was a bit better...everything seemed to be where it should
be. At least now I`ll be able to appreciate the Polycom 501 and judge it in
a functional state.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: July 24, 2006 5:52
Thanks Richard, somebody pointed me to the CDP setting, and to a bad hub
(which, even if I said it wasnt the case, seemed to have been my second
problem)
Everything works now, thank you to this group for giving me such quick help.
Mike
-Original Message-
From: [EMAIL PROTECTED
.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of shadowym
Sent: July 25, 2006 2:48 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Just bought a Polycom 501 - Ifeellike
myGXP-2000 was better...
I don't
the transition would be
much easier on the GXP-2000 (not taking into consideration the provisioning
which does look a lot easier to manage on the Polycom) because it acts 90%
like a typical phone.
And really, the lack of a backlight is a shame on the Polycom.
Mike
-Original Message-
From: [EMAIL
username and reg.password values and then passing off as, let's say,
his boss and wreaking havoc on the business' reputation by getting calls which
weren't meant for him...?
Mike
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I would be interested in knowing if this can be changed. It can`t have been
designed like this with no option to change it.
So I`m throwing this question back in the arena: Can you get the Polycom 501
to ring when a calls comes in and the user is already on a call?
Mike
-Original Message
Thanks, I know your right (I tried the second option). Problem is that the
phone doesn`t RING. The light flashes, the as far as an audio ring goes,
it`s completely silent.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Vincent
(C)
Sent
Thanks. That's an ok solution. I just thought I could make the Polycom
ring normally (or even better, with decreased volume) when a new call comes
in.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones
Sent: August 3, 2006 11:00 AM
Hi,
I`m trying to record a conference, and I`ve been using .wav format to get
decent audio quality. The conference goes fine, but when I listen to the
recording after, I hear horrible echo (which I couldnt hear on the conf
call itself).
Whats causing this?
Mike
for one phone.
Is this the
case?
Mike
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11", "") in new
stack
I am running
1.2.4. Not even sure what the warning means (WARNING[24892]:
app_addon_sql_mysql.c:115 find_identifier: Identifier 2, identifier_type 2 not
found in identifier list).
Any help is
appreciated.
Mike
___
Hi,
Where should I go
toget the Polycom`s latest official (non-beta) version? I am
registered on the Polycomcustomer website but that doesn't seem
accessible.
Regards,
Mike
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asterisk
virtual users (i.e. users not necessarily Linux
users)
3) Can I make this
work with a self-signed certificate? If so, anything in particular that I need
to know?
Regards,
Mike
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d the same problem and found what the problem was?
Mike
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Im replying to me own message to avoid having somebody
write a lengthy response for nothingturns out my problem is Pure-FTP that's
for some reason not letting the file go through properly.
I'm therefore taking my discussion over to some other
mailing list. Sorry about that.
Mike
know I am not wasting my time.
PS: If there is a
better FTP server suggestion Ill take it, but one of my "must-haves" is easy of
use and virtual users functionality (with different chroot
folders).
Mike
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nce.
What can be the
problem? I imagine the NAT isnt the problem, or there would be no audio at
all. My Asterisk is running 1.2.4, and my Polycom phones at running
bootrom 3.2.2 and SIP 2.0.1 (fairly recent).
Mike
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at the beginning of my calls. My Asterisk
server is not behind a NAT, so in theory it should work flawlessly. Also,
the latency between my LAN and my Asterisk server is about 10ms, very
stable.
I am trying to figure it out with Ethereal (first thing I
did) but I'm not sure what to look for.
Mike
pick it up.
Short of that, can
somebody point me to the newest firmware (2.0.2) to see if thatwould
help?
Mike
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Any hints on downgrading? I placed the old SIP 1.6.7
on the right folder, but my phone wont pick it up and install it. It must
be thinking "this is an old version, ignore" or
something
I`ve never downgraded a phone, I tend to like upgrading
more :-)
Mike
Fr
remotely, and I did for a few phones, but my
paranoid self would like to double check and see if the sip.ld 1.6.7
re-installed ok by checking the current version. Is that even
possible?
Mike
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rick
SmithSent: November 7
of the phone is being used.
Mike, happy to contribute answers instead of questions for
once.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rick
SmithSent: November 7, 2006 8:44 PMTo: 'Asterisk Users
Mailing List - Non-Commercial Discussion'Subject: RE:
[asterisk
|
My question is, if the caller spends 28 seconds listening to options before
dialing an extension, and the call last 89 seconds...Should the first leg
have a billsec of 89+28=117sec and the second 89 seconds?
Mike
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I`m impressed. Thanks for the reply, I'll try that!
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Benny Amorsen
Sent: November 21, 2006 4:17 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Call limits and VoIP
Just curious if anyone knows of any hacks to enable announce entry/exit
in MeetMe conferences with SIP (as opposed to ZAP) channels since the |i
option will not work with SIP.
