Hi Tzafrir
I really appreciate all teh help yo have given me over the last few daysI
seemed to have solved my problemI upgraded to latest Asterisk after reading
about incompatibility between Zaptel 1.4 and older Asterisks version...
Now I have Asterisk 1.4.2 and it is working fine...
Saul Bejarano wrote:
Remember the rule of 30Mhz per call when you kill the machine and also
the bandwidth usage on connected calls.
Kind regards,
Saul Bejarano
aby azid wrote:
Hi everyone,
I'm required to make a stress call on Asterisk server ( 2000 calls
per seconds). Are
Ouch - upgrade.
PaulH
Jay Ray wrote:
Yes, I am running 1.0.5
--- On *Thu, 8/14/08, Tzafrir Cohen /[EMAIL PROTECTED]/* wrote:
From: Tzafrir Cohen [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Unable to create ZAP Channel
To: asterisk-users@lists.digium.com
Date:
I had done something similar in the past, and I have one suggestion that
may be helpful; the call files are set to be checked every second, which
can bottleneck the system a bit. You can modify the code (pbx_spool.c)
to check in fractions of a second, which should keep the calls more
fluid.
Hi,
Thanks for the reply mates, to Al Baker, It's a stress test for Asterisk
outgoing calls, this is to see how Asterisk cope when thousands(1000 - 2000)
of calls made simultaneously from the server.
To Mik, where do I find the pbx_spool.c ?, really appreciate if u can
explain more details on
OK - but again - more specifics are needed.
If you are going TDM over T1 that is a Totally Different Animal
than cranking up all these using IAX or .
Also, you still have to identify how many simultaneous calls you will have.
Again 1000 calls done essentially all at once is a different animal
Congratulations with the release!
I'm curious also about the statement on your page:
Realtime Asterisk uses the MySQL relational database to access dialplan,
extension and configuration data.
This allows for dynamic additions and changes to users, extensions and
dialplans without having to
hi saul,could you give me more info on the "VMX-CONTEXT" concept? i tried to google it, but could find nothing.i am trying to do exactly what you state in terms of creating a virtual slice of the box for each user. thanks!todd
Original Message
Subject: Re: [asterisk-users]
2008/8/14 Jon Weisman [EMAIL PROTECTED]
Hey guys its me again! So I need to setup our Asterisk server with multiple
IP providers. The server has two NICs, we have two providers, now we want
redundancy! Any guides on how to set this up? We're running Fedora Core 5
w/
Asterisk 1.4.
What I
On Fri, Aug 15, 2008 at 8:31 AM, aby azid [EMAIL PROTECTED] wrote:
Hi everyone,
I'm required to make a stress call on Asterisk server ( 2000 calls per
seconds). Are there tools for me to do this sort of test. I was thinking of
sending loads of Asterisk call files simultaneously (starting
Todd Fulton wrote:
Hi,
I'm trying to create a multi-tennant asterisk installation where
each of my customers has its own context. Well, I've got asterisk
realtime working, and I can add/update extensions to existing contexts
in extensions.conf without a problem. However, when I
Observation: When the agents talk using the polycom 650 handset the
voice quality on the other end is much better compared to if they talk
using the plantronics noise cancelling headset. If you would like a
recorded phone call when a agent is taking using the headset vs when a
agent is talking
Karl Fife wrote:
Does anyone know enough about the implementation of AstDB to know
whether the data structure is a Hash function, a Balanced-Tree, a
b-Tree, or a Linked List?
I've never looked at the internals of db1. However, by simply looking
at what code is included, it looks like it is
Lee Lundrigan wrote:
Hi everyone,
Are there any incompatibility issues between asterisk and the c-client
using SSL?
When I enable SSL I get the error:
*pbx.c:1832 pbx_extension_helper: No application 'VoiceMailMain'
*whenever I am trying to access voicemail.
But when SSL is disabled
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED],
altrimenti vi risponderò al mio rientro.
Dimitri Osler
I
On Fri, Aug 15, 2008 at 8:56 AM, Karl Fife
[EMAIL PROTECTED] wrote:
Does anyone know enough about the implementation of AstDB to know
whether the data structure is a Hash function, a Balanced-Tree, a
b-Tree, or a Linked List?
I'm trying to estimate the lookup 'cost' of a AstDB with around
thanks! this definitely helps. now, i'm trying to think of a way to make this happen on multiple asterisk nodes at once. i really wish realtime would simply read new contexts from the db. i know that the dialplan is core to the system, but i think this core aspect should be ultimately configurable
Hi,
We have a few Aastra 480ci phones and we've noticed that in order to
get the phone to receive a call, qualify must be = no.
