On Jul 11, 2006, at 11:10 AM, Rick Smith wrote:
teliax had a 2.5 hour outage today. I wouldn't call that short.
They notified about the situation when it happened and explained it
when it ended, so that makes them look extremely professional compared
to other vendors in this department.
You said there are 3 asterisk servers in your system. Are you using
ARA? Is it a multi-asterisk configuration?
Do you mean to tell the configuration of your environment?
On 7/12/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Assuming a standard usage rate of 10-15%, that's 2,400 concurrent
For redundancy on the PRI side, we plug our PRIs into Redfone
Networks' foneBRIDGE boxes. They've worked quite well for us. As
others have stated, use short registration periods combined with some
HA software to handle your SIP redundancy. You might also look into
load-balancing SER proxies.
On Jul 11, 2006, at 2:27 AM, Olivier Saulnier wrote:
Hello,
Sometimes, when i call an outside people, he said me that the
communication is bad:
The voice is low, far, bad poor quality.
How can i know where is the problem, which tests can i make?
Make sure the string is tight between the
On Jul 11, 2006, at 5:14 PM, Ronald Wiplinger wrote:
That is easy to calculate: 3,000 US$ times your zip code times the
phone number you are calling times 2.9cents/5 seconds divided by the
Social Security number of the called party ... Or how does NuFone
calculate that?
But hey, just look at
On Jul 11, 2006, at 5:44 PM, Kevin P. Fleming wrote:
- Stagg Shelton [EMAIL PROTECTED] wrote:
I did ultimately force asterisk to the point where it will not accept
or
send rfc2833. I did this by modifying chan_sip.c in the function
Asterisk should not be sending an SDP with RFC-2833 in
Thank you very much!!!
Giorgio Incantalupo
C F wrote:
Yes you can do that just change the application map to a Goto command
that goes to an exten in the dialplan that does it all for you.
On 7/11/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi C F,
I managed to send the flash
On Jul 11, 2006, at 7:39 PM, Al Lougher wrote:
I have 969mb total mem with 780mb allocated as swap
??
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I have 2 context and each have its trunk but when I active
call forward the off-net international number goes over the 1st trunk
only ,
How can I solve this problem,I want to let each context goes
over its specified trunk
*
No employee or
Hello list
We are having some strange problems.
When we setup trunking between two of our servers, the connection only uses
trunking one way. Ex:
Data From callingserver to receivingserver uses trunking
Data from receivingserver to callingserver does not use trunking.
I discovered this
That worked great, thanks for your help. It has now brought me to another
problem if you/anybody can help.
When a sip device tries to register with asterisk the CLI comes up with:
Registration from 'sip:[EMAIL PROTECTED]' failed for '192.1.3.103' -
Username/auth name mismatch
The 3002 is
I have the following context in the dial plan.
in extension.conf
[default]
1) context1
2) context2
3) context1,1,Macro(a)
4) context2,1,NoOp
5) context2,2,NoOp
[macro-a]
6) exten = s,1, NoOp
7) exten = s,2, MacroExit
As I expect the route of a call is 3,6,7,4,5. However, when I execute
the
Hi Dennis,Thank you for the information. We haven't had that problem so far, in the tests that we have been making (we are not yet in production, but will likely be within a short time frame in several countries), so unfortunately I cannot help you in that specific problem. I hope you can
Hi Guys
I have a client call center that has after hours agents. Once the call
center closes they forward calls to the night affiliates. These nite
operators are not constant and tend to swop with each other and then let
the person in charge know who is on when. I have the mammoth (Mannie)
Since you make call forward the forwarded
number will be dual by default from the context from-internal which
is linked to a trunk
How can I let it find the context ? automatically
$context ?
Please help
*
No employee or agent
I have noticed a couple of issues, unless I'm doing something wrong?
I pulled with svn the
svn.digium.com/svn/asterisk/team/bweschke/polycom_acd_functions/ which got
release 37416
This complies fine, in particular the meetme app.
If I setup a sip device in the sip.conf with a username and
Hi all,
I'm trying to setup a menu in a queue, something like All our agents
are busy, press 1 to leave a voicemail, press 2 for another
department etc.
Anyway, the only thing I found is this on the wiki:
Menu for the user
You can define a menu for the user, while waiting. For this menu,
Hi all, any one know about :Midnight Naughties.
