (tf.voipmich.com) given that
I normally
On Wed, 2008-04-09 at 16:18 -0500, Tilghman Lesher wrote:
No, that's correct. The problem is that you aren't using the peer definition
when you dial (as you said, you've never needed it before).
Use
Dial(SIP/[EMAIL PROTECTED])
NOT
Dial(SIP/[EMAIL PROTECTED])
Ugh. That's a problem.
On Wed, 2008-04-09 at 16:18 -0500, Tilghman Lesher wrote:
No, that's correct. The problem is that you aren't using the peer definition
when you dial (as you said, you've never needed it before).
Use
Dial(SIP/[EMAIL PROTECTED])
NOT
Dial(SIP/[EMAIL PROTECTED])
OK. Trying exactly as you
On Wed, 2008-04-09 at 17:21 -0500, Brent Davidson wrote:
Have you tried the using the SIPDtmfMode function in your dial plan?
Not sure how I would introduce that with my enum macro, but as a test I
did try it for this particular peer:
-- Executing [EMAIL PROTECTED]:1]
On Thu, 2008-04-10 at 11:13 -0500, Brent Davidson wrote:
You might also try canreinvite=no for both your phone and the sip
peer.
Yeah, there is definitely no re-inviting going on. Both Asterisk and
the local handset are in a local network behind NAT with reference to
the SIP server that
On Thu, 2008-04-10 at 13:36 -0500, Brent Davidson wrote:
One more tidbit I just ran across in the upgrade.txt file, since you
mention NAT: In 1.4, you need to set canreinvite=nonat to disable
re-invites when NAT=yes. This is propably what you want. The settings
are now: yes, no, nonat,
I want to add to my dialplan the ability to spy on an arbitrary
extension whether a call originates at it or is terminated at it.
Scenario 1: Given an extension, say 2001, a call comes in on a zap
channel and is Dial()ed to the phone that's at extension 2001, I want to
be able to pick up a phone
On Wed, 2008-04-16 at 13:47 -0700, Steven Kurylo wrote:
exten = s,n(getext),Read(SPY,extension,4)
exten = s,n,GotoIf($[ ${LEN(${SPY})} != 4 ]?nospy)
exten = s,n(spy),UserEvent(ChanSpy,User ${CALLBACKNUM} spied on ${SPY})
exten = s,n,Chanspy(SIP/${SPY},r(monitor-ext-${SPY}))
exten =
I have just noticed the L() argument to Dial() and it seems pretty
obvious that this could be used to create a lightweight prepaid calling
system.
I'm wondering if anyone has some extensions.conf dialplan using
Dial(..., L(...)) and the astdb to do lightweight prepaid service. I
only need to
On Thu, 2008-04-17 at 07:16 -0400, sil wrote:
Simon wrote:
| Is this worth doing? If so, what ports should i specifiy?
http://www.bricklin.com/qos.htm
Yeah, well, that's all fine and dandy as long as more capacity is an
option. Many people are already subscribed to the most capacity
On Thu, 2008-04-17 at 07:54 -0400, sil wrote:
Apparently man people don't understand that those QoS settings on
routers mean little most of the time. Most providers resell QoS as a
premium service, so while many waste their time painting their packets
those markings get stripped.
Maybe your
On Thu, 2008-04-17 at 08:36 -0400, J. Oquendo wrote:
Brian J. Murrell wrote:
| But certainly at my choke point which is of course my Internet uplink,
^^^
I
| can apply QOS (i.e. traffic shaping, which is what the OP's
On Thu, 2008-04-17 at 09:25 -0400, J. Oquendo wrote:
Is it? So you're telling me if you're saturated on the way in, fixing up
your packets on the way out is the solution.
I think I've made it clear that my argument is only about uplink shaping
and the requirement for it given the asymmetric
On Thu, 2008-04-17 at 11:40 -0400, Chris Mason (Lists) wrote:
Mike wrote:
do a decent
job of providing QoS on the upstream, which is where you (usually) need it
anyways.
