Hi,I am looking for a reliable provider that can provide 3 dedicated linux
servers asap.
Unfortunately, the provider I have used for YEARS has become way too slack
in recent times and we have to move on.
Cheers,
Sahil
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Hi All,
I am using asterisk 1.4.22 in my local system
I want to know how can we set ability to log and report RTP and jitter
statistics per call.
Is there any configuration in logger or configuration in rtp?
Please provide some guide lines for this.
Thanks in advance!
Thanks,
Max Alex
Voip
Right now I'm currently working with Sagonet www.sagonet.com. They've been
good for me for a good while now on servers and uptime.
Tom
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sahil Gupta
Sent: Friday, November 14, 2008 3:46 AM
To: Asterisk Users Mailing List
Positively Optimistic wrote:
exten = 3,1,ParkAndAnnounce(|45|SIP/[EMAIL PROTECTED]
mailto:%7C45%7CSIP/[EMAIL PROTECTED]|)
Has anyone else had any success with this application? We have a sip
phone at exten 7607 with the sip add header function allowing for auto
answer and paging...
In article [EMAIL PROTECTED],
Mikel Lindsaar [EMAIL PROTECTED] wrote:
I have the following working (somewhat) setup:
TELCO
|
|
E1 (30 Chan -- TE210 SPAN 2)
|
|
Asterisk box 1.6 with
DAHDI drivers loaded
Digium TE210p
|
|
E1 (30 Chan -- TE210 SPAN 1)
Dear All
I installed a Digium card TE405P with zaptel and its running successfully
with no alarms, but asterisk is not running .
Any one have a cure or advice
03:09.0 Communication controller: Digium, Inc. Wildcard TE405P quad-span
T1/E1/J1 card 5.0V (rev 02)
Nov 14 07:56:58 localhost
I am hearing a you have been kicked from the conference message in
asterisk 1.2.23.
I dont want to hear that.
I am using 1qt for the meetme.
How can I disable that message?
THanks
Jerry
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On Fri, Nov 14, 2008 at 7:59 AM, Khaled Chehab [EMAIL PROTECTED] wrote:
Dear All
I installed a Digium card TE405P with zaptel and its running successfully
with no alarms, but asterisk is not running .
Any one have a cure or advice
03:09.0 Communication controller: Digium, Inc.
On Fri, Nov 14, 2008 at 6:29 PM, Khaled Chehab [EMAIL PROTECTED] wrote:
Dear All
I installed a Digium card TE405P with zaptel and its running successfully
with no alarms, but asterisk is not running .
Any one have a cure or advice
03:09.0 Communication controller: Digium, Inc.
Dear All
I tried to stop asterisk and start it with debugging ,kindly heck the
results
Already I modified asterisk.conf
astrundir = /var/run/asterisk
[EMAIL PROTECTED] ~]# safe_asterisk
[EMAIL PROTECTED] ~]# Asterisk ended with exit status 0
Asterisk shutdown normally.
[EMAIL
I used to use IDEFISK, but since it was taken over/renamed into Zoiper
it's been really hard work - now I'm told that they won't support my
chosen distribution - Debian Etch - the current stable version of Debian I
prefer.
So, looking to dump Zoiper and go with something else - I want
Gordon Henderson wrote:
I used to use IDEFISK, but since it was taken over/renamed into Zoiper
it's been really hard work - now I'm told that they won't support my
chosen distribution - Debian Etch - the current stable version of Debian I
prefer.
So, looking to dump Zoiper and go with
Start asterisk with asterisk -v and see where it is bombing out.
You could also check your Asterisk logs
Usually, some module is not loading or whatever, so you can either figure
out why or add noload=module in modules.conf.
If it is not a module, it will be pretty clear what it is when
http://blog.voipsupply.com/new-products/free-sip-softphone-roundup
Cory J. Andrews
Director New Market Initiatives
Sayers Media Group
VoIP Supply, LLC
454 Sonwil Drive
Buffalo, NY 14225
716-250-3402 OFFICE
716-630-1548 FAX
716-601-4474 MOBILE
[EMAIL PROTECTED]
Have I exceeded your
Dears
My hitch is that no alerts bombing out .all what is bombing out
Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or
directory
My modules.conf
;
; Asterisk configuration file
;
; Module Loader configuration file
;
[modules]
autoload=yes
noload =
In article [EMAIL PROTECTED],
Jerry Geis [EMAIL PROTECTED] wrote:
I am hearing a you have been kicked from the conference message in
asterisk 1.2.23.
I dont want to hear that.
I am using 1qt for the meetme.
How can I disable that message?
There isn't a way in meetme. However, if you
In article [EMAIL PROTECTED],
Khaled Chehab [EMAIL PROTECTED] wrote:
Dear All
I tried to stop asterisk and start it with debugging ,kindly heck the
results
Hi Khaled,
There is not nearly enough information in your message for anyone to
suggest what might be wrong.
