Voipjet locks $1.2 per running call and unlocks when call ends .. so $12 =
10 simultaneous calls ( if rate is 1.2 cents ) .
On 02/02/07, Robert DeVries [EMAIL PROTECTED] wrote:
I have found that if you don't have the minimum balance required for the
voipjet premium server, you get the circuits
config problem . what pbx does ip_pb2 runs ? ( is it asterisk ? ) in peer
definition try allowing all codecs .. ( gsm , ulaw,alaw,ilbc )
On 08/02/07, Florea Igor [EMAIL PROTECTED] wrote:
Hi,
I'm new to *,so i apologize for stupid questions.
I'm having problem with this arhitecture:
I'm calling
You can easily recompile asterisk with mysql logging enabled also use all
add-ons u can use on debian and any other distro ..
On 08/02/07, Chris Earle [EMAIL PROTECTED] wrote:
I'm tempted to rebuild my asterisk network with AsteriskNow - my
question is, can you ADD anything to it? i.e.
check your sip.conf and make sure it has allow=ulaw and allow=alaw line (
you can even remove gsm to test it it works fine or not )
On 09/02/07, Florea Igor [EMAIL PROTECTED] wrote:
ip_pbx2 is not asterisk, it knowk only PCMU,PCMA,g723,g729
On Thursday 08 February 2007 19:00, Vicky wrote
I also encountered the problem of port 5060 being blocked by some user's isp
and redirected port 5098 to 5060 but still asterisk wasnt able to detect
hangup properly and had load of voice problems ( lot of nat involved and
softphones were being used ) so i made asterisk listen on 5098 and
1000 Hz is recommended if you use lot of meetme channels ( and maybe iax
trunking ? ) without a hardware timer .
On 08/02/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Wed, 7 Feb 2007, Mark Coccimiglio wrote:
Ok here is a real geek question,
I building my own linux kernel for my
try changing bindport of asterisk from 5060 to something else .
On 09/03/07, Pezhman Lali [EMAIL PROTECTED] wrote:
Dears
my Internet Provider , prevents , sip connections,
between sip client(sip phone) and sip server,
(asterisk + ser) .
both of client and server are mine.
is there any
I got money back around 6 months ago . It was a via paypal claim and hey
didn't reply till paypal's deadline so i got $30 back .
On 17/03/07, Ira [EMAIL PROTECTED] wrote:
At 02:32 PM 3/16/2007, you wrote:
You were able to cancel service with Sellvoip? That's impressive, that
Actually, it's
Anyone having problems with voxee since last few days or is it just me ? In peek hours i get LAGGED when i do a iax2 show peers or even 1000 ms latency . Most of time it is 20 ms or so but when i start sending traffic to them latency increases to 1000 ms or even LAGGED ( also shows high in peak
] To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, November 09, 2006 9:11 PM
Subject: Re: [asterisk-users] Voxee lag problems ?Hi Vicky, I used to use their termination services, but I had the same problems ... It is impossible to work with that latency ... A lot of gaps
For the time being try putting 212.41.253.181in hostname= line in ur sip config and it should work . Also check if you /etc/resolv.conf has correct dns list ( i guess it does bcoz OS can resolve).Alsocheck/etc/asterisk/dnsmgr.conf.
Here's xample :[general]enable=yes ; enable creation
For the time being try putting 212.41.253.181in hostname= line in ur sipconfig and it should work . Also check if you /etc/resolv.conf has correctdns list ( i guess it does bcoz OS canresolve) . Also check /etc/asterisk/dnsmgr.conf .
Here's example :[general]enable=yes ; enable creation of managed
I doubt how many days more voxee will survive . Its been a month nw and tehir support doesnt want to repair this .
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Yep make the server with dynamic ip register to server with static ip ( sip or iax both will do but in sip keep nat=yes while making extension ) On 12/11/06,
Rosli Sukri [EMAIL PROTECTED] wrote:
u need another box say box a with real/addressable ip address. create an iax entry in box a and have
I think its same as DND (do not disturb ) . It can be activated by *78 and deactivated by *79 . I use freepbx for configuration so i am not sure if its there in default asterisk setup . I snipped some part of my configuration from freepbx's config files
[app-dnd-on]exten = *78,1,Answerexten =
I am having asterisk working with cdr mysql patch and freepbx for configurations . It stores all records in mysql tables and i can do further post paid billing myself . I am looking for a simple system that can show a user live call logs via web
Could this be considered spam ? I believe this is second threas realted to that pbx .On 13/11/06, Jordi Nelissen
[EMAIL PROTECTED] wrote:
Check out the ESCAUX net.PBX operator console. In use in variouscompanies with 200+ extensions. Powerfull and convenient.