The Asterisk box seems to log all agents off Queues at exactly midnight each
night, and nothing in cron jobs.
Thanks
James
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Hi guys,
I would like to know if the Ambient and Intel FA82537EP are X100P/X101P
compatibles.
Thanks
Cheers
Alex
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Hi,
In the queue.conf when you define the queue:
[myqueue]
context = thecontextforusers
...
and in your extensions.conf
[thecontextforusers]
exten = 1,1,NoOp(1 WAS PRESSED)
exten = 2,1,NoOp(2 WAS PRESSED)
If you need more help, just ask ! ;)
Attilla De Groot a écrit :
Hi all,
I'm
Hi Asterisk Users,
Sorry if this is off Topic for this list.
But does anyone have a full XML config file for the SPA-3000, the PAP2 and
the SPA-941.
Or alternatively a way to convert the field names on the web pages to the
corresponding XML filed names.
Thanks
/S
On Jul 12, 2006, at 1:01 PM, Tristan wrote:
Hi,
In the queue.conf when you define the queue:
[myqueue]
context = thecontextforusers
...
and in your extensions.conf
[thecontextforusers]
exten = 1,1,NoOp(1 WAS PRESSED)
exten = 2,1,NoOp(2 WAS PRESSED)
If you need more help, just ask ! ;)
Since you make call forward to an
extension l by default it will attach your DIAL(local/[EMAIL PROTECTED]) from
the context from-internal which is linked to a trunk ,
The script is at
/var/lib/asterisk/agi-bin/dialparties.agi
I
$dialstring =
'Local/'.$extnum.'@from-internal';
Sorry if this is off Topic for this list.
But does anyone have a full XML config file for the SPA-3000, the PAP2
and the SPA-941.
Or alternatively a way to convert the field names on the web pages to
the corresponding XML filed names.
The spa provisioning guide outlines the xml-like syntax.
Question rephrase:
If I have a sip to voicemail call that needs G.729 transcoding, can it use the
Digium hardware transcoder or would I still need a
software transcoding license for this?
--
--
Steven
http://www.glimasoutheast.org
C F [EMAIL PROTECTED] wrote in message news:[EMAIL
We still have a lot of users on a legacy PBX, so the Directory app is
not sufficient.
We also have users with mobile phones.
Has anyone made an LDAP lookup that will pull this info from MS Active
Directory?
My thinking is to add this function to my main IVR.
As long as my AD is accurate, it
On 7/12/06, Henry J. Cobb [EMAIL PROTECTED] wrote:
When I've tried it, app_conference always crashed within the hour.
that's strange. we've use app_conference for months and months on end without incident.are you building app_conference from the main svn trunk? or are you using matt's
Eric \ManxPower\ Wieling [EMAIL PROTECTED] writes:
Wolfgang Zweimueller wrote:
Hi all,
when I receive incoming SIP calls on my Asterisk (1.2.9.1) where the
caller has a username in it's From-Address which also exists in my
sip.conf then my system answers with 407 Proxy Authentication
[EMAIL PROTECTED] wrote:
On 7/12/06, Henry J. Cobb [EMAIL PROTECTED] wrote:
When I've tried it, app_conference always crashed within the hour.
that's strange. we've use app_conference for months and months on end
without incident.
are you building app_conference from the main svn trunk? or
It really depends on the application. app_conference does wonderfully
for long conferences without a lot of entry/exit and no playing of
audio files.
The issues with the double-free crashes that we've had all seem to be
caused by playing of audio files(like the entry/exit sounds or the
DTMF
Erick Perez wrote:
There is an old, very old document that I found somewhere that this
PoE switch was designed for NBX phones at that time.
Does anybody in this list is using this switch with non-3com NBX PoE
phones?
just check the voltage specs. I think you will fry anything other than
an
Hello
When I try to use giztmo whit this configuration i'm unable to receive calls
register = 1747601:[EMAIL PROTECTED]
externhost= host.homeip.net
port=5060
defaultexpirey=3600
localnet= 192.168.0.0/255.255.255.0
But whit this configuration y can receive calls
register =
Sounds like class action lawsuit time.
