want
line 2, I'd remove any references to it in your MACADDRESS.cfg files.
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Dave Cotton [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, November 19
I sure like my Aastra 390, the voicemail ADSI app works pretty well (only a
couple of incompleted functions, like not exiting by hanging up the
speakerphone, rather than go to a reorder tone.
As for the 'look at the wiki' comment, I'm not trying to get on anyone's
badside, but Dmitry was
PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, October 27, 2005 12:07 PM
Subject: Re: [Asterisk-Users] please recommend phones with adsi.
On 10/27/05, Chris Coulthurst [EMAIL PROTECTED] wrote:
I sure like my Aastra 390
If your 1.0.9 install is (on the /usr/src/asterisk tree) complete, you might
unpack the CVS source somewhere else other than /usr/src (maybe
/usr/local/src or /usr/src/cvs). Most importantly, PLAN AHEAD. It would
seem that the more Asterisk evolves, the less-tolerant it is natually
becoming
be in ipmid.cfg if you have that file instead
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Wilson Pickett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, October 17, 2005 12:33 AM
Subject
be done in the asterisk dialplan, but it defeats the purpose
of having that nearly useless, unreassignable DND button on the phone!
Take care, polycom config files become less scary the more you read through
em!
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Wilson
FYI, Im using g729 with Teliax, and have been for about 1 week with no
problems, good audio quality.
They DO seem to drop registrations unexpectedly at times, but as for codec
usage, so far so good.
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Rich Adamson [EMAIL
Make sure you have g729 turned on from the Teliax customer panel on their
website.
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com; John Reynolds
Is there an easy (or even a hard) way to save to
the CDR a userfield value with the call's codec in it?
Chris Coulthurst
[EMAIL PROTECTED]
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have nice littlearrows pointing the direction of the 911
caller's dwelling...
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Joel Newkirk [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, September 30, 2005 7:20 AM
Subject: [Asterisk-Users] 911 Q
customers in the future.
So far, looks like Aastra is winning the bid...(barely)
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Kevin P. Fleming [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Wednesday, September 21, 2005
conventionality.
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Matt [EMAIL PROTECTED]
To: Damon Estep [EMAIL PROTECTED]
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, September 19, 2005 6:55 AM
Subject: Re: [Asterisk
Oh btw: here's my voicemail.conf line for just getting the number, not the
name:
emailbody=Fm:${VM_CIDNUM}\n${VM_DATE}\nDur:${VM_DUR}\n
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Matt [EMAIL PROTECTED]
To: Damon Estep [EMAIL PROTECTED]
Cc: Asterisk Users Mailing
something stupid? Is there
another way to upgrade it?
Chris Coulthurst
[EMAIL PROTECTED]
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SetCIDNum(NXXNXX) ? It hasalways worked
for the Teliax lines.
BUT---
It doesn't have a problem making
it to landline phones Ive tried...
I user Verizon for the cell and Qwest for my
incoming analog (with callerID) lines...
Chris Coulthurst
[EMAIL PROTECTED
this
working?
Chris Coulthurst
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The two most common companies to make paging equipment are Viking and Bogen.
Bogen even resells ATAs for paging now. http://www.bogen.com or
http://www.vikingelectronics.com
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users
are
rerouting through less-capable alternates.
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Chris [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, August 30, 2005 6:31 AM
Subject: Re: [Asterisk-Users
What is the problem you are querying about anyway? I've noticed some VERY
bad audio on the circuit when I initiate calls.
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Chris Mason (Lists) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
someone knows the right person to drop the hints to in the
company?
In the meantime, your best bet is really overhead paging amplifiers, which
you can get from Viking that work over Zap channels quite well. It can even
integrate with call park announcing.
Chris Coulthurst
[EMAIL PROTECTED
many things including voicemail
like the pound key. Since *1 is a pre-set option, I assume that you can do
this, as well as set the interdigit requirement?
Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on
2005-08-15 18:11:51 UTC
Chris Coulthurst
[EMAIL PROTECTED
as to stop the playtones, hangup the channel to clear it,
and immediately ring it back with ALERT_INFO=answer/ring-answer. If I
could make that happen, I swear I'll fart sunshine-dust. ;)
Any ideas to move in the right direction, hints and
similar suggestions are more than welcome!
Chris Coulthu
I'd like to have the ringing a caller hears to be
more like a 'british' ring when I am calling an internal extension. The
phones I'm calling already do this, now I'd like to find a way to make the same
thing happen for the caller who waits...
Any ideas?
Chris Coulthurst
[EMAIL PROTECTED
, checked, deleted, and make a change to all of the vm
users at the same time. Some would be redundant, but its not exactly eating
up a ton of cpu cycles to chown users.