I am not sure if it will help but try to put notansfer=yes in ur iax2 extension (just experiment a bit ;) ).On 12 Nov 2006 17:48:13 -0500, joe a. (
[EMAIL PROTECTED]) [EMAIL PROTECTED] wrote:
Experiencing one way audio using IAX2.I did see some other posts on this, and see there may be some
What is the length of music on old mp3 file ? Maybe file is very short .On 13/11/06, zen Perry [EMAIL PROTECTED]
wrote:
Mac OS X, Asterisk 1.4 beta--- Martin Joseph [EMAIL PROTECTED] wrote: On 2006-11-12 23:08:05 -0800, zen Perry
[EMAIL PROTECTED] said: I'm trying to set up the Music on Hold
You can also use waitexten = X,1,Wait(3)(for3secs) On 13/11/06, Jim Archer [EMAIL PROTECTED] wrote:
--On Sunday, November 12, 2006 11:53 PM -0500 John Novack[EMAIL PROTECTED] wrote: Dovid B wrote: snip How hard would it be to have asterisk detect a dial tone ?
I really can't say. I am not a C
Put canreinvite=no in asterisk sip user extension.Someprovidersdonotsupportreinvitesandhenceyougetsilenceiguess. On 13/11/06, Yuri Veremeyenko
[EMAIL PROTECTED] wrote:
Hi, I am using Asterisk to set up a reminder-like system, with asterisk auto-dialing a user via SIP and playing a reminder file
Its pretty easy . If you have mysql records enabled via a patch just do sql queryuse asteriskcdrdb;select * from `cdr` where billsec 0 ( if answered then billsec always greater than 0 or you cna also use disposition = 'ANSWERED' )
On 13/11/06, Olivier [EMAIL PROTECTED] wrote:
Why is it awful
oops sorry i didnt saw quoted text of other user and it showed as first post in gmail draft so i thought u made a topic for that pbx ( so considered spam :P ) . Sorry again :)On 13/11/06,
Jordi Nelissen [EMAIL PROTECTED] wrote:
Vicky,my other post related to a Web GUI for asterisk. This post
This is more of mysql question then asterisk :D . Most voip providers use 6 second rounding for costing . My asterisk server stores call cdr's in mysql properly with billsec field containing number of billed seconds . I want to know some function to round this to 6 seconds ( or any custom valud
Why not directly use ip address in host= lineinextensions instead of dynamic address like sip.voipprovider.com .. temporary fix but it may work .
On 13/11/06, Steve Langstaff [EMAIL PROTECTED] wrote:
A search of google should turn up some recommendations about running alocal cacheing DNS proxy, or
PROTECTED] wrote:
On 11/12/06, nik600 [EMAIL PROTECTED] wrote: On 11/12/06, Vicky [EMAIL PROTECTED] wrote: Yep make the server with dynamic ip register to server with static ip ( sip
or iax both will do but in sip keep nat=yes while making extension ) the problem is that the server with dynamic ip
Does it happen when you make more than one call from you main voip server alone ? Or it happens when there are more than 1 call on your branch server ? Pin the problem is in which server first , If main server can handle 2-3 calls with no lag then its probably problem in branch server .
On
by 6, /60 for minutes On Tue, 2006-11-14 at 00:07 +0530, Vicky wrote:
This is more of mysql question then asterisk :D . Most voip providers use 6 second rounding for costing . My asterisk server stores call cdr's in mysql properly with billsec field containing number of billed seconds . I want to know
6second=billsec+(6-mod(billsec,6) where 6second=0 if you want a decimal minutes column called billmin
UPDATE cdr SET billmin=round((billsec/60)+0.5),1) where billmin=0 Vicky wrote: Thx and what would the sql query be ?.Iplantoputadditionalfieldas6second
actually 1-2 are rtp ports used by asterisk .. its not really compulsary .. you can set a custom range in /etc/asterisk/rtp.conf .. check ur rtp.conf what range its using and open that in firewall . Default with asterisk is 1-2 unless changed .