Please have a look
This works for us
exten = h,1,Set(CDR(userfield)=${RTPAUDIOQOS})
exten = h,2,Hangup()
results in
Set(SIP/rpx2399a-b61fc5e0,
CDR(userfield)=ssrc=213416392;themssrc=0;lp=0;rxjitter=0.00;rxcount=0;txjitter=0.00;txcount=0;rlp=0;rtt=0.00)
*From:* [EMAIL PROTECTED]
Cory Andrews wrote:
http://blog.voipsupply.com/new-products/free-sip-softphone-roundup
Good roundup, thanks.
Alan
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On Fri, Nov 14, 2008 at 9:27 AM, Tony Mountifield
[EMAIL PROTECTED]wrote:
In article [EMAIL PROTECTED],
Khaled Chehab [EMAIL PROTECTED] wrote:
Dear All
I tried to stop asterisk and start it with debugging ,kindly heck the
results
Hi Khaled,
There is not nearly enough
On Fri, Nov 14, 2008 at 02:02:14PM +, Gordon Henderson wrote:
I used to use IDEFISK, but since it was taken over/renamed into Zoiper
it's been really hard work - now I'm told that they won't support my
chosen distribution - Debian Etch - the current stable version of Debian I
prefer.
I found this property in queue.conf
; Calls may be recorded using Asterisk's monitor resource
; This can be enabled from within the Queue application, starting recording
; when the call is actually picked up; thus, only successful calls are
; recorded, and you are not recording while people
On Fri, 14 Nov 2008, Tzafrir Cohen wrote:
On Fri, Nov 14, 2008 at 02:02:14PM +, Gordon Henderson wrote:
I used to use IDEFISK, but since it was taken over/renamed into Zoiper
it's been really hard work - now I'm told that they won't support my
chosen distribution - Debian Etch - the
On Fri, Nov 14, 2008 at 03:19:22PM +, Gordon Henderson wrote:
On Fri, 14 Nov 2008, Tzafrir Cohen wrote:
On Fri, Nov 14, 2008 at 02:02:14PM +, Gordon Henderson wrote:
I used to use IDEFISK, but since it was taken over/renamed into Zoiper
it's been really hard work - now I'm told
Hi list,
is it possible to get in the running dialplan the status of (SIP) lines
without using AGI or anything like that? What I want is a stepwise
calling: I have several SIP lines (let's say they are three) which I
want to dial to alternatingly. But I do not want to dial to a already
busy line
How about a call queue using the roundrobin strategy?
http://www.voip-info.org/wiki/view/Asterisk+call+queues
Dave
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christophorus
Laube
Sent: Friday, November 14, 2008 11:29 AM
To:
On Friday 14 November 2008 09:19:22 Gordon Henderson wrote:
On Fri, 14 Nov 2008, Tzafrir Cohen wrote:
On Fri, Nov 14, 2008 at 02:02:14PM +, Gordon Henderson wrote:
I used to use IDEFISK, but since it was taken over/renamed into Zoiper
it's been really hard work - now I'm told that they
On Fri, 14 Nov 2008, Tilghman Lesher wrote:
On Friday 14 November 2008 09:19:22 Gordon Henderson wrote:
On Fri, 14 Nov 2008, Tzafrir Cohen wrote:
On Fri, Nov 14, 2008 at 02:02:14PM +, Gordon Henderson wrote:
I used to use IDEFISK, but since it was taken over/renamed into Zoiper
it's been
On Fri, 14 Nov 2008, Gordon Henderson wrote:
On Fri, 14 Nov 2008, Tilghman Lesher wrote:
On Friday 14 November 2008 09:19:22 Gordon Henderson wrote:
On Fri, 14 Nov 2008, Tzafrir Cohen wrote:
On Fri, Nov 14, 2008 at 02:02:14PM +, Gordon Henderson wrote:
I used to use IDEFISK, but
On Fri, 14 Nov 2008, Jeff LaCoursiere wrote:
On Fri, 14 Nov 2008, Gordon Henderson wrote:
On Fri, 14 Nov 2008, Tilghman Lesher wrote:
On Friday 14 November 2008 09:19:22 Gordon Henderson wrote:
On Fri, 14 Nov 2008, Tzafrir Cohen wrote:
On Fri, Nov 14, 2008 at 02:02:14PM +, Gordon
Hello,
I'm the person responsable for the zoiper roadmap, comments inline
snip
This all started because Zoiper really annoyed me - they keep sending me
beta versions of their software (which is nice, thanks you), and they keep
on compiling it for ubuntu or some other distribution of linux
For a long while, Asterisk administrators have had the desire to take PRI
channels out of service and keep them that way, usually to create a window
in which to perform maintenance or another such purpose. We are finally very
close to having that functionality in Asterisk, but what we need is a
On Fri, Nov 14, 2008 at 7:07 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote:
On Fri, 14 Nov 2008, Gordon Henderson wrote:
On Fri, 14 Nov 2008, Tilghman Lesher wrote:
On Friday 14 November 2008 09:19:22 Gordon Henderson wrote:
On Fri, 14 Nov 2008, Tzafrir Cohen wrote:
On Fri, Nov 14, 2008
Anyone know of where to get a Manilla or Philippines DID? I show (1)
on didx.net but is rated too low to purchase.