Apparently the Aastras do not respond to the qualify message (or
respond in a way Asterisk doesn't understand) and Asterisk thinks the
phone is unreachable.
However, this
I've just started consulting for a SME client based in Reston Virginia.
They don't know it yet but they are going to need a hosted asterisk
service and some DID's.
Email me if you are able to provide 10 DID's in Reston (must be able to
be ported away!!) and hosted Asterisk with end user
James Lamanna wrote:
Hi,
We have a few Aastra 480ci phones and we've noticed that in order to
get the phone to receive a call, qualify must be = no.
Apparently the Aastras do not respond to the qualify message (or
respond in a way Asterisk doesn't understand) and Asterisk thinks the
phone is
I am trying to lauch a first outbound call.
I am connected to my telco via a peer which is a little different from what I
consider the norm.
extinsions.conf
[To_Bandwidth]
ignorepat = 9
exten = 9,1,Dial(Sip/g2/)
exten = 9,2,Congestion
sip.conf
[To_Bandwidth]
canreinvite=yes
context=from-pstn
On Thursday 14 August 2008 03:09:42 pm roberto wrote:
I'm looking for some free LATA X Area Code database.
Anyone have any idea where can i found?
this site has lots of info: http://www.localcallingguide.com/
--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72
On Thu, Aug 14, 2008 at 03:23:57PM -0500, Tilghman Lesher wrote:
On Thursday 14 August 2008 13:59:37 Philipp Kempgen wrote:
Jared Smith schrieb:
On Thu, 2008-08-14 at 20:35 +0200, Philipp Kempgen wrote:
Whenever something spits out lines with a different background
color (of a varying
Check if you have some rule to dial under brad1 context
dialplan [EMAIL PROTECTED]
Regards
Felippe Silvestre
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brad
Sent: Friday, August 15, 2008 12:09
To: Asterisk Users Mailing List - Non-Commercial
On Fri, Aug 15, 2008 at 12:56:49AM -0500, Karl Fife wrote:
The key-space is ideal. It's just npa/nxx lookups so it's UNIQUE and
EVENLY DISTRIBUTED
Based on my knowledge of the NPA/NXX space, I wouldn't expect that
either
a) A given batch of random DNs would have either or both NPA/NXX
On Fri, Aug 15, 2008 at 04:59:26AM -0700, Vikas wrote:
Observation: When the agents talk using the polycom 650 handset the
voice quality on the other end is much better compared to if they talk
using the plantronics noise cancelling headset. If you would like a
recorded phone call when a agent
In the past there was ztdummy - what is the new equivalent in dahdi?
Also it used to be Zap/X what is the new channel name?
searching voip-info.org for dahdi didnt show me anything about that...
Thanks,
Jerry
___
-- Bandwidth and Colocation Provided
An educated guess is:
reverse the SIP trunk buttons, so the preferred provider is the top
button, and voila, your speed dial going to the first trunk is now
what you want.
On Wed, Aug 13, 2008 at 7:44 PM, Shawn L [EMAIL PROTECTED] wrote:
This one is a little off-topic, it's more about the phone
Hi,
Just wondering if i can use a card just for timing instead of just using
ztdummy? maybe use 2 port fxs or fxo card so it would not be costly.
will there be any difference on the meetme performance if i used a
hardware compared to just using ztdummy?
is there a limit on conference number
Aby,
Assuming you're building Asterisk from source, you can change the
following in the scan_thread function:
change-
sleep(1);
to-
/*sleep(1);*/
usleep(10);
This will change the delay from 1 second to 10 microseconds (0.1
second).
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED],
altrimenti vi risponderò al mio rientro.
Dimitri Osler
I
Jay R. Ashworth schrieb:
On Thu, Aug 14, 2008 at 03:23:57PM -0500, Tilghman Lesher wrote:
On Thursday 14 August 2008 13:59:37 Philipp Kempgen wrote:
Jared Smith schrieb:
On Thu, 2008-08-14 at 20:35 +0200, Philipp Kempgen wrote:
Whenever something spits out lines with a different
Hello,
I have recently setup my first PBX and am wondering if there might be a
way to send audible notification to the cisco 7960 phone when a call is
put on hold. We lost a call due to a customer being on hold and
forgotten about (yikes). Is there a way to get the phone to beep or ring
down the
On Sat, Aug 16, 2008 at 12:52:33AM +0800, Nhadie wrote:
Hi,
Just wondering if i can use a card just for timing instead of just using
ztdummy? maybe use 2 port fxs or fxo card so it would not be costly.