On 14/11/06, Al Bochter [EMAIL
SERVICES http://www.bochterservices.com/?j=platingt=email
Vicky wrote: actually 1-2 are rtp ports used by asterisk .. its not really compulsary .. you can set a custom range in /etc/asterisk/rtp.conf .. check ur
rtp.conf what range its using and open that in firewall . Default with asterisk
Thereis definitely wrong in your setup . I have ipkall setup on my asterisk and dont have ports 1000-2000 open ( only 1-2,5060,4569 open ) . and incoming calls word fine for me .
On 14/11/06, Al Bochter [EMAIL PROTECTED] wrote:
No 1000 to 2000 is not a typo.Well let me put some light on
Yes asterisk can do that . If you mena for call records then see http://www.voip-info.org/wiki-Asterisk+cdr+mysqlAlso see
http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQLOn 14/11/06, Sharon Lim [EMAIL PROTECTED]
wrote:
Hi there, I am looking around, is there anyone did any integration
oops sorry i thought its my sql didnt notice it's MS SQL :D On 14/11/06, Tony Mountifield [EMAIL PROTECTED]
wrote:
In article [EMAIL PROTECTED],Sharon Lim
[EMAIL PROTECTED] wrote: Hi there, I am looking around, is there anyone did any integration asterisk talk to / connect to MS SQL?Look for the
just dont enter any email address while creating extension / mailbox ;)On 14/11/06, Ma Zhiyong [EMAIL PROTECTED]
wrote:
HI, allCan I disable send e-mail feature in the voicemail application?___--Bandwidth and Colocation provided by Easynews.com --
. Port forwarding should help here .Also edit sip_nat.conf after port forwarding but it will be a burden to setup if asteriskisondynamicip.
On 14/11/06, nik600 [EMAIL PROTECTED] wrote:
On 11/13/06, Vicky [EMAIL PROTECTED] wrote: IF your asterisk server is behind NAT and no port forwarding is done
its normal .if there are many calls going . You should worry if your load or memory usage is very high .On 16/11/06, Andre Courchesne - Consultant
[EMAIL PROTECTED] wrote:
We have 1 server that after a few hours operating has multiple processof asterisk running. Here is the pstree output:#
try :
[John]
type=friend
secret=test
host=dynamic
disallow=all
allow =gsmilbculawalaw
Also try other sip phone slike sjphone just to make sure there is no prob .
On 16/11/06, Charlie Grosvenor [EMAIL PROTECTED] wrote:
I have just installed Asterisk and installed the sample configuration
files.
Please go to bugs.digium.com and file this bug they will difinitely get it
working .
On 16/11/06, Thirumal Saminathan [EMAIL PROTECTED] wrote:
hi,
exten =6000,1,dial(SIP/6000,15,tr)
exten =6002,1,dial(SIP/6002,15,tr)
exten =6004,1,dial(SIP/6004,15,tr)
exten =6006,1,dial(SIP/6006,15,tr)
g729 is not a free codec . YOu have to buy it from digium at rateof $10 per
channel license . If you are just using asterisk and havent bought g729
license then asterisk will just do bridging of g729 and wont edit/transcode
stream .
On 16/11/06, Victor Toofic [EMAIL PROTECTED] wrote:
I have
I have call recording enabled for some extensions and they make lot of calls
. I see there are many files in /var/spool/asterisk/monitor but i need to
know which file belongs to which call .. In old version of asterisk this was
available in lastapp field of mysql table . Now lastapp shows
a las 18:28 +0530, Vicky comentaba:
g729 is not a free codec . YOu have to buy it from digium at rateof $10
per
channel license . If you are just using asterisk and havent bought g729
license then asterisk will just do bridging of g729 and wont
edit/transcode
stream .
On 16/11/06, Victor
I am also searching one for post-paid billing .. but most like astpp wants
to eat whole system themselves managing extensions and all . I need a type
of solution that can just bill people based on mysql cdr using accountcode
and amagflags .. I am thinking to make some myself now but it will take
of asterisk requirements, is to make sure that the cdrs in the
database have an accountcode set. You do not need to use it to manage
your dids and extensions, etc.