QOS can only be on outgoing,
Which is what he meant when he said upstream I believe.
you can't set the priority
On Wed, 2008-04-16 at 19:44 -0400, Brian J. Murrell wrote:
I'm wondering if anyone has some extensions.conf dialplan using
Dial(..., L(...)) and the astdb to do lightweight prepaid service. I
only need to meter a handful of users.
Since I asked and nobody else answered (although I know you
On Tue, 2008-04-22 at 17:58 -0500, Security Officer wrote:
Asterisk Project Security Advisory - AST-2008-006
So given that I'm new to asterisk's svn and bug tracking tool, is it
sufficient then to apply the two patches (iax_dcallno_check-1.2.rev3.txt
and iax_dcallno_check.rev9.txt) listed in
On Tue, 2008-04-22 at 20:34 -0400, Brian J. Murrell wrote:
So given that I'm new to asterisk's svn and bug tracking tool, is it
sufficient then to apply the two patches (iax_dcallno_check-1.2.rev3.txt
and iax_dcallno_check.rev9.txt)
Ahhh. I see. These must be two versions of the same patch
On Wed, 2008-04-23 at 01:06 -0400, Matt Watson wrote:
I can;t imagine what headaches you'd have going from 1.4.11 to 1.4.19.1...
that is a minor version upgrade... no real change in functionality
Yeah, that's the theory anyway. :-)
thats basically 8 versions of bug fixes...
And what
On Tue, 2008-04-22 at 22:59 -0700, Nhadie Ramos wrote:
i want to create a billing system to monitor only the trunks and also
to load amounts on those trunks. is this possible? will i be able to
use app_prepaid for this?
TBH, I don't really understand your description, but I will say that I
On Wed, 2008-04-23 at 09:38 -0700, Nhadie Ramos wrote:
Hi, sorry to confused you with my question.
the normal prepaid application like astcc, if i'm not mistaken, monitors the
amount left on the user (which i usually refer as extension), what i want to
do is monitor prepaid on the trunk
On Wed, 2008-04-23 at 08:52 -0500, Tilghman Lesher wrote:
Please understand that that's NOT the only security fix that has gone in
during that time. If this is the only thing that you fix, you're likely to be
vulnerable on several other levels. See our full list of security disclosures
at
On Wed, 2008-04-23 at 15:41 -0600, Darren Wiebe wrote:
Ok, I'm not aware of this feature in astcc
Keep in mind that astcc is simply a tool that keeps a database of
minutes used for some entity (typically a calling card) and calculates
those minutes used against a pre-charged amount. The number
On Thu, 2008-04-24 at 09:13 -0500, Tilghman Lesher wrote:
Check the archives.
Indeed, you are correct. My apologies. I forgot that I temporarily
unsubbed from the -users list for a period of time where I was just
getting too much volume of e-mail and asterisk-users had to be one of
the ones
On Wed, 2008-04-30 at 08:56 -0700, Andreas van dem Helge wrote:
Could someone please add to the documentation that Zaptel is required
for SLA to work? It becomes sort of frustrating when you read the
documentation a few times, keep on trying to get the thing to work for
a few hours only to
On Thu, 2008-05-01 at 18:25 -0400, SIP wrote:
How about some prepaid balance-related ones that aren't
calling-card-specific.
Indeed, it would be nice to see the sounds supplied in the astcc package
done by Allison.
b.
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On Mon, 2008-05-05 at 16:36 -1000, Julian Yap wrote:
That was a bug in the release.
From the 1.4.20-rc1 Changelog:
2008-04-30 16:30 + [r114891] Russell Bryant [EMAIL PROTECTED]
So basically, r114891 was a fix to AST-2008-006? So if you applied the
patch for AST-2008-006 you now really
On Tue, 2008-05-06 at 13:23 +0100, Julian Lyndon-Smith wrote:
Yes.
Hrm. For those of us that are following along the AST-* train, patching
as per the AST-* release notices, as a matter of process, wouldn't it
have been good to republish AST-2008-006 and include this fix along with
the original
On Tue, 2008-05-06 at 08:42 -0500, Tilghman Lesher wrote:
It's not actually a fix to the security fix.
No, indeed.
The security fix simply
highlighted an issue which was already present in Asterisk.
