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[EMAIL PROTECTED]
(936
On Mon, 24 Apr 2006, Douglas Garstang wrote:
You can't use round robin DNS. Round robin DNS will cause every SIP packet to
potentially go through a different static path, which will break things.
Huh? Has this happened to you in practice?
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users
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hink else
I think the non numeric username is the problem.
Yes, I have done an restart of Asteriks after changing the sip.conf.
Am Wednesday 19 April 2006 23:03 schrieb Aaron Daniel:
Have you tried sending it to a different extension number? I've got the
registrations working on my home
s Winter wrote:
Hi,
[general]
context=Sip_in
register => 1234:[EMAIL PROTECTED]/s
s is the same, it still looks for an extension 1234 in the context Sip_in and
did not use /s
Asterisk is 1.2.7
Am Wednesday 19 April 2006 22:48 schrieb Aaron Daniel:
I'm not gonna say much for the do
digium.com/mailman/listinfo/asterisk-users
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is the 8.2 firmware shows @ip-address at the end of the number
which doesn't do well in the directory on the phone and makes it difficult
to just redial missed numbers.
Are you talking about incoming calls with your php script, or the actual
directory that's being loaded from the p
or not.
The directory on the Cisco's doesn't link up at all with the phone, other
than allowing you to dial the numbers, or save numbers from your missed
call list. There's a lot of stuff Cisco crippled in the SIP version of
the firmware that's kinda like "uh, what we
Hi,
I'm having problems with the voicemail, the app keeps exiting in 3-5 seconds.
Any considerations will be appreciated.
Thanks,
D.K.
Debug messages:
Apr 18 20:53:41 DEBUG[26150] pbx.c: Launching 'Goto'
Apr 18 20:53:41 DEBUG[26150] pbx.c: Launching 'VoiceMail'
Apr 18 20:53:41 DEBUG[26129] chan_
Hi,
when I call the voicemail app, it starts and die suddenly. Has anyone
already had this problem?
Log:
app.c:644 ast_play_and_record: No audio available on SIP/-6fca??
-- User hung up
Tks,
D.K.
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arding redundancy, and how people are
solving these problems.
Regards to all,
Joe
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Thanks Matt...
D.K.
On 4/13/06, Don Pobanz <[EMAIL PROTECTED]> wrote:
> Thank you Matt!!!
>
> Matt Roth wrote:
> > Try switching to native MOH. You'll eliminate the decoding of the MP3s
> > and the host of problems that come along with using mpg123. The MOH is
> > handled by the same thread tha
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On Fri, 14 Apr 2006, Martin wrote:
I too had a server room fry and need to replace h/w.
So what specific Dell servers did/do you deploy?
Where is the link w/Digium/s Dell caveats?
I'm using the Digium TDM400 card w/*
Date: Thu, 13 Apr 2006 19:19:13 -0500 (CDT)
From: Aaron Daniel <[EM
pointed customer is a lost customer forever... Too sad...
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or gateway... don't really know
where to go from there. I did turn vpmdtmfsupport off and that didn't
help at all.
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I had this problem with 1.2.5, 1.2.6 and now with 1.2.7...
On 4/13/06, Gareth Blades <[EMAIL PROTECTED]> wrote:
> What version of Asterisk?
>
> On Thu, 2006-04-13 at 12:38, Daniel Korndorfer wrote:
> > Hi,
> > i'm having problems with the MOH module. In a queue so
Hi,
i'm having problems with the MOH module. In a queue sometimes it just
stop playing, does anyone have some idea what could be wrong?
No verbose data...
Tks,
Daniel Korndorfer
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both of us on this).
Curt
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to features for
the site? Wiki? Forums? ??
Bruce
Nortex Networks
On 4/12/06, Greg Camp <[EMAIL PROTECTED] > wrote:
I'm in Lubbock. A little closer to Amarillo than Dallas.
Thanks,
Greg
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Have you quit and relaunched Asterisk? (not a reload, but a full quit
process and restart) I know in the past when I have a process
already listening to 0.0.0.0 it will not always pick up a newly added
NIC alias address without re-binding.
Daniel
On Apr 11, 2006, at 12:21 PM, Michael
Scratch that :) Figured it out.
