Hello,
LLDP is more and more available on various network elements (endpoint,
switches, ...).
It seems to ease network configuration.
Do you have any experience with it ?
How would you rate LLDP ?
Regards
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Hi,
I have 4 ISDN channels (2 lines) and each line may do calls of up to 360
minutes/month for free.
As I understand asterisk will pick the first available line so the probability
is big that the other lines will not use their free minutes and the firs line
will exceed the free minutes.
How
Anyone can recommend a commercial large scale IVR with easy + pro management
for asterisk?
Thanks.
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Couldnt you do this by calling MySql? Compare who has the least minutes
used and then send it out the appropriate channel?
--Matt
On Tue, Nov 24, 2009 at 7:07 AM, Eckhard Jokisch
e.joki...@orange-moon.dewrote:
Hi,
I have 4 ISDN channels (2 lines) and each line may do calls of up to 360
Hello again,
I just tried version 1.6.1.9, and the MOH works well there. It seems to be a
bug introduced in 1.6.1.10.
Best regards,
Örn
2009/11/23 Örn Arnarson o...@arnarson.net
Hello.
I just upgraded from 1.6.0.9 to 1.6.1.10 and it seems that the Music On
Hold functionality has changed
Hello.
I'm running some asterisks in a small voip provider in Brazil and we're
having some problems with a analogic channel bank.
When I make calls using analogic extensions, I have a crystal clear
quality, but the receptor have a lot of noise and echo.
We tested the situation using SIP, E1
On Tue, Nov 24, 2009 at 01:07:28PM +0100, Eckhard Jokisch wrote:
Hi,
I have 4 ISDN channels (2 lines) and each line may do calls of up to 360
minutes/month for free.
As I understand asterisk will pick the first available line so the
probability
is big that the other lines will not use
Is there a way to keep asterisk in RAM
and tell linux not to swap it out (ever).
There are times when delays are noticed and I presume
its due to linux swapping out the program. As if I call right back in
then everything responds right away. Wait awhile and the same thing
might occur.
How can I
Disable swap space.
swapoff -a
Jerry Geis wrote:
Is there a way to keep asterisk in RAM
and tell linux not to swap it out (ever).
There are times when delays are noticed and I presume
its due to linux swapping out the program. As if I call right back in
then everything responds right
On 24 Nov 2009, at 13:05, jefferson alexandre wrote:
I'm running some asterisks in a small voip provider in Brazil and
we're having some problems with a analogic channel bank.
When I make calls using analogic extensions, I have a crystal
clear quality, but the receptor have a lot of
Next question will be How can I keep my server from crashing? :)
(add more RAM... which may have been a good answer for question 1...)
j
On Tue, 24 Nov 2009, Alex Balashov wrote:
Disable swap space.
swapoff -a
Jerry Geis wrote:
Is there a way to keep asterisk in RAM
and tell linux not
On Tue, Nov 24, 2009 at 08:21:43AM -0500, Jerry Geis wrote:
Is there a way to keep asterisk in RAM
and tell linux not to swap it out (ever).
There are times when delays are noticed and I presume
its due to linux swapping out the program. As if I call right back in
then everything responds
I was proceeding from the give them enough rope to hang themselves
theory of technical support, which calls for doing just that when users
insist on framing their question in terms of a solution they have
already made up their mind on without examining whether they are asking
the right
On Tue, Nov 24, 2009 at 11:21 AM, Jerry Geis ge...@pagestation.com wrote:
Is there a way to keep asterisk in RAM
and tell linux not to swap it out (ever).
There are times when delays are noticed and I presume
its due to linux swapping out the program. As if I call right back in
then
Hi,
I think it can be related to https://issues.asterisk.org/view.php?id=16268
Best regards,
Santi
2009/11/24 Örn Arnarson o...@arnarson.net
Hello again,
I just tried version 1.6.1.9, and the MOH works well there. It seems to be
a bug introduced in 1.6.1.10.
Best regards,
Örn
On Tue, Nov 24, 2009 at 11:29 AM, Steve Howes steve-li...@geekinter.netwrote:
On 24 Nov 2009, at 13:05, jefferson alexandre wrote:
I'm running some asterisks in a small voip provider in Brazil and
we're having some problems with a analogic channel bank.
When I make calls using analogic
On Tue, Nov 24, 2009 at 2:36 PM, jefferson alexandre
jefferson.alexan...@gmail.com wrote:
On Tue, Nov 24, 2009 at 11:21 AM, Jerry Geis ge...@pagestation.com wrote:
Is there a way to keep asterisk in RAM
and tell linux not to swap it out (ever).