Yes, it should work well.
will there be any difference on the meetme performance if
On Fri, Aug 15, 2008 at 12:43:39PM -0400, Jerry Geis wrote:
In the past there was ztdummy - what is the new equivalent in dahdi?
You can start with the README. And it's called dahdi_dummy
Also it used to be Zap/X what is the new channel name?
That's part of Asteirsk. Not of Zaptel/DAHDI
Anyone know where I can get an incoming DID for Bogota, Colombia?
Fred Posner
[EMAIL PROTECTED]
Tel: +1 (212) 937-7844 x501
Fax: +1 (954) 252-4187
www.teamforrest.com
smime.p7s
Description: S/MIME cryptographic signature
___
-- Bandwidth and
I may have to do some work with TBCT, and probably cross-carrier TBCT,
here shortly, and I haven't ever worked with it. If anyone on the list
ever has, I'd be interested to know:
1) Only the carrier first involved with the call has to
actually be provisioned for it, correct?
Well lets go by parts:
In regards to your question about the CPU processor usage I am
evaluating the scenario posted by the customer which is not at all the
generic one you are talking about.
He wants STRESS CALL TEST which is SIP to SIP based on the software he
is trying to use:
On Fri, Aug 15, 2008 at 12:52 PM, Nhadie [EMAIL PROTECTED] wrote:
Hi,
Just wondering if i can use a card just for timing instead of just using
ztdummy? maybe use 2 port fxs or fxo card so it would not be costly.
will there be any difference on the meetme performance if i used a
hardware
Hi Atis:
Based on your experience.
How many calls can be handled by a single Pentium 3.0Ghz processor on a
2GB RAM machine spining a 10Krpm disk?
Thanks for the repply, great tool.
Atis Lezdins wrote:
On Fri, Aug 15, 2008 at 8:31 AM, aby azid [EMAIL PROTECTED] wrote:
Hi everyone,
I'm
I took it out of the field name used by FreePBX on the database.
If you install FreePBX you can see it on the mysql table list.
Saul
Todd Fulton wrote:
hi saul,
could you give me more info on the VMX-CONTEXT concept? i tried to
google it, but could find nothing.
i am trying to do
I have worked with a company out of Virginia, IPCOMMS www.ipcomms.net,
Donald Handsil, you may want to shoot him an email, quality good,
experience with asterisk and DID from all over the country.
[EMAIL PROTECTED]
Kind regards,
Saul Bejarano
Dean Collins wrote:
I’ve just started consulting
I got the file for 1999 in space separated value with the NPA, NXX,
Lat-Long , State and City information, if you want to download it I
placed it in my ftp server.
http://www.procomm100.com/files/npanxx99.txt
Kind regards,
Saul
Anthony Messina wrote:
On Thursday 14 August 2008 03:09:42 pm
1. The carrier you are connected to must be licensed for it and have the
necessary software, if the carrier requires, your circuit(s) must be
provisioned for it. The originating/destination carriers shouldn't matter.
2. Both incoming and outgoing calls can be transferred to a second outgoing
The term DIE is not correct sorry, it will be better say CRASH if that
works for you, if you stress call an asterisk the moment you reach the
treshold of that stress test the application will stop responding,
initially it will start popping errors on the log then it will just stop
responding
Dear All,
Why everybody charge so much for this information, why this information
could not be free ?
Thanks
-- Forwarded message --
From: Saul Bejarano [EMAIL PROTECTED]
Date: Fri, Aug 15, 2008 at 3:04 PM
Subject: Re: [asterisk-users] USA Lata AreaCode Database
To:
On 8/15/08, Don Kelly [EMAIL PROTECTED] wrote:
1. The carrier you are connected to must be licensed for it and have the
necessary software, if the carrier requires, your circuit(s) must be
provisioned for it. The originating/destination carriers shouldn't matter.
Most carrier sales people
On Fri, August 15, 2008 2:31 pm, roberto wrote:
Why everybody charge so much for this information, why this information
could not be free ?
Welcome to incumbent communication industry in the US - and, to a some
extent, pretty much anywhere. Everything is ridiculously expensive,
viciously
On Fri, Aug 15, 2008 at 02:37:46PM -0400, Matt Florell wrote:
Most carrier sales people don't know what TBCT is unfortunately, and
even if a carrier is capable of doing it, it is a possiblity that not
all of their equipment is capable of doing it. One client of mine
tried to get TBCT working
Hi list!