Michael Workman wrote:
So Nufone Screwed ya
I feel Sorry... W Take your Lumps... Cut Your Losses and Get on with
Life
Your not the only one Nufone Screwed They Screwed me Out of $3,000.00
NEXT TIME BEFORE YOU GET SCREWED BUY SOME KY
interesting. i didn't realize the problem seems to specifically be the sound playback via the manager interface.a couple weeks ago, my asterisk on my dev box crashed, i did some preliminary investigation, but since we hadn't had any problems in production or qa, i chalked it up to me messing up my
Hi all,
I'm having a problem with receiving calls from a VOIP provider who is
providing inbound PSTN termination using IAX2 to my [EMAIL PROTECTED] 2.6 box.
The box is a mini-ITX based P5000 system running off a 2.5in drive with
a digium TD400P (3 FXO). But this problem does not relate to
Since I make call forward to an
extension l by default it will attach your DIAL(local/[EMAIL PROTECTED])
from the context from-internal which is linked to a trunk ,
The script is located at
/var/lib/asterisk/agi-bin/dialparties.agi
I
$dialstring =
That's the fourth time you've asked the same question in the space of
a few hours - please have a little more patience and wait for someone
to answer.
On 12/07/06, Khaled Chehab [EMAIL PROTECTED] wrote:
Since I make call forward to an extension l by default it will attach your
Thanx for the tips guys. I need one more favour. can anybody tell me where to find help for writing AGI scripts in C language. I have read the pdf book called Asterisk TFOT, but it explains AGI scripting in languages other than C. I feel comfortable using C language, so i didnt understand the
Dan Brummer wrote:
This has worked. I downgraded from 1.2.9.1 to 1.2.7.1 and I'm not
having the warm transfer issue anymore. Does anyone know if this is a
known issue and is going to be fixed in upcoming release? Should I
possibly put in a bug request?
-Dan
I'm pretty sure I saw this in
Can you explain why this would be different?
On 7/12/06, Steven [EMAIL PROTECTED] wrote:
Question rephrase:
If I have a sip to voicemail call that needs G.729 transcoding, can it use the
Digium hardware transcoder or would I still need a
software transcoding license for this?
--
--
Steven
Hi,
is several 1000s of extensions in a context a problem?
Roger.
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Thanks All
First off I never mentioned Teliax (but yes correctly ASSUMED they are
my provider) and this is not a Teliax issue per se
My issue is more the fact that I have Qualify = yes in sip.conf but
repeatedly get REACHABLE and UNREACHABLE
as can be seen below. even when I set Qualify =
Hi, We could makecalls to USA using Teliax service upto 11, July 2006 with Asterisk. But, since 11, July 2006 evening, we are unable to make calls sometimes and could not connect to Teliax server sometimes. I have realized that Teliax server was down for few hours. Currently our Asterisk
On 12/07/06, Crazy Boy [EMAIL PROTECTED] wrote:
Hi,
We could make calls to USA using Teliax service upto 11, July 2006 with
Asterisk. But, since 11, July 2006 evening, we are unable to make calls
sometimes and could not connect to Teliax server sometimes. I have realized
that Teliax server was
several thousand extensions or several extensions called 1000 ?
- Original Message -
From: Roger Schreiter [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, July 12, 2006 9:15 AM
Subject: [asterisk-users] 1000s of extensions in one context?
Hi,
is several 1000s
do what we all do. get backup routes
- Original Message -
From: Barry Fawthrop [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, July 12, 2006 9:15 AM
Subject: Re: [asterisk-users] Provider UNREACHABLE
Hello,
I need to install Asterisk on a test machine that will soon become a
production environment.
Do you think that 1.2.9.1 is reliable? I read some posts that say it
isn't as good as the previous versions. Should I install 1.2.8 or
1.2.7.1?
Please give me an advice!
Thanks in advance,
--
Hi,
I want to implement a lookup for valid extensions
using agi.
Thus I want chan_zap to accept some digits,
then check via agi if the number is complete,
run waitexten if necessary and check again ...
Unfortunately waitexten only accepts one digit, regardless
how may key strokes I did on
(b) What is happening ?
If i unplug network cable from my ipphone asterisk will say
UNREACHABLE after few seconds.
Sometimes it occurs for my sip provider. I simply lose connection with
my phone/provider or
connection is so poor that transmission of voice will be very choppy, so
beeter not
Crazy Boy wrote:
Hi,
We could make calls to USA using Teliax service upto 11, July 2006 with
Asterisk. But, since 11, July 2006 evening, we are unable to make calls
sometimes and could not connect to Teliax server sometimes. I have
realized that Teliax server was down for few hours.
snip
Hi,Thank you for response. I sent an email to Teliax people also. I may get reply from Teliax within few hours. Please tell me the solution. Thank you.Regards,Chandra.Peter Bowyer [EMAIL PROTECTED] wrote: On 12/07/06, Crazy Boy <[EMAIL PROTECTED]> wrote: Hi, We could make calls to USA using Teliax
It's the internet...maybe for you the path to Teliax is kinda crappy?