On a closely related note, has anyone built a
On 24 Nov 2009, at 13:48, jefferson alexandre wrote:
Steve, the hardware don't have echo cancellation.
Thats probably it You're relying on Asterisks software echo
canceling I have seen mixed results. Have you tried adjusting
gains? I'd do the following
1. Turn off echo canceler
On Monday 23 November 2009 21:30:10 Eric Chamberlain wrote:
We've encountered a strange issue with the trunk version of asterisk.
Our dialplan makes CURL calls and occasionally CURL stops working.
The dialplan looks something like this:
[macro-curl]
; ${ARG1} CURL URL
; ${ARG2} CURL POST
Ok Steve. I will follow these instructions until i get some results.
The next time, I will consider buy a hardware with echo canceller ;)
On Tue, Nov 24, 2009 at 12:05 PM, Steve Howes steve-li...@geekinter.netwrote:
On 24 Nov 2009, at 13:48, jefferson alexandre wrote:
Steve, the hardware
On a closely related note, has anyone built a normal (not embedded)
system on SSD?
I've been running Asterisk on a 20GB SSD drive for a while now.
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asterisk-users mailing list
I recently implemented a vmware host using SSDs for the VM storage.
I wish you could see the grin on my face right now. It's so fast.
Remember thought that all SSDs are NOT created equal... Be careful what you buy.
snip
On a closely related note, has anyone built a normal (not embedded)
If you have a network that doesn't support CDP (such as an all Juniper
network), LLDP will do the job for you, as long as your phone supports
it. The latest Polycom sip firmware supports it (but none of their
older phones can run the new firmware, just the newer ones), as well
as the
On Tue, Nov 24, 2009 at 3:42 PM, Richard Kenner ken...@gnat.com wrote:
On a closely related note, has anyone built a normal (not embedded)
system on SSD?
I've been running Asterisk on a 20GB SSD drive for a while now.
And? Noticed any significant performance advantage?
/r
On Tuesday 24 November 2009 07:21:43 Jerry Geis wrote:
Is there a way to keep asterisk in RAM
and tell linux not to swap it out (ever).
There are times when delays are noticed and I presume
its due to linux swapping out the program. As if I call right back in
then everything responds right
On Tue, 24 Nov 2009, Richard Kenner wrote:
On a closely related note, has anyone built a normal (not embedded)
system on SSD?
I've been running Asterisk on a 20GB SSD drive for a while now.
What mft/model?
I was recently quoted a 4GB Compact Flash drive as part of a small system
we plan
What mft/model?
Actually, it's 16GB, not 20GB. It's a Transcend TS16GSSD25S-S.
I know that CF cards have a limited number of writes before frying.
If we keep it from using swap am I really only concerned about
voicemail and logs?
That number is quite large, though. I'm taking backups and
On Tue, 24 Nov 2009 14:56:32 + (UTC), Jeff LaCoursiere wrote:
On Tue, 24 Nov 2009, Richard Kenner wrote:
On a closely related note, has anyone built a normal (not embedded)
system on SSD?
I've been running Asterisk on a 20GB SSD drive for a while now.
What mft/model?
I was recently
And? Noticed any significant performance advantage?
I never ran it any other way, so have no comparison point. I didn't do it
for performance reasons, but reliability.
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snip
And? Noticed any significant performance advantage?
/snip
Massive increase in performance on mysql VMs with database sizes that exceed
memory size (file caching). Boot times on VMs (windows and linux) under 10
seconds.
There is no noticeable change in performance for normal operations on
Norbert Zawodsky wrote:
Leif Neland schrieb:
- Original Message -
*From:* Norbert Zawodsky mailto:norb...@zawodsky.at
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com
*Sent:* Monday, November 23, 2009
Can you move the transfer functionality to the end device rather than
through Asterisk? That's what we do - John
On Tue, 2009-11-17 at 14:07 +0100, Ignacio wrote:
Thank you very much to both of you.
My problem was that I used transfer in the dialplan. I have read that
If I have Tt, wW, or
Hi All,
I am having a problem with the ring group where when an incoming call comes
it rings all the 3 extensions associated to that, but intermittently it
rings one extension only once, but the others will be continuously ringing
and the goes to generalized voicemail. When I check the log using
What is the channel technology in use?
das sandesh wrote:
Hi All,
I am having a problem with the ring group where when an incoming call
comes it rings all the 3 extensions associated to that, but
intermittently it rings one extension only once, but the others will be
continuously
Hi all,
I'm running 1.6.2.0-rc6, and I'm running into a problem: sometimes the
audio vanishes in the middle of listening to an IVR background prompt.