Thanks,
Mike
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number rather than the
userr's default callerid?
Is this correct?
Mike
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callerid?
Is this correct?
Mike,
Exactamundo.
Doug.
Ok.
How about:
;outgoing context for company A
[companyA]
;Various include statements
include = foo
.
.
.
;Handle calls from A - B
;Here will match company B numbers
exten = , 1, Set(CALLERID=CompanyAMain)
exten
(${EXTEN}
You can do the inverse for companyB, or you could l have a single
macro that deals with calls to/from each company and decides what do
to based on the callerid making the call.
Mike.
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C F wrote:
You Dont Have A Priority 1 And You Have Priority 2 Twice
Also, the timeout param is number of seconds, not number of rings.
On 12/24/06, Charlie Grosvenor [EMAIL PROTECTED] wrote:
I am using Voip Talk and have my extensions.conf set up to make outgoing
calls:
exten =
missed.
Does anyone have any idea what may cause this?
Thanks,
Mike.
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Vulpes Velox wrote:
Jan 3 08:05:03 NOTICE[66269]: app_dial.c:1056 dial_exec_full: Unable
to create channel of type 'SIP' (cause 3 - No route to destination)
I end up getting this when I call from 2000 to 2001.
2000, 2002, and 2001 all exist in sip.conf and I connect using them.
I have all
any real QoS functionality.
Mike
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Mike wrote:
Hi,
I'm looking for opinions on the best value router to use for home
offices. It should work for a scenario in which there are 3 computers
and 2 SIP phones, handling QoS so that the phones always have higher
priority traffic than the PCs. (and not rely on the phones to do
Yes, I knew that but it's nice that you mention it. I want QoS specifically
to prevent large downloads/kids using BitTorrent in their bedrooms locally
from interfering with the calls.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent
be purchased and
installed easily (Linksys type of product)
Mike
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: Thursday, January 04, 2007 15:56
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Best inexpensive home
Al Bochter wrote:
What about the free open source G729
To use a g729 codec you must pay a license fee to the patent holder. It
is immaterial as to whether the implementation is open/closed source.
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Al Bochter wrote:
Mike,
So tell me what this FREE open source G729 is
I am told that you can use these Codecs with your Asterisk !
http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
You can do it Freely !!
Please read the entire page. From the link you sent:
Why NOT G.729
Al Bochter wrote:
Mike
I understand that.
but it states on there site and note the key words may need
What I want to know is if you buy 10 licenses from digum can use the
Open Souce code?
That is not what you said or asked. You were asserting that a free as
in beer solution existed
Mark Greene wrote:
I have googled and I do not understand how the pager field is what is
causing the problem.
Could you explain?
Think of it as a CSV file. The ,, entry for pager is just a placeholder
saying that for pager there is nothing. Omitting means that the next
field will be
. Which is NOT what I
want.
Is there a standard way to say hid my number?
Mike
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is a SIP connection, not a PRI, is there anyway to do something
like that with SIP? Would that be provider-specific?
Mike
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, than the
queue can try him again.
Is this (or something similar) possible?
Mike
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Hi,
Is there a way to have a Do-While sort of loop, as opposed to a simple
While?
I have a condition that the loop depends on even for the first iteration, as
it often happens in life.
Regards,
Mike
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Hi,
The Asterisk Wiki (page:
http://www.voip-info.org/wiki/view/Asterisk+func+cdr) mentions I can set any
custom CDR field I want. Here is the example it gives:
; Update our accountcode field and then save some random music facts too
exten = s,1,Set(CDR(accountcode)=8675309)
exten =
] Setting custom field in CDR
Mike wrote:
Hi,
The Asterisk Wiki (page:
http://www.voip-info.org/wiki/view/Asterisk+func+cdr)
mentions I can
set any custom CDR field I want. Here is the example it gives:
; Update our accountcode field and then save some random
music facts
too
Hi,
I've had a functioning Asterisk system (1.2.18), which I haven't
reconfigured in any way, that is just now refusing to forward calls. I
only have Polycom phones. When I use the phone's forward feature
(forwarding the phone with extension 204 to extension 206, which used to
work as
Hi Noah,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Noah Miller
Sent: Thursday, December 13, 2007 21:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.2.18 and Polycom
phones
Noah,
Turns out I found the problem, BUT I don't understand it exactly. My phones
are on a LAN, and the PBX is on a different IP (Hosted PBX basically).
I had to open out port 5060 on my router (where the phones are). The thing
is, conversations flowed perfectly (with multiple phones at a
The only reason I am not upgrading to 1.4 is because out-of-the-tar it just
won't build on my Fedora Core 4 machine.
http://bugs.digium.com/view.php?id=9643
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Johansson Olle E
Sent: Saturday
-Lite (which has only 2 lines, not enough). The commercial
version of X-Lite looks nice, but doesn't support provisioning. At the
moment, it's my fallback plan.
