, record-enable|75002|OUT|) in new stack
-- Executing [...@macro-record-enable:1] GotoIf(SIP/75002-b7705298,
1?check) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [...@macro-record-enable:4] AGI(SIP/75002-b7705298,
recordingcheck|20100326-141638|1269584198.62) in new stack
To celebrate three years of the VoIP Users Conference, we're doing a
24-hour VoIP conference call today.
Details are at http://voipathon.org
IRC: #vuc on Freenode.net
SIP: voipat...@vuc.onsip.com - Enter 22622# and your PIN# if you have
no PIN you can listen using 1#
iNum - +883 51007 039
-088e7938, 1?check) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [...@macro-record-enable:4]
AGI(SIP/192.168.0.151-088e7938,
recordingcheck|20100326-101436|1269569676.20) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck
I tried to get in and it said the code was not recognized?
Randy R randulo2...@gmail.com wrote:
To celebrate three years of the VoIP Users Conference, we're doing a
24-hour VoIP conference call today.
Details are at http://voipathon.org
IRC: #vuc on Freenode.net
SIP:
Dear sir,
Thanks for your reply.
our memory size is 4GB.
concurrent calls no : 30.
Our memory condition is below :
Cpu(s): 0.3%us, 0.7%sy, 0.0%ni, 98.5%id, 0.0%wa, 0.1%hi, 0.3%si,
0.0%st
Mem: 4147888k total, 3986540k used, 161348k free,76852k buffers
Swap: 2031608k total,
Hello Platt,
Thank you for help.
I have tested and it works fine.
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New to
Hi List,
I'm finding a solution to provide failover and load balancing features to my
IVR system.
Anyone suggest me what is the best solution please?. what the hardware I should
use ?.
I heard about RedFone, but someone on the mail list said that it is not good
because TDMoE module in
About two years ago I setup two high availability solutions using DRBD and
Heartbeat. The worked great and shutting down or unplugging one server
stayed transparent for the callers, as IVRs stayed available. Having said
this, it was not very straight forward to set it up, but not very difficut
Hi,
My SIP service provider terminates calls in meetme in my Asterisk PBX
and am getting delay on those channels. I found following link to
measure delay in meetme and to decrease it eventually.
http://lists.digium.com/pipermail/asterisk-dev/2005-August/014958.html
It says, enable USE_RTC for
Hi all
my asterisk server, 2 sip client softphones are the same LAN
asterisk ip address : 192.168.1.5
sip client 1 : 192.168.1.4
sip client 2 : 192.168.1.2
asterisk starts ok with sip
setup the sip.conf
[test]
type=friend
username=test
secret=1000
host=dynamic
context=cucku
directmedia=yes
Hi Zeeshan
I know a solution using DRBD, Heartbeat and RedFone hardware to provide
failover ability to Asterisk.
If I have two Asterisk Servers, and each server has a TDM card and a PRI line
connect to each card, how your solution can provide failover ability to
Asterisk ? Do you need any
Hey, is there any Diguim distributor in Lahore,Pakistan? I need to buy X100P.
Muhammad Faheem
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HEllo
try this http://www.voip-info.org/wiki/view/Digium
On Fri, Mar 26, 2010 at 3:29 PM, Faheem faheem_...@yahoo.com wrote:
Hey, is there any Diguim distributor in Lahore,Pakistan? I need to buy
X100P.
Muhammad Faheem
--
Unfortunately not. DRBD and Heartbeat solution is good for pure VoIP.
On 2010-03-26 6:33 AM, huu giang huugiang...@yahoo.com wrote:
Hi Zeeshan
I know a solution using DRBD, Heartbeat and RedFone hardware to provide
failover ability to Asterisk.
If I have two Asterisk Servers, and each server
I have 1 PRI and 1 EM Wink Circuit.
If I call a non working number and route it through the PRI, I get the
following:
RingingYou have reached a non-working number.
