why don't you write an AGI which talks to asterisk manager API 5038 port and
executes the asterisk commands. You execute asterisk command via agi not
using system command
-ag
On Feb 11, 2008 11:24 AM, Rob Hillis [EMAIL PROTECTED] wrote:
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Hi guys,
Hi,
I have a scenario that * Server A ( behaving as client) has sip peers, P1,
P2, P3 with different contexts. Peers register to another * or any other SIP
server. Using realtime * I am able to create a peer entry in sip buddies
table and a register statement in sip.conf on client side and it's
If this is the only real alternative, then in this instance I'll stick
with using the System command. Writing an AGI to execute two manager
commands in this case is even greater overkill than using the System
command.
I understand that normally anything that calls multiple manager commands
would
One thing to keep in mind is that the Grandstream's firmware is
notoriously buggy and unreliable. I've got one GXP2000 here that is on
the 1.1.5.15 firmware, and I wouldn't even consider upgrading other
phones to them. Unfortunately, the quality of the Grandstream firmware
is appalling and
Peder @ NetworkOblivion wrote:
I hate to reply to my own message, but I have some more info from
debugging. A Grandstream tries to register and uses a nonce and it is
accepted by *. The next time it tries to register, it uses the same
none and * says SIP/2.0 401 Unauthorized. The
Retail for these phones are $300 but I think in quantity they will be $200
John Bittner
Simlab.net
-Original Message-
You're right, these phones look great !
Do you have an idea about its price (for a 100 quantity) ?
It would be interesting to know how to localize its menu ?
(you
Hi John,
I have managed to get the 1140E phones working using the UniSTIM driver
compiled into my asterisk server. The config wasn't too tricky as it is
very well documented. The only real problems I had was the Nortel phones
talking to my LIP-68000 SIP handsets. They seemed to have some
Quoting Jaap Winius [EMAIL PROTECTED]:
After wrestling with the voicemail system for a while (Asterisk 1.4.14,
Debian etch), I got it to work, but I still have lots of questions,
like:
* Why can't I delete any voicemail messages?
(Response: Message undeleted.)
* Why can't I
Can you test this whole thing with a phone?
On Feb 11, 2008 4:52 PM, Peder @ NetworkOblivion
[EMAIL PROTECTED] wrote:
I have 20-30 GXP2000's connected to * over a T1 line. Neither end is
NAT'd and there is plenty of bandwidth available over the line. The
GXP's are 1.1.5.15, which is the
What kind of test would you like? An IAX call for example?
On Feb 12, 2008 8:29 AM, TC [EMAIL PROTECTED] wrote:
ANeyone ??
do we have no asterisk users in costa rica that would be so kind as to allow
some usability testing
I am considering an extended vacation but will need to some ggos VoIP
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If it is being removed in 1.6, I'm a little concerned since there's no
mention of this when you show the application, nor on voip-info.org.
What application/function is it being replaced by?
Atis Lezdins wrote:
| On 2/13/08, Rob Hillis [EMAIL
Matt Riddell wrote:
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Rilawich Ango wrote:
Hi all,
I found that there will be a memory leak if asterisk running day by
day without restart. Is it good to restart asterisk service daily?
What is the better way to restart it daily like apache?
Rizwan Hisham wrote:
Hi all,
I am planning to implement LCR routing on my already running asterisk
server. Uptill now i have found out that asterisk has no support for
lcr, i have to do something about it myself, for example using the AGI.
Im looking for ideas here. Whats the best way to
Hi all,
I have a Netgear TA612V voip adapter which I am trying
to convince to work with asterisk. If I activate one of the two lines
(line one or line two). The unit registers with the server no problem. If
I try to register both lines with different usernames passwords the
registrations fail
Hi,
I am using Asterisk 1.4.13
The call comes in from a Coppercom soft-switch with a private IP of
192.168.104.2. This gets forwarded to a SessionBorderController with public
IP of x.x.x.x,. This then gets to our asterisk server with a public IP of
8.7.192.58 and a private IP of 192.168.5.0.
On 2/13/08, Rob Hillis [EMAIL PROTECTED] wrote:
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Atis Lezdins wrote:
| By RealTimeUpdate do you mean func_realtime? It shouldn't care, as
| cache is not implemented in realtime level, but higher (chan_sip).
|
| Are you sure you need sip show XXX
I found that there will be a memory leak if asterisk running day by
day without restart. Is it good to restart asterisk service daily?
What is the better way to restart it daily like apache?
Probably depends on the version of Asterisk, but I don't restart daily
From one in production used
Mohammad Salaque wrote:
Dear all,
Anyone can point me how to soft hangup all channels using single
command ? I am using Asterisk 1.4.15.