On Tue, 11 Apr 2006, Aaron Daniel wrote:
Just doing some test installs of asterisk running on branch (noticed first on
branch), and noticed if you move to virtual terminal 9 (may be different on
everyone else's), the CLI is running. Anyone have any idea h
Just doing some test installs of asterisk running on branch (noticed first
on branch), and noticed if you move to virtual terminal 9 (may be
different on everyone else's), the CLI is running. Anyone have any idea
how to turn this off?
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Sam Ho
?
I can dial 1002 from 1001 and that works fine.
I use context= from-internal
My box is [EMAIL PROTECTED] 2.7
Hope you can help me out here.
Daniel
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To
with asterisk" mailing list? I can understand not wanting it
on asterisk-dev, but asterisk-users?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron
Daniel
Sent: Thursday, April 06, 2006 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial Disc
Next time you better ask chan_sccp related questions on the
chan-sccp list
Shoot me for not wanting to subscribe to yet another mailing list when
someone on here might have the answer.
Guess I won't ask if anyone's gotten ringing notification working on it.
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Comput
e his time if it is going to just be a "yeah, try
our product" booth and not something he can spend time talking to
them about what it can/can't do, see it in use, etc.
Daniel
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ook wrote:
Are you using chan_sccp for you cisco implementation?
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;vm-intro' (language 'pt')
== Spawn extension (macro-ramais_sip, s, 224) exited non-zero on
'SIP/200.234.208.250-0840f548' in macro
Here are the "show voicemail users for company" results
ContextMbox User Zone NewMsg
company
risk-users
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Ok, so multiple people have said that hinting is possible with chan_sccp
on the 7940/7960's and such, has anyone got this working? How do you go
about getting this to work?
I'd use the wiki, but it's link to the mailing list topic on that doesn't
work anymore :(
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line.
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hint for the _watched_ extension like this:
exten => 2348,hint,SIP/2348
Let me know if you have any more questions.
Regards,
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information to the phone after it
subscribes.
Aaron
On Mon, 3 Apr 2006, Kevin P. Fleming wrote:
Aaron Daniel wrote:
Ok, with the buddies, what "device" do you hint to? The last line of
the phone?
I don't understand the question... the 'buddy' is effectively a
s
... for Zap, it's obvious, but for SIP and IAX2, it is less clear.
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Of the people in here that have hinting working with the polycom 601's (or
any phone for that matter)... do you have it working so that the shared
line appearance shows that there's someone on the phone? If so, any hints
on how to do it?
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thoughts and opinions of everybody that has contributed to this
discussion.
Regards,
Daniel Hazelbaker
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Hello Dinesh
I got a Panasonic KX-TDA100, can you tell me please how can you
configure the PBX side? Qsig slave? master? and the other side of the
asterisk? I got TE100P
Regards,
Daniel
Dinesh Nair wrote:
On 03/31/06 19:49 Wolfgang Zweimueller said the following:
My conclusion with
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m.com/mailman/listinfo/asterisk-users
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Hello, Has anyone test how to automatically rejects calls on the blacklist?
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+LookupBlacklist
Seems not working
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will tell them the status
of a phone.
Daniel - Good to hear that people from the manufacturing companies
traffic these lists!
On Mar 28, 2006, at 6:29 PM, Christian Stredicke wrote:
Well the problem with the sidecar is simple. Just try to light all
lights three times within one second. I
call came from.
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going with a different solution later. The phone would still be
perfectly good.
Daniel
On Mar 28, 2006, at 8:12 AM, Bob McDowell wrote:
Very true. I am currently debating whether or not to offer it as an
option for my employer's system. As it currently stands, we do not
have
eve
st do 12 per phone?
Daniel
On Mar 27, 2006, at 2:28 PM, <[EMAIL PROTECTED]>
<[EMAIL PROTECTED]> wrote:
Yes - set up about 10 of them at a business last year.
Monitoring is fine - picking up calls is a bit iffy at the best of
times.
(that is, picking up a ringing call by push
Yes, I keep reading on the mailing list archives and the wikis that
(wether or not it is indeed a Asterisk issue) Polycom keeps saying
that an issue with Asterisk prevents you from monitoring more than 7
total (not per sidecar) extensions.