The topic-line already describes it all. I´m in the process of setting up
Asterisk with mISDN.
versions used: Asterisk: 1.4.21.2, Asterisk GUI: 2.0 as obtained from DIGIUM
just today, mISDN: 1.1.7.2, OS: Debian 4.01r3 (Kernel 2.6.18 ).
With the GUI I can not manage to set up mISDN
I have two Asterisk 1.2 boxes across a WAN. Calls between them are sent
via SIP g729a. The issue is that the original calleridnum is
overwritten by the value of the fromuser parameter in sip.conf on the
originating server. Is there any way to preserve the original
calleridnum value?
Now this, even though not on the bug tracker, is a great example of a
bug report.
Because of that error code, I was able to easily fix the bug.
Fixed in revision 3671
-bk
Klaus Ruebsam wrote:
Hi list!
The topic-line already describes it all. I´m in the process of setting
up Asterisk
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED],
altrimenti vi risponderò al mio rientro.
Dimitri Osler
I
Russell Bryant wrote:
Lee Lundrigan wrote:
Hi everyone,
Are there any incompatibility issues between asterisk and the c-client
using SSL?
When I enable SSL I get the error:
*pbx.c:1832 pbx_extension_helper: No application 'VoiceMailMain'
*whenever I am trying to access voicemail.
On Friday 15 August 2008 13:45:11 Jay R. Ashworth wrote:
On Fri, Aug 15, 2008 at 02:37:46PM -0400, Matt Florell wrote:
Most carrier sales people don't know what TBCT is unfortunately, and
even if a carrier is capable of doing it, it is a possiblity that not
all of their equipment is capable
Nevermind, I just answered my own question. Used username instead of
fromuser.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Robins
Sent: Friday, August 15, 2008 3:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP Callerid
Tilghman Lesher wrote:
On Friday 15 August 2008 13:45:11 Jay R. Ashworth wrote:
On Fri, Aug 15, 2008 at 02:37:46PM -0400, Matt Florell wrote:
Most carrier sales people don't know what TBCT is unfortunately, and
even if a carrier is capable of doing it, it is a possiblity that not
all of their
Adam Robins schrieb:
I have two Asterisk 1.2 boxes across a WAN. Calls between them are sent
via SIP g729a. The issue is that the original calleridnum is
overwritten by the value of the fromuser parameter in sip.conf on the
originating server. Is there any way to preserve the original
This links: http://downloads.digium.com/pub/telephony/dahdi-linux-complete/
appear broken. thy just take me back to /pub
nothing downloads.
Jerry
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 -
Works for me, try again.
-bk
Jerry Geis wrote:
This links: http://downloads.digium.com/pub/telephony/dahdi-linux-complete/
appear broken. thy just take me back to /pub
nothing downloads.
Jerry
___
-- Bandwidth and Colocation Provided by
I am having a difficulty with AGI and PHP. The following is my script
agi file
#!/usr/bin/php
?php
include(phpagi.php);
$agi = new AGI;
$agi-verbose($argv[1]\n);
?
However, my Asterisk CLI returns this
AGI Tx agi_request: cid-to-acct.php
AGI Tx agi_channel: SIP/5073-0821eda0
AGI Tx
Bk,
Thanks a lot for this immediate fix. mISDN trunk config now works like a
charm!
Best regards,
Klaus
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von bkruse
Gesendet: Freitag, 15. August 2008 21:43
An: Asterisk Users Mailing List -
bkruse schrieb:
Works for me, try again.
Jerry Geis wrote:
This links: http://downloads.digium.com/pub/telephony/dahdi-linux-complete/
appear broken. thy just take me back to /pub
nothing downloads.
downloads.digium.com features redirecting links via a shady
marketing company called Eloqua.
Tzafrir Cohen tzafrir.cohen at xorcom.com writes:
On Thu, Aug 14, 2008 at 04:06:27PM +, Vadim Lebedev wrote:
Hello
I'm looking for a wayy to modify extensions.conf
It seems that PutConfig AMI command is not
supposed to work on extensionsq.conf
It should. Do you have a test
Tzafrir Cohen wrote:
That's part of Asteirsk. Not of Zaptel/DAHDI itself . Asterisk has
renamed chan_zap.so to chan_dahdi.so . It now supports DAHDI . It also
supports Zap/ for the moment for backward compatibility .