That may be true, but the security fix now depends on that new fix, so
it's tangentially
Does anyone have a better ENUM lookup handler than the built-in
ENUMLOOKUP() function? The built-in function does not properly handle
multiple return values such as:
8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 u E2U+SIP
!^\\+1866(.*)$!sip:[EMAIL PROTECTED] .
On Wed, 2008-05-07 at 14:26 +0200, Johansson Olle E wrote:
Quoting RFC 3824:
Only one SIP URI, ideally, appears in an ENUM record set for a
telephone number. While it may initially seem attractive to
provide multiple SIP URIs that reach the same user within ENUM,
if
On Wed, 2008-05-07 at 08:21 -0400, Matt Watson wrote:
There is a enumlookup.agi that is included with FreePBX and thus trixbox, PBX
in a flash, etc. etc.
Yeah, I had gotten that impression somewhere too.
If you have trouble finding it let me know and I can send you it.
If you would be so
On Wed, 2008-05-07 at 22:54 +0300, Tzafrir Cohen wrote:
Slightly off-topic:
Yeah.
On Wed, May 07, 2008 at 10:29:47AM -0400, Brian J. Murrell wrote:
I guess a code audit will tell. :-) Although I got an impression that
it was written in PHP. I'm not much of a fan of PHP. Don't really
On Wed, 2008-05-07 at 13:40 -0700, John Todd wrote:
1) The ENUMLOOKUP function is currently being fixed for TRUNK.
Ahhh. Sweet. I wonder how difficult a backport will be.
Take a look at http://bugs.digium.com/view.php?id=8089 for the
current status. Testing would be appreciated.
Will
To this end, I have taken a first pass at a Perl AGI script to look up
and return a list of URIs for a given phone number. I will not pretend
that I have read the relevant RFCs but have implemented based on the
knowledge I have gathered about ENUM lookups from various sources.
Given my dialplan
On Thu, 2008-05-08 at 10:51 -0500, Russell Bryant wrote:
Have you taken a look at the ENUMQUERY() and ENUMRESULT() functions that are a
part of Asterisk 1.6?
I have not even entertained thinking of 1.6 yet. :-/
The ENUMQUERY() function lets you do a single enum query for a number. Then,
On Thu, 2008-05-08 at 10:51 -0500, Russell Bryant wrote:
Have you taken a look at the ENUMQUERY() and ENUMRESULT() functions that are a
part of Asterisk 1.6?
The ENUMQUERY() function lets you do a single enum query
From a single zone it seems. So that means a for zone in $ZONES type
of
I have a Wildcard FXO: Wildcard X100P (clone) in my Asterisk (1.4.17)
machine and as of late, Caller-ID on it seems to be failing more
frequently than not. Sometimes I get callerid.c:613 callerid_feed:
Caller*ID failed checksum sometimes it fails without even that.
In Zapata.conf I have:
On Sat, 2008-05-17 at 18:38 +0100, Adrian Marsh wrote:
All,
Does anyone know of a SIP URI direct to googles 800-GOOG-411 service?
Yeah, I suppose a direct SIP connection would be nice.
An enum lookup shows 3 URIs listed, none of them seem to be google
directly,
No, they are SIP-PSTN
On Mon, 2008-05-19 at 11:13 +0100, Adrian Marsh wrote:
Hi Brian,
Thanks for the reply. I tried searching for your posts, but no luck.
I find effective use of the Internet absolutely depends on knowing how
to search for stuff.
On Thu, 2008-05-22 at 14:05 +0200, nik600 wrote:
No, i'm just wondering because there is creating a greater difference
between my installation and the actual Asterisk.
If it ain't broke, don't fix it. You are already so far behind that any
upgrade is going to be a major task of testing and
On Sat, 2008-04-26 at 18:41 -0400, Andreas van dem Helge wrote:
Does anyone have a script for manual wardialer for asterisk? not sure
if wardialer is the correct term but basically I want to call X
number say 555- through 555-0050 and be able to listen to each
call and when I hang up or
Right now I have an Asterisk 1.4.18ish server and a Wildcard POTS
interface. As it is now, when the zap line gets a call, Asterisk
answers it and waits for the analog CID to be presented, then rings the
SIP phones with the call and the CID. There's a significant latency
involved in doing this.