If I call a non working number and route it through the EM Wink
Circuit, I get the following:
NO RINGING... DEAD AIR
A
Hi,
I have a problem with polarity reverse on answer
I use asterisk 1.4.30 linux kernel version 2.6.27 dahdi version 2.2.1 and
analog card is Sangoma a400 with fxo ports
this is my config
On Mar 25, 2010, at 11:26 PM, Jeff Brower wrote:
Jim-
Jim-
There will be up to 150 phones so there will be 300
channels when they are all on the phone at one time.
I will be using a current 1.4 version.
That's a lot of channels for Asterisk... IIRC the TC400B transcoding card
is
25 mar 2010 kl. 13.14 skrev Michelle Dupuis:
I can't find this in the wiki/email history..but I'm sure it's based asked
before.
The port range define in rtp.conf - is that for connections initiated by
asterisk? Or the port range asterisk listens on? Or both?
These are the ports
Digium hasn't sold the X100P for something like 2 years now.
Leif.
Faheem wrote:
Hey, is there any Diguim distributor in Lahore,Pakistan? I need to buy
X100P.
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Hello List.
I am having problems retreiving voicemails on my system. I noticed when someone
leaves a message through the pstn line I can't hear anything. I tested leaving
a message from one of the extensions and that can be heard. I don't know if is
the type of card I'm using for analog (
Can you post your /etc/zaptel.conf and /etc/asterisk/zapata.conf.
--
Zeeshan
On 2010-03-26 9:06 AM, Robert Grignon rgrig...@fleetone.com wrote:
I have 1 PRI and 1 EM Wink Circuit.
If I call a non working number and route it through the PRI, I get the
following:
RingingYou have reached a
Hi again,
In other asterisk it happened the same... No internet, no justvoip
resolution, no sip...
Remove the trunk, sip up... I'm going to test using bind with a local
zone.
More ideas/suggestions?
Regards
Luis Silva
Hi ,
I had some problems in the past with sip trunks, asterisk-users
Hi Philip,
So i looked at the codecs in the device (polycom) it says only G.711 and
ulaw can be used, i made an internal call using two phones that are
configured just with sip (so IAX not involved) but the static noise is
there, i typed show sip peer username and this is the only thing i got:
If I have two Asterisk Servers, and each server has a TDM card and a
PRI line connect to each card, how your solution can provide failover
ability to Asterisk ? Do you need any other hardware?
Have a look at this article and how they shared a single T1 line across
two servers for failover:
so doesn't looks like overload
Could it be a problem with the firmware of your softphones? Have you been
using some new phones lately? someone else in a different thread pointed on
attended transfer bugs with SNOM phones.
We are eagerly waiting for your solution.
Hope we can help but don't so
Michelle Dupuis wrote:
I can't find this in the wiki/email history..but I'm sure it's based
asked before.
The port range define in rtp.conf - is that for connections initiated by
asterisk? Or the port range asterisk listens on? Or both?
Thanks!
MD
The port range specified is
Just to check, have you set up
srvlookup=yes
under the general context in your sip.conf?
Alyed
2010/3/26 Luis Silva luis.si...@dreamware.pt
Hi again,
In other asterisk it happened the same... No internet, no justvoip
resolution, no sip...
Remove the trunk, sip up... I'm going to test
Olle E. Johansson wrote:
25 mar 2010 kl. 13.14 skrev Michelle Dupuis:
I can't find this in the wiki/email history..but I'm sure it's based asked
before.
The port range define in rtp.conf - is that for connections initiated by
asterisk? Or the port range asterisk listens on? Or both?
I guess to do what you want you need to dial directly between the phones.