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Barry
Fawthrop
Sent: Wednesday, July 12, 2006 9:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hello,
I need to install Asterisk on a test machine that will soon become a
production environment.
Do you think that 1.2.9.1 is reliable? I read some posts that say it
isn't as good as the previous versions. Should I install 1.2.8 or
1.2.7.1?
Please give me an advice!
Thanks in advance,
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andrea Spadaccini
Sent: Wednesday, July 12, 2006 9:31 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk version: 1.2.9.1 or older?
Hello,
I need to install
Hi All
I have an Asterisk server with two PRI's into a Samsung DCS500 which in
turn has 4 PRI's into the Telco. I have echo on some calls only over
500km away. I have set:
echocancel=yes
echotraining=800
echocancelwhenbridged=yes
rxgain=1
txgain=-1
Volumes on the ZAP channels are set
I just finished building a gtk gui that's geared toward call centers.
Check it out.
You're of course welcome to help me code more for it if you like it :)
amsuite.sourceforge.net
j
On Wed, 2006-07-12 at 11:39 +0200, Terry Wade wrote:
Hi Guys
I have a client call center that has after
I personally have had some issues with 1.2.9.1 in production and had to
revert to an older version.
We are using 1.2.6 which has proven to be pretty stable.
Others might have different experiences.
j
On Wed, 2006-07-12 at 15:31 +0200, Andrea Spadaccini wrote:
Hello,
I need to install
I have 2 SIP phones ext 401 and 402 and am using 1.2.9.1 (this seemed to
work in previous versions).
In my dialplan I have
exten = 55,1,Dial(SIP/402,,tTD(222))
When 401 comes offhook and dials 55 and I answer on 402 I do not hear
the DTFM of 222.
Am I doing something wrong?
when I change
Bell Canada is going
to give Canada Wide Calling free in Next few months and Skype is Giving out
North American Calls Free until it fixes the security hole in software/system
and VoIP Discount is giving out free North American Calling.
Anyone who is trying
to Sells / Provides VoIP is and
Andrea Spadaccini wrote:
Hello,
I need to install Asterisk on a test machine that will soon become a
production environment.
Do you think that 1.2.9.1 is reliable? I read some posts that say it
isn't as good as the previous versions. Should I install 1.2.8 or
1.2.7.1?
I've had no issues at
Bill's right. But, it happens to me too, ALL the time, w/Teliax.
I can't wait for their NYC node...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs
Sent: Wednesday, July 12, 2006 9:46 AM
To: Asterisk Users Mailing List - Non-Commercial
Michael,
Maybe I am not understanding your question, are you saying that when you
configure your phone with a static IP address, you cannot find the boot
server and when in DHCP you can? If you are having problems with the
phone having a static IP address, make sure it is getting the correct
Hi all I need a help with asterisk+fax - fax to email I am trying to setup fax to emailwith asterisk with nosuccess.I have asterisk 1.2.9.1 running on CentOS i have created extension 300 which should receive faxes.extensions.conf - exten =
Yes, thanks again for the suggestions. I wrote a few scripts for
different things that we needed in the office and by the time I got to
that one, I was tired and wasn't thinking straight anymore. I am
probably going to just set a dummy variable for now and have asterisk
update the global. Down
A comcast representative told me the other day they are planning on doubling
their
internet speed from 8Mb to 16Mb at the end of this year.
:-D
signature.asc
Description: OpenPGP digital signature
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Crazy Boy wrote:
Hi,
We could
makecalls to USA using Teliax service upto 11, July 2006 with
Asterisk. But, since 11, July 2006 evening, we are unable to make calls
sometimes and could not connect to Teliax server sometimes. I have
realized that Teliax server was down for few hours.
Hello,
My backtraces never actually mention play_sound, but the crashes only
happen right after app_conference attempts to play out DTMF tines with
the playing function.