On Wed, 2008-06-11 at 10:12 -0400, Steve Totaro wrote:
Do you actually have callerID on your line? That takes about two
seconds. Try removing it and see how much faster Asterisk answers.
That brings up a question though, on a regular landline with caller ID
the phone rings right away, it
On Wed, 2008-06-11 at 10:38 -0400, Steve Totaro wrote:
Exactly! It is funny how when idea or technology is ready, many
people have the same thougts at the same time.
Indeed. But what is even more interesting is that this technology is
not just ready. It's been ready for a long time and
On Wed, 2008-06-11 at 15:57 +0100, Gordon Henderson wrote:
Intersting idea... However, I live in a country where on a regular
landline with caller ID, the caller ID is displayed before the phone
rings, so make sure it's an option and not hard-wired...
Well, I think your situation makes the
On Wed, 2008-06-11 at 13:30 -0500, Brent Davidson wrote:
On the subject of CallerID and ringing, I'm not sure if it's like this
everywhere in the US, but where I live in Texas, our caller ID signal
is sent between the first and second rings.
It's like that here in Canada too.
If the phone is
On Wed, 2008-06-11 at 14:51 -0400, Steve Totaro wrote:
If you ever have problems with a call dropping after 30 seconds,
Answer() is usually the cause.
Interesting. I can't say that I've ever had that problem.
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On Sun, 2008-06-15 at 17:43 +0300, James Mutuku wrote:
Please advice on channel bank
Dude. There's the cool new website you should check out. It's
www.google.com.
Seriously. This list is not full of people waiting to do the simplest
research at your request. Spend a few minutes and do some
On Sun, 2008-06-15 at 11:03 -0400, Steve Totaro wrote:
On Sun, Jun 15, 2008 at 10:53 AM, Brian J. Murrell
[EMAIL PROTECTED] wrote:
On Sun, 2008-06-15 at 17:43 +0300, James Mutuku wrote:
Please advice on channel bank
Dude. There's the cool new website you should check out. It's
I have an outbound-ld context as follows:
[ Context 'outbound-ld' created by 'pbx_config' ]
'_1NXXNXX' = 1. Macro(enumdial|${EXTEN}) [pbx_config]
102. Wait(1) [pbx_config]
103.
On Wed, 2008-06-25 at 11:25 -0500, Tilghman Lesher wrote:
That's only true within the same context. ONLY if a match is not found in the
current context will it go into an included context.
Ahhh. Well, then that explains it. Any thoughts on how to achieve my
goal, without having to encode all
On Mon, 2008-06-30 at 12:03 -0400, Andrew Joakimsen wrote:
Does anyone know of a spam filter that will work with Asterisk?
What does spam have to do with Asterisk? Or do you mean spit perhaps?
http://en.wikipedia.org/wiki/VoIP_spam ? Probably the same techniques
such as whilelisting,
On Mon, 2008-06-30 at 11:15 -0500, spectro wrote:
I need a way to block that IP from connecting to my
asterisk server, please advice.
netfilter. aka iptables.
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I have a wildcard 100 xp on my pots line and all was working just fine
up until a few days ago when all of a sudden it stopped receiving caller
id on incoming calls. I know caller id is being presented on the line
as the analog set on the same line always gets it.
What is strange is that this
On Sun, 2008-07-13 at 10:22 -0400, Brian J. Murrell wrote:
I have a wildcard 100 xp on my pots line and all was working just fine
up until a few days ago when all of a sudden it stopped receiving caller
id on incoming calls. I know caller id is being presented on the line
as the analog set
On Tue, 2008-07-15 at 22:31 +1000, Rob Hillis wrote:
Brian J. Murrell wrote:
One thing I have noticed is that in the cases where the wildcard cannot
determine the CID (i.e. because the rxgain is up around 10.5), I get
this in my asterisk console:
[Jul 15 08:04:09] NOTICE[26696
On Tue, 2008-07-15 at 12:49 -0400, Noah Miller wrote:
It is odd that it would work one day and not the next.
Indeed.