Daniel
On Mar 27, 2006, at 12:08 PM, Justin Moore
We may end up using a software solution, but there are two main
issues with a software solution (for us at least):
1) For us in particular, our receptionists have ALWAYS (for the past
15 years at least) used a physical switchboard style for "routing"
and seeing availability. From past hard
like such a common thing.
Daniel Hazelbaker
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How can I edit the DB?
Tamás Bondár wrote:
OK, if I see well, this is the key idea here:
exten => 9,1,DBPut(CallBack/${${UNIQUEID}}=${CALLERIDNUM})
that is, putting the caller and callee number into AstDB under the CallBack
family.
Can you confirm that Asterisk takes care of the rest? If
How does the hinting work on the polycoms? I've got a polycom set up with
hinting, I can see when the shared line rings, but I can't tell if
someone's on the line. Any suggestions?
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[EMAIL PROTECTED]
t;." and "*" right. Never put a 0 timeout on "*" or nothing else
will work right.
Hope that helps.
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a good phone brand that actually IS Asterisk
compatible.
Daniel
On Mar 26, 2006, at 12:01 AM, Jared Valentine wrote:
I would not recommend the 3Com phones for use with Asterisk.
3Com 3100 series phones do not support SIP with non-3Com systems.
They have
a basic boot loader which must downloa
t;" msg.mwi.6.subscribe=""
msg.mwi.6.callBackMode="disabled" msg.mwi.6.callBack=""/>
Is there anything wrong?
Thanks,
Kevin
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sk.
3Com 3101 (model with speakerphone)
3Com 3102 Business Phone
3Com 3103 Manager Phone
3Com 3105 Attendant Console (if these don't work, can somebody
recommend another receptionist alternative?)
Daniel Hazelbaker
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las.
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, March 24, 2006 4:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Transferring a call with IAX
Ok, add g to the option list on the dial:
Dial(IAX2/[EMAIL
ER. If I shut my ACD server down, I get
CHANUNAVAIL, and THAT jumps to s-CHANUNAVAIL.
*sigh*
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, March 24, 2006 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Trans
I
shut my ACD server down, I get CHANUNAVAIL, and THAT jumps to s-CHANUNAVAIL.
*sigh*
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, March 24, 2006 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Transfe
tters/e)
exten => s-ERROR,9,Playback(digits/9)
exten => s-ERROR,10,Playback(digits/0)
exten => s-ERROR,11,Playback(digits/0)
exten => s-ERROR,12,Set(i=$[${i} + 1])
exten => s-ERROR,13,EndWhile
exten => s-ERROR,14,Hangup()
exten => s-OK,1,NoOP(CONTROL BACK INSIDE MACRO)
exten =&
ch should return control back to where
the Macro was called from! How weird.. it looks like I _AM_ getting control
back, sort of...
Doug.
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, March 24, 2006 4:07 PM
To: Asterisk Users Mailing List - Non-Commercial
t;, "QUEUE DONE") in new stack
-- Executing Hangup("IAX2/216.187.142.203:4569-5", "") in new stack
... on the caller:
-- Hungup 'IAX2/acdserver1-3'
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, March 24, 2006 3:52 P
rver doing it. I thought you where doing something
similar?
Douglas.
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, March 24, 2006 2:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Transferring a call
get this to work.
Doug
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, March 24, 2006 1:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Transferring a call with IAX
Heh, lots of voodoo... I've got a dra
Well, that's kinda frustrating :) Time to start digging.
Aaron
On Fri, 24 Mar 2006, Douglas Garstang wrote:
Prtty darn sure it isn't an option with the Polycoms.
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, March 24, 2006 1:23 PM
To
em. I have to dial the VM server
from the ACD box. I don't understand how that could work anyways. Once you've
transferred the call, you've transferred it.
What voodoo are you using?
Doug.
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[EMAIL PROTE
that backup proxy if the primary was
unavailable? In that case, they'd only be registered with a single proxy, not
two.
We're using polycom phones... they support DNS SRV and seem to work (mostly)
well. Phones haven't seemed to have been the issue.
-Original Message
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Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, March 24, 2006 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Failover without SER
We've actually got two servers handling all the call volume,
and when
with this sort of setup without the use
of SER?
Bryan Mahin
Rediscover Personal Service with UNETA
Please visit us @ www.uneta.com
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mpanies a product of this
type. And with Asterisk it's worse because it gets Linux FUD as well as
VoIP FUD.