And the relevant information is in the Zaptel-to-DAHDI.txt file,
extensions.conf
[To_Airspring]
exten = 55,1,Playback(demo-echotest) ; Let them know what's going on
exten = 55,2,Echo ; Do the echo test
exten = 55,3,Playback(demo-echodone) ; Let them know it's over
exten = 100,1,Dial(SIP/100,20)
sip.conf
;; twinkle softphone
[100]
user=100
nat=yes
This what they sent me
You need to send:
- 11-digit originating # (i.e., 1-NPA-NXX-)
- 10-digit terminating #
This got me a lot further in extensions.conf
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
I am getting a 503 error on the phone and asterisk is giving me:
== Auto
I know what you mean, however it's all about business.
Why to give it free?
Saul
roberto wrote:
Dear All,
Why everybody charge so much for this information, why this information
could not be free ?
Thanks
-- Forwarded message --
From: *Saul Bejarano* [EMAIL PROTECTED]
Vadim Lebedev vadim at mbdsys.com writes:
Tzafrir Cohen tzafrir.cohen at xorcom.com writes:
On Thu, Aug 14, 2008 at 04:06:27PM +, Vadim Lebedev wrote:
Hello
I'm looking for a wayy to modify extensions.conf
It seems that PutConfig AMI command is not
supposed to work
I get congestion (same error) with
exten = _NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
not dialing 1
exten = _1NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
dialing 1
exten = _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
dialing 9
All the same
== Parsing
Tzafrir Cohen wrote:
/ That's part of Asteirsk. Not of Zaptel/DAHDI itself . Asterisk has
// renamed chan_zap.so to chan_dahdi.so . It now supports DAHDI . It also
// supports Zap/ for the moment for backward compatibility .
/
And the relevant information is in the Zaptel-to-DAHDI.txt file,
I figure it out, asterisk is using the wrong ip address.
I have bind address set to the correct ip address. How to I force asterisk to
use the correct ip address?
--- On Fri, 8/15/08, Brad [EMAIL PROTECTED] wrote:
From: Brad [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Basic outbound
On Fri, Aug 15, 2008 at 02:49:17PM -0500, Tilghman Lesher wrote:
To be more clear, what I'm after is to have *someone else besides me*
place calls out their PRI, and then TBCT those placed calls to my DN.
By the time the calls get to me, they should just be standard phone
calls.
So I
On Fri, Aug 15, 2008 at 03:03:23PM -0500, Matthew Fredrickson wrote:
Under no circumstances can Asterisk receive a TBCT request. We just
ignore them. We can initiate them however.
There are different TBCT implementations, dependent on which switch type
is used, with different
On Fri, Aug 15, 2008 at 07:08:33PM +0200, Philipp Kempgen wrote:
True. But lines of varying length with a black background in a
white terminal window don't make it any better. Just causes the
ragged margins to stand out.
Run it into a file with screen(1l) and use less -r to watch it. That's
thank you. would it not limit the number of conference? or maybe the
number of user in a conference?
regards,
ron
Tzafrir Cohen wrote:
On Sat, Aug 16, 2008 at 12:52:33AM +0800, Nhadie wrote:
Hi,
Just wondering if i can use a card just for timing instead of just using
ztdummy? maybe use 2
On Friday 15 August 2008 22:16:34 Jay R. Ashworth wrote:
On Fri, Aug 15, 2008 at 02:49:17PM -0500, Tilghman Lesher wrote:
To be more clear, what I'm after is to have *someone else besides me*
place calls out their PRI, and then TBCT those placed calls to my DN.
By the time the calls
I used to run an Asterisk server in the office, ... was looking for a
small replacement. I am not sure if that one is a good idea yet either.
How about this one:
I have VoIP phones, I have a Welgate 3804 (=2 FXO), all what I need is
an Asterisk server.
Is there a Asterisk hoster out there?
On Friday 15 August 2008 22:31:18 Jay R. Ashworth wrote:
On Fri, Aug 15, 2008 at 07:08:33PM +0200, Philipp Kempgen wrote:
True. But lines of varying length with a black background in a
white terminal window don't make it any better. Just causes the
ragged margins to stand out.
Run it into
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED],
altrimenti vi risponderò al mio rientro.
Dimitri Osler
I
On Sat, Aug 16, 2008 at 01:46:54AM +, Vadim Lebedev wrote:
Vadim Lebedev vadim at mbdsys.com writes:
Tzafrir Cohen tzafrir.cohen at xorcom.com writes:
On Thu, Aug 14, 2008 at 04:06:27PM +, Vadim Lebedev wrote:
Hello
I'm looking for a wayy to modify
Why everybody charge so much for this information,
why this information could not be free ?
Probably because Telecordia makes a decent amount of money from selling it.
If they gave it away that revenue stream would dry up.
John
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