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: hugolivude [EMAIL PROTECTED]
To: Asterisk Users Mailing List
to the right
devices.
This, however, would require two miracles, an act of congress, and a note to
the telco from santa claus.
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk
Its left as default, and when I press the # nothing happens, but the remote
caller doesn't hear the DTMF tone.
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Michiel van Baak [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, August 16, 2005 10:05 AM
My suggestion would be, use the externnotify=/usr/bin/myapp feature in
voicemail.conf to chown the permissions to something else. Since they are
root, asterisk should have no problem deleting and moving them around with
less privileges.
Chris Coulthurst
[EMAIL PROTECTED]
- Original
Sounds like you have a DTMF mode problem. Check that you are using RFC2833
for dtmf signaling. I had the same thing happen with my dialing of *98 to
check voicemail..It would transpose it in to 9*8, as if the * was being some
sort of a tab key.
Chris Coulthurst
[EMAIL PROTECTED
,s,1)exten =
i,1,Goto(aa_chris_start,s,1)
Chris Coulthurst
[EMAIL PROTECTED]
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listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo
UNSUBSCRIBE or update options visit:
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,s,1)exten =
i,1,Goto(aa_chris_start,s,1)
Chris Coulthurst
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,s,1)exten =
i,1,Goto(aa_chris_start,s,1)
Chris Coulthurst
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and sleep at night ;)
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Jim Duda [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, August 06, 2005 7:50 AM
Subject: [Asterisk-Users] polycom 301 phone advice
Can anyone tell me if the CallerID
What do I put in voicemail.conf to let me send another user a voicemail
from inside Comedian? I've CVS-HEAD, and the instructions are a bit
ambiguous on the voicemaill.conf.sample. Advanced option 5 is the only on I
don't have, and a very important one to have, indeed.
Chris Coulthurst
THE DOORPHONE
CALL
exten = s,6,Dial(SIP/101SIP/102SIP/104SIP/201SIP/203Zap/2r3Zap/3,22)
;RING SOME PHONES
exten = s,7,Playback(nobody-but-chickens) ; NOBODY'S HOME
exten = s,8,Hangup
Note that you need the first underscore for ALERT_INFO if you are using
CVS-HEAD.
Hope that helps!
Chris Coulthurst
]
; if
not listed, sending messages from inside voicemail will not
be
;
permitted
Why does this say "yes"? Is this referring to
a context in extensions.conf, or is it a voicemail context (i.e.
default)?
Chris Coulthurst
[EMAIL
the 'buddies' button to see them. To
reset it back to working, I just press 'directories' - Contact Directory -
And edit the entry (without changing anything really), and pressing the SAVE
button, and its all back again.
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From
appearances per CO key, rather than the default (I
think it was 8?)
Worked like a champ!
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, August 02, 2005 9:05 AM
Subject: [Asterisk-Users] Polycom phones w
Anyone know where to find a Thai DID to ring in SIP to asterisk?
(probably Bangkok)
Chris Coulthurst
[EMAIL PROTECTED]
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Anyone know where to find a Thai DID to ring in SIP to asterisk?
(probably Bangkok)
Chris Coulthurst
[EMAIL PROTECTED]
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Anyone know where to find a Thai DID to ring in SIP to asterisk?
(probably Bangkok)
Chris Coulthurst
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Anyone know of a place to get a Thailand DID that will ring in to
asterisk in the US at a nice price?
Chris Coulthurst
[EMAIL PROTECTED]
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Anyone know of a place to get a Thailand DID that will ring in to
asterisk in the US at a nice price?
Chris Coulthurst
[EMAIL PROTECTED]
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Anybody know where to find Thailand DIDs that can ring in to my * in the
USA on SIP?
Oh, and a good price, too! ;)
Chris Coulthurst
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information.
Try the -biz list for biz' related questions.
Or, the Wiki has a lot of information:
http://www.voip-info.org/tiki-index.php?page=VOIP%20Service%20Providers%
20by%20Country
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Coulthurst
Sent
If you haven't seen it yet, go here with a Flash
enabled browser:
http://www.theringingmovie.com
Chris Coulthurst
[EMAIL PROTECTED]
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and then
pickup the receiver.
So, I guess in a way, its really a feature! ;)
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Rudolf Ladyzhenskii
|Sent: Friday, July 15, 2005 11:05 PM
|To: Asterisk Users Mailing List - Non
, or a point in the right direction for
the right documentation would be appreciated.
P.S. This is CVS-HEAD Zaptel on a P3 550, Host bridge: Intel Corp.
440BX/ZX/DX - 82443BX/ZX/DX Host bridge (rev 3)
Chris Coulthurst
[EMAIL PROTECTED]
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can't remember if this was happening before Speex or not. Anyone have any
similar happenings?