Regards,
Mike
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and needs to be hungup */
if (p-mwimonitor_rpas) {
ast_hangup(chan);
return NULL;
}
}
I have set usecallerid=no on both interfaces and globally but I still
cannot get it to stop.
I have failed to turn anything up on Google regarding this.
Does anyone have any suggestions please?
Mike
=no?
Mike.
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that what happens is that the FXO line rings, so Asterisk rings
the FXS phone as per the extensions.conf, this creates a MWI event which
goes to the voicemail system, which then passes a MWI event to the SIP
phone (as per sip.conf)? Or I could just be talk rubbish!
Mike.
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Description
On Wed, Aug 05, 2009 at 02:13:01PM +0800, D Tucny wrote:
The problem is that your mailbox line was below channel=1, as such, it
applied to the next channel, channel=3 not channel=1...
d
Nice one. Thanks for spotting that.
Mike.
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(no calls, lots of
registrations of course, but nothing worth 2Mbits/s)
Regards,
Mike
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Sorry, that is running 1.4.26.1.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, August 13, 2009 23:22
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Stale auth
appreciate any tips.
Mike
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Hi,
yes I did, I did have errors at first but that hurdle has been cleared.
Thanks for the try :-)
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Thursday, September 24, 2009
I've tried turning logging way up for the relevant portions of the sip
application, but no telnet. Not sure how I would go about this to get more
info that what I already have. The phone is giving me a response, it's just
that the response
is push message cannot be displayed
Mike
I am looking to configure the asterisk voicemail system to stop asking for
the folder (work, personal, etc) in which to save messages when I do
save them.
Is there any configuration to do this?
Mike
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)} is NULL fo the rest of the
dialplan.
My dialplan logic depends heavily on knowing the accountcode.
Any idea what I am missing? Things work well with a normal non-blind
transfer.
Mike
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Just to follow-up: I know there is a variable ${BLINDTRANSFER}. I`d like to
get the CDR out of that channel, but can`t find a way how.
The CHANNEL func gets the info of the current CHANNEL, is there a function
to get variables from another CHANNEL, references by ${BLINDTRANSFER}?
Mike
?
Thanks,
Mike.
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plugged the phone directly into the phone line and the dialer
works just fine. Plug it into the TDM400 and it doesn't work, although
I can tap the number usin the hook.
Mike.
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On Wed, Nov 25, 2009 at 12:26:10PM +0200, Tzafrir Cohen wrote:
On Tue, Nov 24, 2009 at 11:03:16PM +, Mike wrote:
Folks,
I've got one of those GPO 1950's rotary dial phones that I'm trying to
get working in the UK. I've got pretty much everything working with my
TDM400, the phone
?
Mike
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Hi,
Running 1.4.26.1 here. I have installed TE420B card in my server, and
followed the appropriate steps (as far as I know to configure it). This
TE420B is connected to a CLEC (T1s), so I am using pri_cpe as singalling
type.
When I dial out, I get this message:
Dec 4 11:37:31]
Forget it, found my issues. I have been looking for hours, but as soon as I
write this I find it. dahdi-channels.conf wasn't included in
chan_dahdi.conf.
That being said, I have other issues now, but at least that one is fixed.
Regards,
Mike
From: asterisk-users-boun
channels correctly,
but not my outbound. My outbound never show up, even during a conversation.
Thanks for helping me figure this out.
Mike
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(although I could
work it out from the former if it was available)
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, December 04, 2009 14:22
To: 'Asterisk Users Mailing List - Non-Commercial
Thank you, at least I am getting the same thing.
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: Friday, December 04, 2009 16:37
To: Asterisk Users Mailing List - Non-Commercial
? Ideally, have two values,
one for each T1.
dahdi show channels doesn't show outgoing calls. Is there another command I
am not aware of?
Mike
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Thanks Tim and Danny. It seems a more direct way should be there, but that`ll
work.
Regards,
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Tuesday, December 08, 2009 16:45
To: Asterisk Users Mailing
, but not the whole server!
Mike
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by it being indirect.
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Carlos Chavez
Sent: Tuesday, December 08, 2009 19:25
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
I have Asterisj 1.4.26.`1, and Dahdi 2.2.0.2.
I restarted for no good reason (I was playing around), but it did worry me
that if Dahdi crashed while Asterisk was running that not only Dahdi and
Asterisk would crash, but the whole machine too.
Mike
-Original Message-
From: asterisk
I am new to the list and wanted to get the professionals here input on
Switchvox 305 Appliance ?