Here's the backtrace for two of the crashes that we had with app_conference:
Kevin P. Fleming wrote:
Can we please keep the discussions about carriers, money, jobs, work, etc. off
of this list? This is not the place to discuss your experiences with _any_
company, it's a place to talk about Asterisk and using Asterisk.
Please move flamewars and similar discussions to
On Tue, 2006-07-11 at 14:39 +0100, Roshan Sembacuttiaratchy wrote:
On Tue, Jul 11, 2006 at 06:23:05AM -0700, Al Lougher scribbled:
Hi -
This is the first Linux server I have ever built with an
installation of [EMAIL PROTECTED] 2.7. For development I have been
running on
On 7/12/06, al gav [EMAIL PROTECTED] wrote:
Hi all
I need a help with asterisk+fax - fax to email
I am trying to setup fax to email with asterisk with no success.
I have asterisk 1.2.9.1 running on CentOS
i have created extension 300 which should receive faxes.
Which version of spandsp did
Asterisk
is trying to send an email to users when they receive a voicemail. Can this be
shut off? I have not entered any email addresses in voicemail.conf so it tries
to send to [EMAIL PROTECTED].
This of course gets rejected since the user does not exist and the root users
mailbox on
and cable vision now has 30/2 and they will have 50/50 real soon
- Original Message -
From: Derek Whitten [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, July 12, 2006 10:45 AM
Subject: [asterisk-users]
Hi all
I need a help with asterisk+fax - fax to email
I am trying to setup fax to email with asterisk with no success.
I have asterisk 1.2.9.1 running on CentOS
i have created extension 300 which should receive faxes.
extensions.conf
-
exten =
trixter aka Bret McDanel wrote:
On Tue, 2006-07-11 at 20:51 -0400, C F wrote:
While I don't disagree with you, look at what my point was, just
accusing them for such without any documentation doesn't make sens.
He only brought that up after people started questioning it. So I
dunno.
Matt Florell [EMAIL PROTECTED] wrote:
My backtraces never actually mention play_sound, but the crashes only
happen right after app_conference attempts to play out DTMF tines with
the playing function.
This is because Malloc isn't crashing when the mistake is made.
It crashes later because of
Dovid Bender wrote:
and cable vision now has 30/2 and they will have 50/50 real soon
- Original Message - From: Derek Whitten [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, July 12, 2006 10:45 AM
Subject:
On 7/12/06, al gav [EMAIL PROTECTED] wrote:
Hi all
I need a help with asterisk+fax - fax to email
I am trying to setup fax to email with asterisk with no success.
I have asterisk 1.2.9.1 running on CentOS
i have created extension 300 which should receive faxes.
extensions.conf
unplug ha scritto:
I feel interested about you can support 16,000 users of your system.
As I have tested using sipp in a dual CPU Xeon with 2G Ram, the
maximum number of current call is about 160. In some forums, most of
ppl claim the maximum current call is about 100-200. What do you
expect
Hi,
We're using Polycom IP501 SIP phones (app version 1.6.4.0043) with
Asterisk 1.2.9.1. I set up call parking last week and for a while, it
worked great. It stopped working yesterday, all of the sudden. What
happens is that when the phone user dials #999 (our parkext), the call
does not get
Dovid Bender schrieb:
several thousand extensions or several extensions called 1000 ?
Several thousend extensions.
exten = 497111234,1,goto(...)
exten = 497111235X,1,goto(...)
exten = 497111236XX,1,goto(...)
exten = 497111237,1,goto(...)
Several thousend extensions of maybe different
On Jul 12, 2006, at 7:16 AM, Rich Adamson wrote:
Andrea Spadaccini wrote:
Hello,
I need to install Asterisk on a test machine that will soon become a
production environment.
Do you think that 1.2.9.1 is reliable? I read some posts that say it
isn't as good as the previous versions. Should I
On Jul 12, 2006, at 7:24 AM, Rick Smith wrote:
Bill's right. But, it happens to me too, ALL the time, w/Teliax.
I can't wait for their NYC node...
I found that sellvoip.net is closer to me here in Seattle and also has
a better rate. They also have a server in NYC. I still like Teliax
Kevin Savoy wrote:
Asterisk is trying to send an email to users when they receive a
voicemail. Can this be shut off? I have not entered any email addresses
in voicemail.conf so it tries to send to [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]. This of course gets rejected
since the user does not
On Jul 12, 2006, at 7:45 AM, Derek Whitten wrote:
A comcast representative told me the other day they are planning on
doubling their
internet speed from 8Mb to 16Mb at the end of this year.