Fred Posner
[EMAIL PROTECTED]
Main: +1 (212) 937-7844
Direct: +1 (503) 914-0999
www.teamforrest.com
smime.p7s
Description: S/MIME cryptographic signature
Hello all,
I'm trying to develop a dialer interface from my
application, basically to originate calls on asterisk using the Manager
Interface. During this development I came across a situation and I realized
that the asterisk Originate command could be a little better than it is
Hello list,
Topology diagram first:
[PSTN]-[SPA 400][NETGEAR SWITCH]---[ASTERISK BOX]
| |
|
|__[SPA 921]
On Fri, Nov 14, 2008 at 08:34:48PM +0200, Atis Lezdins wrote:
On Fri, Nov 14, 2008 at 7:07 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote:
On Fri, 14 Nov 2008, Gordon Henderson wrote:
On Fri, 14 Nov 2008, Tilghman Lesher wrote:
On Friday 14 November 2008 09:19:22 Gordon Henderson
Khaled Chehab wrote:
Dears
My hitch is that no alerts bombing out .all what is bombing out
Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file
or directory
check to see if the directory /var/run/asterisk exist and the
permissions are correct if you run asterisk as
On Fri, Nov 14, 2008 at 10:49 PM, Edwin Lam [EMAIL PROTECTED] wrote:
Khaled Chehab wrote:
Dears
My hitch is that no alerts bombing out .all what is bombing out
Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file
or directory
check to see if the directory
Positively Optimistic [EMAIL PROTECTED] writes:
exten = h,1,Set(CDR(userfield)=${RTPAUDIOQOS})
exten = h,2,Hangup()
results in
Set(SIP/rpx2399a-b61fc5e0,
CDR(userfield)=ssrc=213416392;themssrc=0;lp=0;rxjitter=0.00;rxcount=0;txjitter=0.00;txcount=0;rlp=0;rtt=0.00)
Does it
Quick question ... I am interested in installing Asterisk and using SIP or
IAX to connect to another system. If I am not planning to install analog or
digital cards to connect to another system do I need to install zaptel
still?
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Not for SIP, IAX2 may need ztdummy (provided by Zaptel or whatever it is
called in DAHDI, wouldn't know since I don't use it.
I usually throw an empty, usually RMAed or otherwise broken TDM400 card in
an use that for timing when needed. I have plenty of them
Thanks,
Steve Totaro
On Fri,
Hi folks,
I am building a new box. Want it to look
pretty much like an older Asterisk 1.2,
Debian box that is in production. The new
box will used as a test box before we
implement changes to the production box.
New box:
# cat /etc/issue; uname
No data is logged on the call. Probably because the status is reporting
down
*CLI pri show span 1
Primary D-channel: 24
Status: Provisioned, Down, Active
Switchtype: National ISDN
Type: Network
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203
Dear Tony,
Thanks. Found that problem and that now works :)
Now I have a different problem, but different thread for that.
Mikel
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Hello all.
I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box.
NEC - E1 - TE210P:1 - * - TE210P:2 - E1 - Telco
Incomming calls from the telco to the asterisk box to the NEC work fine with
indials and everything. Works sweet.
Outbound from the NEC to the Asterisk box fail.
Hi All,
Thanks for reply
i have tried for this, it looks fine for me,
but is there any way to check rtp log while call is connected or any way to
enable it to write in log file.
Please give me some guide lines!
thanks in advance.
Thanks,
Max Alex
Voip Developer
On Sat, Nov 15, 2008 at 3:21 AM,
See:
http://astrecipes.net/index.php?q=AstRecipes/Music-on-hold%20without%20MPG123
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On Sat, Nov 15, 2008 at 6:47 AM, Max Alex [EMAIL PROTECTED] wrote:
Hi All,
Thanks for reply
i have tried for this, it looks fine for me,
but is there any way to check rtp log while call is connected or any way to
enable it to write in log file.
Please give me some guide lines!
thanks in
On Fri, Nov 14, 2008 at 10:27 PM, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
On Fri, Nov 14, 2008 at 08:34:48PM +0200, Atis Lezdins wrote:
On Fri, Nov 14, 2008 at 7:07 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote:
On Fri, 14 Nov 2008, Gordon Henderson wrote:
On Fri, 14 Nov 2008, Tilghman
Using a Polycom 550 and 650 phones on my Asterisk server for testing. I can't
figure out why the volume is so low. How can I adjust the volume control on
Asterisk? It's at max on the handset phones.
Thanks!
Hin
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