Chris Coulthurst
[EMAIL PROTECTED]
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Not sure why I see *97 and *98 here, but I would check your dtmfmode= line
in sip.conf. Often times, using rfc2833 works when inband or sip-info
doesn't.
See http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+dtmfmode
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message
I claim to be NO expert, but is there a chance that the 'ztdummy' driver is
also being loaded? I'm thinking it might cause a timing conflict of some
kind...I may be way off here, but I'd still check for something as simple as
that...
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Martin Czarnowski
|Sent: Thursday, June 30, 2005 12:58 AM
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] Resolving groupcalls
|
|
|Hi,
|
|I'm
level 1: accountcode=019284718233 --account code
unique to the user
level 1: uniqueid=1120122635.400
Anyway, maybe something like that...
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf
have to change back over to PSTN lines temporarily, since
I can't rely on service from Teliax. I hope any/all of you that use their
service have better luck than I have with them.
Chris Coulthurst
[EMAIL PROTECTED]
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Try voip-co2.teliax.com to register with. And read my other letter I
suppose. This domain is apparently working as of 4:30, but have had the
same problem since 1:30 AM PDT.
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED
Does anyone have anything +/- to say about TeleSIP? They appear to have
local DIDs where I live and all comments on the wiki indicate they are
reputable..
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Rich
Yes I was just reading that TeleSIP and Telasip are often mistaken, and was
just editing my dialplan for my mistakes!
When you meen porting numbers, I assume you are talking about LNP? If so,
not a problem for me anyway.
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From
t SIP port of (usually) 5060 and the right
address, as well as having a '1' in the REGISTER box. See if that is all
correct.
Without more config information, its just a shot in the
dark.
Chris
Coulthurst
[EMAIL PROTECTED]
-Original Message-From:
[EMAIL PROTECTED]
[mai
love some real-dialplan working examples...
Desparately yours,
Chris Coulthurst
[EMAIL PROTECTED]
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Which software pack to you have for the IP600? Sip.ld, bootrom, etc...
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of Johann
|Sent: Tuesday, June 21, 2005 11:54 AM
|To: Asterisk Users Mailing List - Non
Check out DISA.
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Oswaldo Arratia
|Sent: Friday, June 17, 2005 7:51 AM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: [Asterisk-Users] 2nd
than an hour. Sounds to me like that problem is theirs, this would
help prove it.
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Mark Johnson
|Sent: Sunday, June 12, 2005 7:53 AM
|To: asterisk-users
# or the pilot
# of the DIDs, and helped me right from the switchroom!
If you are lucky enough to get a number like this, its GOLD.
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Mark Johnson
|Sent: Sunday, June 12
on channel
2
Is
that what you wanted?
Chris
Coulthurst
[EMAIL PROTECTED]
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Todd A.
RikerSent: Sunday, June 12, 2005 1:41 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] how
that helps...
Chris
Coulthurst
[EMAIL PROTECTED]
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Todd A.
RikerSent: Sunday, June 12, 2005 2:17 PMTo: 'Asterisk
Users Mailing List - Non-Commercial Discussion'Subject: RE:
[Asterisk-
after 60 seconds
Chris Coulthurst
[EMAIL PROTECTED]
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, if Polycom would put in a little backlight, and make a matching
SIP-enabled DSS console with REAL LEDs, I'd run with them and never look
back!
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Matthew
It sounds like there are quite a few people willing to aid in
bandwidth for voip-info. I was just wondering if it wouldn't make sense to
mirror the site across several locations with a round-robin DNS for a little
bit of load balancing? Any thoughts?
Chris Coulthurst
[EMAIL PROTECTED
I swear I read somewhere on one of the MANY pages that there is a script out
there that can read the extensions.conf file and create the MySQL DB records
on the fly. Anyone know where I look for such a thing?
Sure speeds up migration!
Chris Coulthurst
[EMAIL PROTECTED
Destroying call '[EMAIL PROTECTED]'
morse*CLI sip no debug
SIP Debugging Disabled
Chris Coulthurst
[EMAIL PROTECTED]
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, the program /usr/bin/head DOES exist.
Hey, if it works, post it!
Chris Coulthurst
[EMAIL PROTECTED]
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Sturtevant
Sent: Thursday, June 09, 2005 1:25
PM
To:
asterisk-users@lists.digium.com
Subject
Anyone else unable to get to www.voip-info.org? Site is returning
'connection refused' here.
Chris Coulthurst
[EMAIL PROTECTED]
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if voip-info would let anyone 'snake' the site for backup? Who
is the authority for that site?
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Neal Walton
|Sent: Thursday, June 09, 2005 3:17 PM
forward to the rebirth...