Darren Wiebe
[EMAIL PROTECTED]
Vicky wrote:
I am also searching one for post-paid billing .. but most like astpp
wants to eat whole
I am not sure if i understood what you mean but yes asterisk cdr's can be
used for billing with some modifications of your own. Asterisk can make cdr
in csv,mysql,postgresql with complete call
info which can be used for billing system's .
On 19/11/06, Noc Phibee [EMAIL PROTECTED] wrote:
Hi
I
I am using asterisk to receive call from a DID provider . In configured
everything in freepbx properly and its working . I forwarded incoming calls
from did to a certain extension . Now i tried calling from another sip
provider to this box , when i call from other provider to my DID number
then
I doubt that . I think qualify=500 means asterisk checks every 500 ms if the
other extension is available or not . Because when qualify=( value in ms )
is set and you do a sip show peers in console asterisk whos how much latency
is there between extension and asterisk . If i set qualify = no then
Yeh even a
simple UDMA 5 enabled hard drive can handle 30 calls recording easily .
Sata hard drives are even better .
On 22/11/06, Marcus Franke [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
[EMAIL PROTECTED] wrote:
Does anyone have experience with recording
On 22/11/06, Marcus Franke [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Vicky wrote:
Yeh even a
simple UDMA 5 enabled hard drive can handle 30 calls recording easily .
Sata hard drives are even better .
Hehe, UDMA sounds like EIDE drives.. nice to see
If canreinvite=yes is specified in sip.conf for 2 sip extensions and call
recording is disabled in asterisk, both legs have same codec . Doesit
always does native bridging . I am
using freepbx . How can i know if a call is going through asterisk or
they are bridged directly to each other ? Does
I am looking for a toll-free US 1800
DID which can be setup quickly . I have seen nufone there quality is
very good but they charge for 30 seconds minimum ( others do 6/6 i
guess
) . east coast gateway
server preffered . . Plz lemme know if you have some suggestions i
want it to be setup very
I am using asterisk along with freepbx . When recording is enabled for a
extension the call record file made in /var/spool/asterisk/monitor contains
information like OUT(extension number)-(timestamp)-(uniqueid).wav . This can
be a big mess if there are more than 1000-2000 files in that folder and
Asterisk wont sit in media path if both callee and caller agrees on common
codec, both have canreinvite=yes in sip.conf, no t,T are used in dialplan (
please correct me if i am wrong ) , no call recording is enabled .
I think asterisk does native bridging even if one is behind nat ( i tested
heres my scenariosoftphone-Asterisk( outgoing call recording
)Call Provider
I am recording all outgoing calls on asterisk so its obvious that there is
no native bridging . Suppose if i am using gsm from softphone--asterisk and
then what codec should i prefer for asterisk-
I am planning to put up a asterisk server with around 50-60 phones over a
lan . I am planning on keeping a decent server ( for outbound pstn ) and all
phones connected via linksys pap2 ( all 60 phones as pap2 registering to
asterisk) . Does this kind of setup give problem ?
I am not sure but i think that fix is for compiling zaptel not asterisk .
Asterisk runs on centos with 0 problems :)
On 05/12/06, varun [EMAIL PROTECTED] wrote:
Thanks Karl.
On Tue, 2006-12-05 at 08:20 -0500, [EMAIL PROTECTED] wrote:
I have CentOS 4.4 on several boxes with Asterisk 1.2 and
I have around 20-30 softphones behind NAT .. My sip.conf has nat=yes and
they all are able to register and make calls with no problem . My voip
carrier supports gsm as well as ilbc .. Server takes calls from sip phones ,
does call recording in between and forwards to voip carrier . My problem is
, this is one of the most wanted feature,
but unfortunately will not be in asterisk 1.4 and we must wait for 1.6
to be officially supported feature :'(
PJ
Vicky wrote:
I have around 20-30 softphones behind NAT .. My sip.conf has nat=yes
and
they all are able to register and make calls with no problem
Yeh problem is they are directly buying from providers in US/UK without
paying 12 % tax on voip .. i guess people who buy itsp license can resell
this minutes by paying tax to government in between .