Can't do it with xlite but you can with SJphones
Don't remember the exact syntax but guess it's something like
sip:usern...@the.phones.ip:5060
Alyed
2010/3/26 haloha haloha...@gmail.com
Hi all
my asterisk server, 2
chan_dahdi.conf:
=
;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
;autogenrated on 2009-12-04
;Dahdi Channels Configurations
;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak
[trunkgroups]
[channels]
context=default
Date: Fri, 26 Mar 2010 00:30:50 +0200
From: tzafrir.co...@xorcom.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] new server install errors starting asterisk
On Thu, Mar 25, 2010 at 09:58:17PM +, Ott Rose wrote:
well here is what i did to solve it but i
Just to check, have you set up
srvlookup=yes
under the general context in your sip.conf?
Alyed
No, but I put it now but the result is the same. And googleing further
https://issues.asterisk.org/view.php?id=3723, it seems that is an old
issue...
Don't know for witch version is, 1.2?... But is
Just for the sake of this thread I'll paste part of the last post regarding
this issue in the asterisk bug tracker.
kpfleming on 2005-03-10 post: Essentially, what we are saying is that if
you are going to use DNS to resolve critical information in your Asterisk
configuration, you need to do
Le 26/03/2010 15:01, Landy Landy a écrit :
Hello List.
I am having problems retreiving voicemails on my system. I noticed when
someone leaves a message through the pstn line I can't hear anything. I
tested leaving a message from one of the extensions and that can be heard. I
don't know if
I get this when my brother in law tries to call in from his box to mine.
WARNING[4855]: chan_sip.c:12675 check_auth: username mismatch, have
100, digest has s
or after changing the register line:
WARNING[4855]: chan_sip.c:12675 check_auth: username mismatch, have
100, digest has 199
I have
i have posted this question couple of times and never really got any hits i
wasn't able to provide any debug info
Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid = 3309)
Verbosity is at least 4
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using
Hi!
it should be some commands that can give me a better idea about the
codecs, if anyone know them, please help!
Use sip show channels and iax show channels and look at the Format
column.
About the Polycom devices: Others will have to help you there. I have no
good guess why you might
Seems like an Amportal configration problem not and Asterisk issue. Maybe
you should try in one of the FreePBX users list.
Alyed
2010/3/26 Ott Rose sixfourimp...@hotmail.com
i have posted this question couple of times and never really got any hits
i wasn't able to provide any debug info
James Lamanna wrote:
Hi,
Does anyone have any good empirical data suggesting what the maximum
number of PRI calls (incoming and outgoing)
without hardware echo cancellation can be handled on a single box is?
I have a TE410P T1 (1st gen) card and I'm seeing interesting errors of
D-Channels
I've seen this before and I know the reason but not really the solution:
You have used this same username/password combination for another SIP
client, or maybe the same one but with different IP. Even when that one is
offline from some time on, Asterisk doesn't renew it's internal database, so
At 05:47 PM 3/26/2010, you wrote:
You have used this same username/password combination for another
SIP client, or maybe the same one but with different IP. Even when
that one is offline from some time on, Asterisk doesn't renew it's
internal database, so still thinks it might be somewhere
Hi Alyed
xilte softphone work perfectly on other sip server(opensips server)
Don't remember the exact syntax but guess it's something like
sip:usern...@the.phones.ip:
5060
you mean i config the extension.conf look like exten =
1000,1,Dial(SIP/1...@ip address:5060), is it right?
the problem
If your sofphones are registering to the asterisk, then asterisk needs to be
in the middle, otherwise there's no way your 101 sofpthone user can actually
know where (by where I mean which IP) is the 102 softphone user.
UNLESS (yes, there's a big unless) you dial from 101 DIRECTLY to 102. How?
Hi Alyed
so the asterisk is in middle in all version, right? thank you for your
explanation
all devices i mean are asterisk + softphones
my goal is asterisk is on internet - WAN IP address and the softphones are
in NAT but the xlite supports the ICE function that is why i ask the media
should be
so the asterisk is in middle in all version, right? thank you for your
explanation
is the one whom everyone goes and says hey I'm 101 and live downstairs can
I play with you guys?
my goal is asterisk is on internet - WAN IP address and the softphones are
in NAT but the xlite supports the ICE
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