Such a command does not exist.
You could run a simple script to do this, except for the annoying fact
that 'show channels' truncates the full channel
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Rilawich Ango wrote:
Hi all,
I found that there will be a memory leak if asterisk running day by
day without restart. Is it good to restart asterisk service daily?
What is the better way to restart it daily like apache?
What makes you think
Sorry for the late follow-up to this... it was on my to-do list for
over a month... Sigh.
I've submitted a configuration bug and for this:
http://bugs.digium.com/view.php?id=11969
The hope being that if the examples provided in the configs/ directory
work better out of the box for
I'm currently getting SIP trunking from my PSTN provider, but they don't
quite grok the whole any-service/any-device philosophy...
I'm wondering if it's possible to get SIP voice carriage from one
provider, but have SMS associated with the same phone numbers being
provided by another carrier?
Hi all,
I found that there will be a memory leak if asterisk running day by
day without restart. Is it good to restart asterisk service daily?
What is the better way to restart it daily like apache?
ango
___
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The release notes in the Subversion trees and bugs.digium.com will
probably serve to illuminate that.
Khaled Chehab wrote:
What are the differences between asterisk 1.2.4 and 1.4.6 beta
In functionality ,services and bugs.
Regards
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Atis Lezdins wrote:
| By RealTimeUpdate do you mean func_realtime? It shouldn't care, as
| cache is not implemented in realtime level, but higher (chan_sip).
|
| Are you sure you need sip show XXX load. If you sip prune peer
| data, it should be
Tilghman Lesher wrote:
On Monday 11 February 2008 11:55, Mojo with Horan Company, LLC wrote:
William F. Acker WB2FLW +1-303-722-7209 wrote:
Thanks for mentioning contexts. All of us are in the default
context. So I started playing around with the options pertaining to
Tilghman Lesher wrote:
On Monday 11 February 2008 11:55, Mojo with Horan Company, LLC wrote:
William F. Acker WB2FLW +1-303-722-7209 wrote:
Thanks for mentioning contexts. All of us are in the default
context. So I started playing around with the options pertaining to
I have voicemail configured to store messaging in an odbc database. Does
anyone have any thoughts on how best to play back someones greeting from
the db?
Thanks,
~jerry
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On 07/02/2008, Brent Davidson [EMAIL PROTECTED] wrote:
We're deploying an asterisk-based phone system at all of our branch
offices in an effort to eliminate long-distance costs incurred from the
constant branch to branch calls. We're using the Snom 300's at all
offices for the desk phones and
I thought I had the echo out of the system, but it keeps coming back...
What I'm being told is that when the users call out from their snom
phones they hear their own voice. There's no delay, but it's extremely
loud. If I cut their mic volume down to the point where the sidetone is
not a
On Mon, 11 Feb 2008 21:18:17 -0500, John Bittner wrote:
Anyone get the Nortel 1140E phones working with Asterisk ?
These look like great phones and I would like to start using them on
our deployments. I know these will work with Asterisk but the sample
config files are hard to find. My next step,
asterisk -rx 'restart now'
PaulH
On Wed, 2008-02-13 at 13:49 +1100, Mohammad Salaque wrote:
Dear all,
Anyone can point me how to soft hangup all channels using single
command ? I am using Asterisk 1.4.15.
thanks
Salaque
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On Feb 12, 2008 10:40 AM, Ian [EMAIL PROTECTED] wrote:
Hi all,
its been quite a busy few day with pc's packing up etc, I recompile my
whole asterisk today using zaptel 1.4.7.1 and now the problem is
miraculously fixed, I will be sending this report to Digium bugs as well.
Just a quick
Brent Davidson wrote:
I thought I had the echo out of the system, but it keeps coming back...
What I'm being told is that when the users call out from their snom
phones they hear their own voice. There's no delay, but it's extremely
Does it happen on all-digital calls (e.g., intercom
I am having similar problems running the same versions of Asterisk,
libpri zaptel.
The Asterisk bug (http://bugs.digium.com/view.php?id=11855) was
supossed to be related to FXO only, but I am having issues with a PRI
line and Digium's TE120P.
Do you guys think it can be the same issue?
--
Dear all,
Anyone can point me how to soft hangup all channels using single
command ? I am using Asterisk 1.4.15.
thanks
Salaque
___
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asterisk-users mailing list
To UNSUBSCRIBE or
I want to make sure that 192.168.1.10 is the IP of the asterisk machine.