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Peter A. Solomon
|Sent: Thursday, June 09, 2005 9:20 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion
directories and the like.
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Kris Boutilier
|Sent: Thursday, June 09, 2005 9:40 PM
|To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Destroying call '[EMAIL PROTECTED]'
morse*CLI sip no debug
SIP Debugging Disabled
Chris Coulthurst
[EMAIL PROTECTED]
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with something like this?
Chris Coulthurst
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http://www.voip-info.org/wiki-Asterisk+h+extension
This might help
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Eric Smith - Fruitcom
|Sent: Tuesday, June 07, 2005 1:02 AM
|To: Asterisk Users
Does anyone know if there is any way to make the 'services' key do
anything?
How about a way to remap it?
Chris Coulthurst
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I just assigned one yesterday in a 10.X.X.X network with a netmast of
255.0.0.0, and had no problems..FYI.
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Charlie Watts
|Sent: Tuesday, June 07, 2005
to imagine that Polycom would add this button on the
500, an obviously different phone layout, unless they plan to add it as
a feature later.
I wish I knew how to program my own sip.ld files. I could make this
phone 'asterisk-specific'!! :)
Chris Coulthurst
[EMAIL PROTECTED]
|-Original
no need for account codes.
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Jodie Crouch
|Sent: Saturday, June 04, 2005 10:39 PM
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users
, monitor the channel until it becomes available, then
immediately ring back your phone to initiate the call.
Ok, I redundantly described what it does. Anyone have a way to make it
happen?
Thanks much...
Chris Coulthurst
[EMAIL PROTECTED
for me!
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
|Sent: Friday, June 03, 2005 8:16 PM
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] Caller ID Routing using
...just like the telco can do.
...but I do appreciate the response nonetheless!
Chris Coulthurst
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|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Jay Milk
|Sent: Saturday, June 04, 2005 10:28 AM
|To: 'Asterisk Users
...just like the telco can do.
...but I do appreciate the response nonetheless!
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Jay Milk
|Sent: Saturday, June 04, 2005 10:28 AM
|To: 'Asterisk Users
Sip/203s become Zap/2r2s.
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Steve
|Sent: Saturday, June 04, 2005 5:09 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re
and IAX2 extensions?
Can you concatenate extensions together?
(i.e. exten = 200,hint,SIP/101SIP/202Zap/4)
And the big one is.
...how does this work on a Polycom 500?
Chris Coulthurst
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Asterisk
,Hangup()
exten = 911,203,ChanIsAvail(Zap/5)
exten = 911,204,Dial(Zap/5/911)
exten = 911,205,Hangup()
exten = 911,304,SoftHangup(Zap/5-1)
exten = 911,305,Wait(2)
exten = 911,306,Goto(204)
Did I get the Priority + 101 idea right here?
Chris Coulthurst
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is
sound enough...
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Rich Adamson
|Sent: Friday, June 03, 2005 5:50 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk
to the list, does anyone recognize this
behavior, and have a workaround?
Chris Coulthurst
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There is no extension listed in extensions.conf. Maybe I'm just not
understanding this 'built-in' function like I should. The wiki page
isn't all that detailed, so maybe its me?
Chris Coulthurst
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|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL
to do it, I can smile again :)
Chris Coulthurst
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|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Matthew Marlowe
|Sent: Friday, June 03, 2005 1:16 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
Teliax is down, and can't even get to their www.teliax.com website.
Anyone else having problems?
Chris Coulthurst
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Might be a clock off somewhere, you actually responded to this within 10
minutes of me sending to the forum... and teliax is back up again
anyway..mustve been a network glitch down there in Colorado.
Chris Coulthurst
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|-Original Message-
|From: [EMAIL PROTECTED
and tries again (thus reentering a 7 second timeout if teliax is
still down).
Any suggestions? Dialplan examples?
Thanks for the help..
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Chris
the moment a
call went off hook, and having Answer in the extensions.conf
contexts made it all go away.
Am I under-thinking the use of Answer()?
Chris Coulthurst
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for jitter, packet loss, etc, and still be able to determine what the potential
culprit is for the problem?
Thanks again,
Chris Coulthurst
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Has anyone seen a situation where, upon connecting two asterisk servers
together with IAX registration, outgoing/incoming calls that route through
both servers are choppy and jittery? I don't have this problem when I call
out to teliax (my ITSP) directly, but if I try to make the call through the
I am sure this has been addressed somewhere else, but I
havent found it
Is there a way to make multiple extensions have their MWI
light flash, all for the same common voicemailbox? And to make it even
trickier, what if its a mix of SIP and ZAP channels?
Id like to be at my
.
Sip version 1.5.2.0054 and using the old bootrom
Chris Coulthurst
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