On 08/12/06, ram [EMAIL PROTECTED] wrote:
I'm not sure, but does this only apply to VoIP
canreinvite = yes in sip,conf ( trunk section ) ??
No t,t in dial command . No call recording in between , same codec should be
supported by both trunk as well as extension . If trunk is iax2 and
extension is sip then also asterisk will sit in media path .
On 08/12/06, Alex Guan [EMAIL
Yeh asterisk seems to use extension number for calls between extensions on
same server and sends callerid only for outside numbers ( via sip trunks ) .
On 08/12/06, Greg Kennedy [EMAIL PROTECTED] wrote:
I have a site running asterisk 1.2.8 with a hand full of polycoms and
grandstream 2Kxp's.
that site also has g729 codecs for asterisk but is it legal to use them ?? (
digium charges $10 each g729 channel )
On 08/12/06, Pavel Jezek [EMAIL PROTECTED] wrote:
g723 codec isn't problem, you can obtain for all asterisk versions from:
http://kvin.lv/pub/Linux/Asterisk/
PJ
Jean-Michel
callerid=John Doe 1234
On 05/12/06, Sven Beisiegel [EMAIL PROTECTED] wrote:
Hi...
I just started working with Asterisk and found something that looks
like an error, but i want to be sure, so that's why I'm asking you.
When i make a call from A to B (both SIP clients), I don't see the
name of
Asterisk can record all outgoing calls ( see voip-info.org for asterisk cmd
monitor and mixmonitor ) hardware requirements depends on volume of calls to
be recorded . Faster sata raid or scsi drives recommended for high number of
alternate calls .
On 09/12/06, Raja Chidambaram [EMAIL PROTECTED]
One main disadvantage would be the media stream will pass through asterisk (
no reinvites like sip-sip ) but its not a problem if client pc'a and your
asterisk server are on same network .Sip-iax conversion takes less cpu but
it will be more if codec transcoding is involved .
On 12/12/06, David
I am sure rtp ports arent blocked ..
On 16/12/06, Derek Whitten [EMAIL PROTECTED] wrote:
Mail list wrote:
Hello my isp has blocked outgoing and incoming connection for port 5060
. I
have ssh access to server so i want to send all traffic from port 5091
to
port 5060 of asterisk .so i
I have shifted asterisk port to 5091 . Now i am able to register properly
using sjphone but still when dialing number it keep on showing calling ..
and do not go ahead . I change asterisk's rtp ports too but still i am
unable to make call . My other softphone on different internet isp is
working
[EMAIL PROTECTED] wrote:
Am Samstag, den 16.12.2006, 02:32 +0530 schrieb Vicky:
I have shifted asterisk port to 5091 . Now i am able to register
properly using sjphone but still when dialing number it keep on
showing calling .. and do not go ahead . I change asterisk's rtp ports
too but still i am
I have configure it by using the *2 atxfer feature of asterisk but its not
as good as other attended transfer which sipphones give ( like sjphone where
you can switch between two anytime ) . Also tried zoiper but it do not have
even blind transfer yet . Any idea when idefisk 2.0 is going to be
If you are really new to linux then go for trixbox . I started with trixbox
and eventually went away from it by removing extra stuff and putting custom
compiled asterisk's and removing their rpm's . If you are good at linux then
definitely go for debian + asterisk or centos+asterisk and put
Besides that you can use centos-plus repository which has lot of updated
stuff not available in RHEL4 like php5 , mysql5 and all .
On 18/12/06, Carla Schroder [EMAIL PROTECTED] wrote:
On Sunday 17 December 2006 10:47 pm, Andrew Joakimsen wrote:
I've used Asterisk on a bunch of RH 7.3 machines
I am looking for exactly same kind of billing stuff but i dont think you
will get it without letting ur billing program make some changes in asterisk
.
On 20/12/06, Carlos Rojas [EMAIL PROTECTED] wrote:
a2billing
Is very good
On 12/19/06, Giedrius Augys [EMAIL PROTECTED] wrote:
Post this at bugs.digium.com along with some more info like if it crashes at
use of some specific application or randomly .