This happens with both analog (Digium card) and IAX2 incoming calls.
The prompts are stored in ulaw format (and the IAX2 calls use ulaw).
The
I am talking about the endpoints (extensions). SIP? DAHDI? IAX? H.323?
das sandesh wrote:
Hi Alex,
I am using Ring All channel strategy...
Thanks
Sandesh
On Tue, Nov 24, 2009 at 10:21 AM, Alex Balashov
abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote:
What
Hi Alex,
I am using Ring All channel strategy...
Thanks
Sandesh
On Tue, Nov 24, 2009 at 10:21 AM, Alex Balashov
abalas...@evaristesys.comwrote:
What is the channel technology in use?
das sandesh wrote:
Hi All,
I am having a problem with the ring group where when an incoming call
Hello Guys,
Hope everyone is fine, I have one issue coming in asterisk , What i am doing
is i am generating a callback if some one calls at a specif access number on
asterisk,
Asterisk sends a busy signal to the calling party that he received a request
from party and then sends the call back to
We are using SIP channel technology...
On Tue, Nov 24, 2009 at 11:03 AM, Alex Balashov
abalas...@evaristesys.comwrote:
I am talking about the endpoints (extensions). SIP? DAHDI? IAX?
H.323?
das sandesh wrote:
Hi Alex,
I am using Ring All channel strategy...
Thanks
Sandesh
Which cards exactly?
It's 2 T1/E1 cards!
Specifically, on of it is a TE110P and the other is a TE122!
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Hi all,
Does anyone else use the SG560 firewall with Asterisk? I do, and it
normally works great, except when it randomly reboots. Has anyone else
experienced this annoyance? Did you fix it?
- Mike
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On Nov 24, 2009, at 6:17 AM, Tilghman Lesher wrote:
Sounds like your local DNS resolver isn't answering queries promptly.
Thanks, I'll look into it. Our CURL function only calls one hostname over and
over.
Would setting CURLOPT dnstimeout be of use in this situation?
--
Eric
On Tue, 24 Nov 2009, Eric Chamberlain wrote:
On Nov 24, 2009, at 6:17 AM, Tilghman Lesher wrote:
Sounds like your local DNS resolver isn't answering queries promptly.
Thanks, I'll look into it. Our CURL function only calls one hostname over
and over.
Would setting CURLOPT
Hi All,
I'm struggling with an intermittent crosstalk issue resulting in a caller's
audio being broadcasted to other calls (only one way as they are unable to
hear the others listening in).
Doing do diligence I've scoured the web in hopes of triaging the issue.
So my thoughts are leaning
On Tue, Nov 24, 2009 at 7:12 AM, B.Masoud @ SH i...@saudihome.com wrote:
Anyone can recommend a commercial large scale IVR with easy + pro management
for asterisk?
Those words don't mean anything to anybody except you. For instance,
large scale is meaningless. You need to say out loud how large
On Tue, Nov 24, 2009 at 11:47 AM, Dr. Michael J. Chudobiak
m...@avtechpulse.com wrote:
Hi all,
I'm running 1.6.2.0-rc6, and I'm running into a problem: sometimes the
audio vanishes in the middle of listening to an IVR background prompt.
This happens with both analog (Digium card) and IAX2
On Tue, 2009-11-24 at 14:05 -0500, JT wrote:
I'm struggling with an intermittent crosstalk issue resulting in a
caller's audio being broadcasted to other calls (only one way as they
are unable to hear the others listening in).
Crosstalk like this isn't a common occurrence, especially on
mos...@infolog.mr escribió:
Which cards exactly?
It's 2 T1/E1 cards!
Specifically, on of it is a TE110P and the other is a TE122!
Hi,
That would be a very special need, I'm wondering why connect two
asterisk with expensive E1/T1 cards when you can connect them with
simple network
On 11/24/2009 02:14 PM, David Backeberg wrote:
The asterisk console claims that the IVR prompts are proceeding in the
expected fashion, but I can't hear anything.
Are you playing with the system clock?
...
dramatic ntp changes?