I'd have to
say, though that I've seen that rxgain/txgain values beyond +-8 seem
to yield unpredictable results in many areas,
Yeah, I was pretty alarmed months ago when I
On Wed, 2008-07-16 at 15:08 +1000, Rob Hillis wrote:
I hadn't realised this was for a home server... yes I agree, for a home
server the Digium or Sangoma cards are a little too expensive.
Indeed.
I can't speak for the SPA-3102, but the SPA-3000 I use here at home
doesn't do a brilliant
On Wed, 2008-07-16 at 09:31 -0400, Brian J. Murrell wrote:
IIRC -- this was 7+ years ago
Er, I mean 2+ years ago, just to keep the facts straight.
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On Thu, 2008-08-14 at 14:15 +0200, Klaus Darilion wrote:
Use the ENUMLOOKUP function, e.g.:
And take note that it's very naive. See my previous posting for an enum
AGI that is more intelligent. The only thing it does not do that I
would like to add is give up on the DNS lookup much earlier
On Thu, 2008-09-25 at 20:49 +0300, Tzafrir Cohen wrote:
Great news! You mean that there is finally a free implementation of the
skype protocol so I can start using it?
Free? AFAICT, not. Neither free as in beer nor speech. Move along,
nothing to see here.
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So, I have been testing ekiga 3.0 with Asterisk, and sadly, it don't
work. I am told by the ekiga devs in
http://bugzilla.gnome.org/show_bug.cgi?id=553595 and
http://bugzilla.gnome.org/show_bug.cgi?id=553810 that the problem is
that Asterisk does not support SIP forking.
The issue is that I have
On Thu, 2008-09-25 at 14:56 -0400, SIP wrote:
Sending from multiple different points of origin doesn't make any sense
at all in either a logical or rational fashion. What's it supposed to
accomplish?
It seems to be a shot-gun approach to making a SIP connection. The
assumption being I suppose
On Thu, 2008-09-25 at 15:31 -0400, SIP wrote:
That strikes me as being careless and unreliable.
That's one argument. I can also see the ekiga developers' argument
though and that's to strive for the most automatic functionality
possible. The less things you have to ask users, the more likely
On Thu, 2008-09-25 at 17:25 -0700, Fred Posner wrote:
I talked with both Skype and Digium today at Astricon for a while on this...
it's actually going to be amazing.
It's still early, but still, nobody has answered my question as to
whether Skype will be using my Asterisk server's CPU and
On Fri, 2008-09-26 at 08:43 +0100, Grey Man wrote:
It's not particularly difficult to determine the best IP address for a
piece of client software to use.
Oh?
Check the local machines default
gateway, apply the subnet mask and then compare it against all the
local IP's.
Yeah? And if
On Fri, 2008-09-26 at 14:54 +0100, Grey Man wrote:
On Fri, Sep 26, 2008 at 2:36 PM, Brian J. Murrell [EMAIL PROTECTED] wrote:
Yeah? And if more than one matches? Then what?
Use one of them!
And if the one I choose to use doesn't work because of some kind of
policy routing or filtering
On Fri, 2008-09-26 at 10:16 -0400, SIP wrote:
The RFCs are there for a reason. All SIP forking is UAS territory. Not
UAC territory.
From http://bugzilla.gnome.org/show_bug.cgi?id=553810 Damien Sandras
asks:
I repeat, Ekiga is doing something perfectly legal.
The real
On Fri, 2008-09-26 at 10:41 -0400, SIP wrote:
Oh yes. It's perfectly legal.
It's also a) NOT SIP forking, b) Lazy, and c) Poorly designed.
Sending multiple requests and hoping and praying that the recipient will
ignore two of them (it will NOT in many cases -- specifically set out by
the
On Sat, 2008-09-27 at 15:11 -0400, Dean Collins wrote:
Who is Chris Langford in Huntsville Alabama and is he seeking Digium’s
permission in order to report the asterisk mailing lists out onto the
internet
Digium's permission? Does Digium own the copyright on what I write? I
think not. I do.