Bob McDowell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron
Daniel
Sent: Friday, March 24, 2006 11:50 AM
To: Asterisk Users Mail
t the removing identifying information part is such a good idea, since
the best way for people to trust a system is to talk to people that have
used it before. Or do we just want the information to filter through the
asterisk-users list?
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info. I can understand the cost factor. What
phones are you switching to?
Thanks!
Lacy
On 3/23/06, Aaron Daniel <[EMAIL PROTECTED]> wrote:
The 1300 phones we're moving over in the next two months are being moved
off of cisco. The reason we're moving them over is a) cost and
..
0004f2030925.cfg
then I have phone4701.cfg that contains all of the line information
and phone specific data
then the stock sip.cfg with the digitmap and global options
Sean
Aaron Daniel wrote:
Does anyone have the polycom soundpoint ip's successfully remotely
provisioning?
ne for an Enterprise
Class piece of software are unacceptable, or maybe reacted in incredulation
that it would even be considered being done in a certain way.
It's quite obvious that any negative statements about Asterisk are not taken
too well.
-Original Message-
F
I'm asking is that if you don't like it, or think it
needs something better, let us know in a manner that people will be
receptive of it. I'm sure you're very capable in what you're doing, and
everyone's been through the frustration before. I fought with T1 lines
a lot of
what he's going through, especially with a project with as high a learning
curve as asterisk, so I know where he's coming from, but when you're
asking for help, ask for help, don't dog the project. That's all I was
really trying to get at :)
Sorry if I upset
Yeah, everything but the individual phone configuration is working fine.
The logs are uploading and everything. I'll look at those more closely
tomorrow.
Aaron
On Fri, 24 Mar 2006, Avi Miller wrote:
Aaron Daniel wrote:
voicemail stuff is working, just not the registration inform
stang wrote:
Please don't tell me what I think you are. Are you saying that to change a
configuration setting for the phone I have to remove it as a peer, and then
wait for it to re-register? Are you serious???
Doug.
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECT
know it's pulling those files? Is there an error reported
by the phone reading those files? Maybe a typo in the xml files?
On 3/23/06, Aaron Daniel <[EMAIL PROTECTED]> wrote:
Does anyone have the polycom soundpoint ip's successfully remotely
provisioning? I've got the phone pu
extens in such a created [context_name]
are not seen or used by Asterisk to dial out.
There is something missing.
Larry
Aaron Daniel wrote:
Yes.
Just create a context that you want the phones to dial from in
extensions.conf.
[context_name]
exten => 1234,1,Dial(SIP/1234)
exten => 12
Does anyone have the polycom soundpoint ip's successfully remotely
provisioning? I've got the phone pulling default configs, and it's
downloading phone specific information, but it's not actually using that
information. Any help would be appreciated :)
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ps.
Thank you,
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lt] and I'd like to avoid that.
Larry
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, don't respond negatively when you post something to
the list and people ask for information Doug.
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If I add or remove a pickupgroup or call group from a phone in the
database, I need to sip reload.
No you don't... sip prune realtime works like a charm. The single
phone re-registers and the new info is in the system. No reloading
required :)
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ne (and this is probably not a
concern for most people) but the SIP firmware doesn't let you dial before
picking up the phone, but SCCP does. Not a show stopper, but just one of
the annoying things they've done.
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Aaron Daniel
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Sam Houston State University
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Aaron Daniel
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I
Texoma Healthcare Systems
903.416.4398
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Aaron Daniel
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Sam Houston State University
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o rely on some remote forum to keep the information for me.
Just my 2 cents :)
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Aaron Daniel
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Sam Houston State University
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Uhm... Why?
Doug
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Comput
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Aaron Daniel
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the developers, the only thing this causes problems with
is phones behind NATs. I don't remember us ever having problems using the
same DB for sip registrations ever, even for NAT'ed phones. I think it's
one of those "try and see" type things, cause we tried and
f messages that causes that problem. Nobody here would
notice this "fart" you speak of :-P
Aaron
On Wed, 22 Mar 2006, Andrew Kohlsmith wrote:
On Wednesday 22 March 2006 10:21, Aaron Daniel wrote:
Yeah, once they re-register after the default time period, they come back.
We've got our
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