They certainly don't deliver anywhere near 8Mbits per second here...
So I don't know what those kind
Hi all,I'm new to asterisk and I just installed it.I already have a PCMCIA THOR-2 which supports two T1/E1 and I noticed asterisk doesn't support this card, but I fount over the ineternet that I would have to write a glue code in order to work with this card.
I'm not familiar with this code
Hi,In my Outbound routes i have created International Local Calls. I have 2 trunks for both ITL and LC. All calls are dialed using 011.but all 011254, 01125473, 01125472 should use the local trunk. NB Local Route is 1st priority in my list or routes. Everyone has to dial 011(number) to make a
I have the same problem before with 2 different providers. We resolved this
by turning off qualify (qualify=no).
KC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Filip
Dragowski
Sent: Wednesday, July 12, 2006 6:32 AM
To: Asterisk Users Mailing List -
-Original Message-
From: Simone Cittadini [mailto:[EMAIL PROTECTED]
Sent: Wednesday, July 12, 2006 10:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] where the bottleneck lies ? (was:
Serverredundancy)
unplug ha scritto:
I feel
Indicating rfc2833 is exactly what asterisk does when it receives an
invite from a server or device that indicates rfc2833 is available
regardless of whether or not dtmfmode=inband.
I will get a sip debug and open a bug report when I have a few minutes.
Kevin P. Fleming wrote:
-
Martin Joseph wrote:
On Jul 12, 2006, at 7:45 AM, Derek Whitten wrote:
A comcast representative told me the other day they are planning on
doubling their
internet speed from 8Mb to 16Mb at the end of this year.
They certainly don't deliver anywhere near 8Mbits per second here... So
I
I'm trying to do something I've not see written up here before. I have
an asterisk on a box with 2 interfaces like the drawing below. I want
to have SIP extensions regsitering to both interfaces and able to
communicate. Is this possible? What suggestions do you have?
Hi,
I`m looking for a
SIP-PSTN adapter, basically to switch a customer from a cheap PBX to mine, but
resuing their own Norstar PSTN phones. They have 10phones.
>From a price point of view, it seems that 10 individual GrandStream SIP adapters
is the best way to go, but it seems so inelegant
Kevin Smith wrote:
Michael,
Maybe I am not understanding your question, are you saying that when you
configure your phone with a static IP address, you cannot find the boot
server and when in DHCP you can?
The phone uses DHCP to get its IP address. In the phone's server
params, I enter
On Jul 12, 2006, at 10:18 AM, KC wrote:
I have the same problem before with 2 different providers. We resolved
this
by turning off qualify (qualify=no).
Of course this doesn't fix anything, it just stops the warnings from
showing up...
Marty
My working experience with 100s of extensions, usually associated to
personnel that will *not* change from my defaults is:
; Extensions
exten = 1000,1,Macro(call-sip-local,1000,SIP/1000,default) ; Operator
exten = _1XXX,1,Macro(call-sip-local,${EXTEN},SIP/${EXTEN},default)
Then,
Henry J. Cobb wrote:
I tried several different combinations of app_conference and Asterisk
versions and then I had to get back to actually providing phone service
that didn't crash.
I hate to me-too, but my experience was identical. Crash after crash,
and I tried everything that was
Why can't you do it?
I have an internal 192.168.100.x address (eth0) and 200.x.x.x (eth1)
interface. Internal users register to the 192 and internet users
register to the 200.x address
internal extensions are 1XXX and external extensions are 2XXX
What errors do you have?
On 7/12/06, Bruce
I have attach=no in my voicemail.conf so that can't be doing it. Not sure
where that sendmail command is. Don't see it in voicemail.conf or any other
config in the asterisk directory.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of VoIP Street
Sent:
the problem I'm seeing is one way audio between extensions. I've splpit
up the numbering plan internal/external. All are in the same range.
I'll try splitting them and see what happens.
Erick Perez wrote:
Why can't you do it?
I have an internal 192.168.100.x address (eth0) and 200.x.x.x
thanks brian, this is all really helpful feedback!just to be clear, which app_conference code were you using?the svn trunk version from sourceforge? or the VD_app_conference matt's been working on?
j-On 7/12/06, Brian Capouch [EMAIL PROTECTED] wrote:
I hate to me-too, but my experience was
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