I'm looking into getting a new phone and wondering what the difference
in functionality is between a single line phone with call waiting and a
real 2 line phone (either a real SIP phone or an analog 2 line phone and
a 2 port ATA) is. Why would I want the real 2 lines vs. just being able
to take
On Tue, 2008-09-30 at 08:23 -0500, Lyle Giese wrote:
1) a two line phone can register with two different * servers or sip
carriers.
Indeed. But if I only had the one * server which itself registered to
my carriers...
2) It's easy for both incoming and outgoing to separate business from
On Tue, 2008-09-30 at 17:29 -0500, Lyle Giese wrote:
I have never been convinced that VM via email is a convenence. You
have to use the loudspeakers on the PC or headphones, which is not as
convenient as a handset.
Depends on your working environment I guess.
Not to mention the privacy
I'm looking at the app_voicemail.c from both 1.4 and 1.6.1 and seeing
that neither allows an individual mailbox to override the imapfolder
value. It seems entirely intuitive to me that one might want to do
that, not to mention how trivial it looks to add that to
app_voicemail.c.
Maybe my
Given the following SRV records:
_sip._udp.tollfree.sip-happens.com. 38400 IN SRV 10 0 5060
sometimes.sip-happens.com.
_sip._udp.tollfree.sip-happens.com. 38400 IN SRV 20 0 5070 ares.sip-happens.com.
Why is asterisk (1.4.17) not honouring the priority and not failing over
to using other records
On Fri, 2008-10-17 at 10:32 -0500, Andres wrote:
Because Asterisk does not support that.
Which is just another way of saying Asterisk is broken then. SRV
records have requirements for their correct use. If those requirements
are ignored, that is a broken implementation.
The only thing that
On Fri, 2008-10-17 at 11:18 -0500, Eric ManxPower Wieling wrote:
It should be fairly easy to write an AGI script that does the SRV query,
do whatever you want with the response, set a channel variable with
the results and use that in your dialplan.
Maybe. If I were an AGI hacker. But
On Fri, 2008-10-17 at 11:35 -0500, Eric ManxPower Wieling wrote:
If you fight Asterisk's oddities then you will have a depressing and
miserable life. If you embrace Asterisk's oddities then you will have a
joyous and enlightened life. 8-)
I just want something that works. :-)
I agree
On Fri, 2008-10-17 at 12:11 -0500, Tilghman Lesher wrote:
Have you considered upgrading to 1.6?
Not to this point, no. 1.4 does everything I want and if it ain't
broke, don't fix it. Well, now it's broke I guess. Still, Ubuntu still
uses 1.4 and I don't like having to maintain my own
On Sun, 2008-10-19 at 13:31 +0300, Kevin P. Fleming wrote:
Asterisk 1.6 supports proper SRV record sorting, so that the lookups
will return the correct record to the module that requested the lookup.
That would be in ast_get_srv() then? In 1.4.17 I do see weight
processing but not priority
On Sun, 2008-10-19 at 21:14 +0300, Kevin P. Fleming wrote:
Never mind... I was mistaken. The srv_callback() function puts the
records returned by the DNS lookup into priority order (lowest numbers
first),
Yes, I can see that function and while I have not audited it detail, it
looks as though
I've configured asterisk 1.4 to use imap storage for voice-mail and
while I'm happy with it generally speaking it really seem to hammer the
IMAP server. It appear, from the IMAP server logs that it's polling
the imap server every *second* for mailbox updates for the users'
voice-mail folders.
On Mon, 2008-10-27 at 10:11 -0400, Brendan Martens wrote:
I found this in the sample voicemail.conf:
;pollmailboxes=no; If mailboxes are changed anywhere outside of
app_voicemail,
;; then this option must be enabled for MWI to
work. This
;
On Mon, 2008-10-27 at 11:13 -0500, Mark Michelson wrote:
The behavior you are seeing is most likely due to SIP's MWI behavior in
Asterisk
1.4. The way it works is to poll the mailboxes every so often to see if new
messages are available.
Yeah, that sounds like it.
2. If you want to
So I have (and have had) jabber configured for some time, specifically
for GTalk, but something has occurred to me. If somebody happens to
send an IM (text) to that account, nobody is going to be receiving it.
I'd like to send a canned message back to any sender of an IM.
Possible?
b.
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