On 22/12/06, Edwin Lam [EMAIL PROTECTED] wrote:
our * crashed twice in a month with segmentation fault
a core dump. here's the stack trace:
#0 0xb7e11965 in mallopt ()
I tried it on a intel 3 ghz p4 box and a athlon 3000 768 mb ram running
vista and host for centos 4 ( vmware ) considering the load on athlon
running asterisk ( that too under vista plus vmware ) while intel 3 ghz p4 1
GB ram box was sitting idle with centos , there was hardly a 1 ms difference
I have 3 toll free did's with nufone since 1 month .. Until now i dont have
a problem with them .. their portal was good enough to do proper
configuration and call quality wasnt bad ( even though i havent used them
in really huge traffic yet ) .
On 23/12/06, John Novack [EMAIL PROTECTED]
this really is a great program as far as i have heard even though i am not
able to make it work for me _
On 23/12/06, Matt Florell [EMAIL PROTECTED] wrote:
Hello,
We've released another update to our astGUIclient suite: 2.0.2
http://astguiclient.sf.net/
The client suite runs on most modern
Asterisk can manage dynamic hostnames itseld type dnsmgr refresh in
asterisk cli . Also see /etc/asterisk/dnsmgr.conf
On 10/01/07, Ale [EMAIL PROTECTED] wrote:
Hi all,
My asterisk box have some peers with as host the name of a dynamic dns
resolver ex: foo.dyndns.org.
All works fine, until
If the other server doesnt have any hardware device that can act as timer.
then just compile zaptel and modprobe ztdummy .. This kernel module should
act as timing source i think . ( it works with meetme ) .
On 16/01/07, Andy Hester [EMAIL PROTECTED] wrote:
I have read that an IAX trunk
its notransfer=yes in iax.conf not transfer=no :)
On 16/01/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Monday 15 January 2007 6:21 pm, Anselm Martin Hoffmeister wrote:
could you verify or negate that adding the T option makes it work?
That or transfer=no in iax.conf for hte user/peer
: Launching card: 0
TE410P: Setting up global serial parameters
Found a Wildcard: Wildcard TE410P-Xilinx
Registered tone zone 0 (United States / North America)
TE410P: Span 1 configured for ESF/B8ZS
SPAN 1: Primary Sync Source
==
--
With regards,
Vicky
for
download. You just have to request a free login. they also provide
excellent dialin support - also free. If your framing LED is blinking I
would double check that both ends of your span are set for ESF.
zttool is the tool for working on the cards.
On Mar 17, 2005, at 4:40 AM, Vicky Shrestha wrote
Dear all,
I have tried a lot of things to make broadvoice work with asterisk , but I
failed each time.
Please suggest a good service providers that I can use with asterisk for
outbound and inbound calls.
--
With regards,
Vicky Shrestha
System Director
WorldLink Communications
Jawalakhel
is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Congestion(SIP/502-c147, 5) in new stack
== Spawn extension (vicky, 0018086749157, 2) exited non-zero on
'SIP/502-c147'
-- Got SIP response 400 Bad request back from 147.135.8.128
-- Executing Dial(SIP/502-8efd
description
--
-- 0.1 FORGED_RCVD_HELO
Received: contains a forged HELO
--
With regards,
Vicky Shrestha
System Director
WorldLink Communications
Jawalakhel , Kathmandu, Nepal
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--
With regards,
Vicky Shrestha
System Director
WorldLink
/congested at this time (1:0/1/0)
-- Executing Congestion(SIP/502-c147, 5) in new stack
== Spawn extension (vicky, 0018086749157, 2) exited non-zero on
'SIP/502-c147'
-- Got SIP response 400 Bad request back from 147.135.8.128
-- Executing Dial(SIP/502-8efd, SIP/[EMAIL PROTECTED
hi all,
can any one helpme in how to invite a user(exisiting person) to an already started conference, by using meetme app. in asterisk.
hope every got what i mean.
with regards
asteriskuser
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hi all,
can any one helpme, how to invite a user to an already started conference by using meetme app. in asterisk.
with regards
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anyone help me regarding this please??
with regards
vicky
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hi,
thanks for reply. even after specifying the port, we are getting the same error.
with regards
vicky
On Mon, 02 Jan 2006 Karsten Wemheuer wrote :
Hello,
as You are running two processes handling SIP (asterisk and openser), I
think the Call-File addresses the wrong instance.
If Your
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