Anyways without looking at the phone's config I'd recommend you switch
to a newer Asterisk 1.2.x release at the very minimum (I suppose it
would be a best practice to recommend 1.4, however) all you will
need is a very few
Hello,
I have put dozens of these in production at client sites. Here is the
perl script I use to auto-config them. I have also seen images that
you can tftp to them in the past, but I can't seem to find them
through google at the moment.
I have a Cisco 7912G connected to Asterisk using SIP. When I'm trying to
call out via my Fritz! PCI card in a CAPI context the call is not bridged
instantly after the called party answers the call. I always have a 2-3
seconds timeout, after the called party answers.
This problem just appears,
On 2/13/08, Rob Hillis [EMAIL PROTECTED] wrote:
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Hash: SHA1
If it is being removed in 1.6, I'm a little concerned since there's no
mention of this when you show the application, nor on voip-info.org. What
application/function is it being replaced by?
Hi all,
I am planning to implement LCR routing on my already running asterisk
server. Uptill now i have found out that asterisk has no support for lcr, i
have to do something about it myself, for example using the AGI. Im looking
for ideas here. Whats the best way to start implementing lcr in
Hi all,
its been quite a busy few day with pc's packing up etc, I recompile my
whole asterisk today using zaptel 1.4.7.1 and now the problem is
miraculously fixed, I will be sending this report to Digium bugs as well.
Just a quick heads up for the order in which I had to recompile in order
Hi,
I tried to use chan_mobile with an ericksson w300i with no chance (few audio
unresolved problems). I'm planning to get a new mobile in order to use it
with asterisk. I'd like to use it with chan_mobile. I'd like to hear
successfull or unsuccesfull experience from some user in order to
ANeyone ??
do we have no asterisk users in costa rica that would be so kind as to allow
some usability testing
I am considering an extended vacation but will need to some ggos VoIP in N.A ?
- Original Message -
From: TC
To: asterisk-users@lists.digium.com
Sent: Sunday,
I am using about 70 Grandstream GXP2000 phones on 1.1.5.15 code with *
1.4.16.2 and have not experienced any of these issues. The one thing
that I would suggest is make sure that you are using RFC2833 for you
DTMF Mode. I was originally using INFO and ran into some strange issues
with dropped
On 2/12/08, Rob Hillis [EMAIL PROTECTED] wrote:
This is implemented in the Asterisk Dialplan.
What we're doing is to write a custom roaming extension application that
(among other things) alters the mailbox that the device looks at to set the
MWI indicator to that of the roaming extension.
This /is/ implemented in the Asterisk Dialplan.
What we're doing is to write a custom roaming extension application
that (among other things) alters the mailbox that the device looks at to
set the MWI indicator to that of the roaming extension. All the
systems we sell have SIP peers stored
On 2/12/08, Rob Hillis [EMAIL PROTECTED] wrote:
If this is the only real alternative, then in this instance I'll stick with
using the System command. Writing an AGI to execute two manager commands in
this case is even greater overkill than using the System command.
I understand that
Le lundi 11 février 2008, bilal ghayyad a écrit :
Hi All;
How can I let Asterisk start automatically once the
machine restarted without need to type asterisk -cvvv?
you need a init script. Depends on your distribution. It looks like some
packages exists : asterisk-initscript for example.
This may be a long shot, but I suppose it can't hurt to ask!
I have seen chatter regarding Zhone channel banks amongst the asterisk users
primarily. In all of my searching I am guessing that the majority of people
using the Zhone's are probably reading this, and with the support that I have
I got my Cisco PIX reconfigured as the below given. The issue one-way
audio, still exists. Here is the call flow. The call comes on an inbound
trunk to asterisk. Asterisk plays an IVR. When the user presses 5 it makes
an outbound call Dial(SIP/[EMAIL PROTECTED],30) using the same
inbound trunk.
Lutgring, Sam wrote:
I am using about 70 Grandstream GXP2000 phones on 1.1.5.15 code with *
1.4.16.2 and have not experienced any of these issues. The one thing
that I would suggest is make sure that you are using RFC2833 for you
DTMF Mode. I was originally using INFO and ran into some
Actually, I donno it is a memory leak or not. I have a server only
running asterisk. As time goes by, the free memory shown in the top
is decreased. After I restart the asterisk, the free memory comes
again. That's why I wonder if regular restart asterisk is necessary.
Use a crontab to restart
On 2/13/08, Rilawich Ango [EMAIL PROTECTED] wrote:
Hi all,
I found that there will be a memory leak if asterisk running day by
day without restart. Is it good to restart asterisk service daily?
What is the better way to restart it daily like apache?
ango
I have cron script that restarts
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