No, that shouldn't be happening. But I'll keep it in mind while
2009/11/24 Warren Selby wcse...@selbytech.com
If you have a network that doesn't support CDP (such as an all Juniper
network), LLDP will do the job for you, as long as your phone supports
it. The latest Polycom sip firmware supports it (but none of their
older phones can run the new firmware,
On 11/24/2009 01:19 PM, Dr. Michael J. Chudobiak wrote:
Does anyone else use the SG560 firewall with Asterisk? I do, and it
normally works great, except when it randomly reboots. Has anyone else
experienced this annoyance? Did you fix it?
Oops, never mind. The SG560 was fine. The AC power to
Hello,
I was wondering if their is a way to use the Asterisk ACD to initiate a
call that will route variables through the ACD, which can then be read at
the other end by an application. The idea here is instead of terminating a
call to an agent I would be terminating some variables/text data.
On 11/24/2009 02:14 PM, David Backeberg wrote:
I'm running 1.6.2.0-rc6, and I'm running into a problem: sometimes the
audio vanishes in the middle of listening to an IVR background prompt.
Are you playing with the system clock?
Actually, setting the internal_timing option seems to have fixed
On Tue, Nov 24, 2009 at 12:49 AM, Olivier oza-4...@myamail.com wrote:
Hello,
LLDP is more and more available on various network elements (endpoint,
switches, ...).
It seems to ease network configuration.
Makes Voice VLAN assignment much easier for sure.
Do you have any experience with it ?
Hi,
I am using codec g729 on two asterisk machines, but when call is forwarded
from asterisk server 1 (32bit) to server 2(64bit). server 1 outputs
following error and there is no audio. Also the IVRs being played have
choppy voice.
Insufficient information for SDP (m = 'audio RTP/AVP 18
ast guy escribió:
Hi,
I am using codec g729 on two asterisk machines, but when call is
forwarded from asterisk server 1 (32bit) to server 2(64bit). server 1
outputs following error and there is no audio. Also the IVRs being
played have choppy voice.
Insufficient information for SDP (m
On 25/11/09 5:47 AM, Dr. Michael J. Chudobiak wrote:
Hi all,
I'm running 1.6.2.0-rc6, and I'm running into a problem: sometimes the
audio vanishes in the middle of listening to an IVR background prompt.
This happens with both analog (Digium card) and IAX2 incoming calls.
The prompts are
On Tue, Nov 24, 2009 at 10:49 PM, Miguel Molina mmol...@millenium.com.cowrote:
ast guy escribió:
Hi,
I am using codec g729 on two asterisk machines, but when call is
forwarded from asterisk server 1 (32bit) to server 2(64bit). server 1
outputs following error and there is no audio.
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 rings and I can receive calls but I cannot dial with
the rotary dialer. I have set pulsedial=true or whatever the exact
setting
Hi,
I've recently discovered Dial examples such as Dial(DAHDI/g4d/${EXTEN})
but I wonder where I can get an uptodate doc.
Is there any CLI option such as core show channel dialoption that would
explain what g4d exactly means ?
core show application dial doesn't explain much about those
2009/11/24 Jonathan Thurman jthurma...@gmail.com
I would rate LLDP as a very useful vendor-agnostic protocol.
-Jonathan
So I guess, the next item on my todo list is to test LLDP !
Thanks for the advice.
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On Wed, Nov 25, 2009 at 1:12 AM, B.Masoud @ SH i...@saudihome.com wrote:
Anyone can recommend a commercial large scale IVR with easy + pro management
for asterisk?
I don't know what you mean by pro management but you can write IVR
applications in any language you want. Personally I like ruby
Have you tested the dialer mechanism to confirm that it actually works?
Sounds like your dialer mechanism MAY not be opening/closing the loop
properly.
Have you tried using the phone on a Telco-provisioned loop?
-K
- Original Message -
From: Mike asterisk-us...@norgie.net
To:
On Tue, Nov 24, 2009 at 4:16 PM, Landy Landy landysacco...@yahoo.com wrote:
How about adding:
insecure=invite,port
That didn't work.
How weird.
I have reset the device to factory settings too. Nothing seems to work.
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On Tue, Nov 24, 2009 at 3:38 PM, Michael Wyres mwy...@cdm.com.au wrote:
I would without the deny and permit directives in the SIP, and rule out
some sort of clash there that is rejecting the address the registration is
coming from, and take it from there.
It made no difference to remove
On Tue, Nov 24, 2009 at 06:55:16PM -0600, Karl Fife wrote:
Have you tested the dialer mechanism to confirm that it actually works?
Sounds like your dialer mechanism MAY not be opening/closing the loop
properly.
Have you tried using the phone on a Telco-provisioned loop?
-K
Yeah, I've
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