List price is 4k, ouch! Is there a better cost-effective way ?
Also feedback (neg/pos) about this appliance.
-mike
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Hi Hin,
thanks for the reply back.
Is there a ready-to-go appliance running Elastix? or what type of
hardware do I need to have features as switchvox 305 (for example: it
can handle upto 150 users)
-mike
On Thu, Dec 10, 2009 at 5:03 PM, hin lee hi...@yahoo.com wrote:
I had once considered
Hi,
I know having Asterisk aware of Polycom Do No Disturb state wasn't working
before (1.4), but is this working in any recent version? Is there any
custom way of doing this?
Regards,
Mike
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.
Where could be the difference? Both are using the same context to dial out.
Mike
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be done as long as the
feature makes it into trunk. Heck, I'll give 200$ for someone just to tell me
how to configure it properly if it's a matter of just missing a config line.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
Hey Jimmy,
3.2.0 is what I am using.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Godbout
Sent: Thursday, February 04, 2010 22:21
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
it into trunk. Heck, I'll give 200$ for someone just to
tell me how to configure it properly if it's a matter of just missing a
config line.
Mike
Which polycom phones are you using and what SIP firmware are you using?
I am using 3.2.0, with a variety of phones (321, 331, 430, 450, 550
I may be late to this thread, but my own restarted every 3-5 days until I
upgraded to 1.4.29 (I skipped 1.4.28).
It`s been running for 8 days now, which isn't long enough for me to declare
whatever-it-is fixed, but enough to say it's at least better with 1.4.29
stability wise.
Mike
?
Mike
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asterisk-users mailing
Hi Bob,
Thanks for replying. I've thought of doing that, but softkeys are limited
and for a phone with many call appearances (4-5) that would be using many of
the softkeys.
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun
Hi All,
Anyone one info of where I can get a 'free' DID number ?
I have setup my asterisk box (home) and want to learn more but I need a #.
thanks in advance,
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Ok, I see there's alot out there of voip providers.
Curious what to watch out for ? charges and fee's, etc ?
If anyone has feedback as to a GOOD voip provider experience (one that
gave FREE DID) Please share.
Again, I am doing this to learn about asterisk, I'm currently testing
it at home.
My bad, I'm in Los angeles california usa
On Thu, Mar 18, 2010 at 1:06 AM, SIP s...@arcdiv.com wrote:
What country are you in? Makes somewhat of a difference.
N.
On 3/17/2010 8:49 PM, Mike wrote:
Ok, I see there's alot out there of voip providers.
Curious what to watch out for ? charges
system (And more to the
point, allowing easy outgoing routing based on which NIC was used).
Am I correct?
Bonus question if I am indeed correct: how stable is 1.6 right now, compared
to the latest 1.4 (1.4.31)?
Mike
Hi,
This is a bit off-topic, but still related to telephony. Is there a
barebones TAPI driver that exists that would allow me to call up a command
line with, as parameter, the number to dial.
For exemple, Outlook integrates with TAPI, so that TAPI driver would allow
me to call my own app
Thanks, will take a look. Althought none of those things seem to allow me
to call up my own handler for calls, does it? Or am I misreading?
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, May
?
Regards,
Mike
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I should have mentionned this is already done. I can see that is a SIP
response when trying 192.168.1.3, but the phones fails to register. I
suspect a NAT/firewall issue because packets are leaving for 192.168.1.3,
but coming back from 192.168.1.2.
Mike
From: asterisk-users-boun
respond from the IP
address used for registration
On Thu, 27 May 2010, Mike wrote:
Hi,
I have a test server with 2 NICs, each with it own IP address. Let`s say
192.168.1.2 and 192.168.1.3. I would like some phones to register by
using
192.168.1.2 and some by using 192.168.1.3
Hi Andrew,
Thanks, I'll look this up. The term packet mangling wasn't used in my many
google searches.
Mike
On 28/05/2010, Mike l...@virtutel.ca wrote:
That was a simplified example. I actually have two links from different
ISPs, totally different networks. Those on provider A should
See bindaddr here:
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
That should do exactly what you want.
Regards,
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of CDR
Sent: Sunday, May 30, 2010 10:06
it is, at least on 1.4. I read somewhere (can`t
find the page) that 1.6 works differently.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Monday, May 31, 2010 9:55
To: 'Asterisk Users Mailing List - Non-Commercial Discussion
policy (if it came in
on NIC 1, send it back the same way even if it`s a less direct route).
Somebody told me to lookup Packet Mangling, which I have yet to do. Will
probably write a wiki page about this if that works, because I don`t seem to
be the only one with this need.
Regards,
Mike
reflected
when a new call comes in, or when I reload the dialplan.
What do I need to do for the changes to be shown in the CLI, short of
restarting Asterisk?
Regards,
Mike
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