Greetings!
Iam runninga small callcenter with 10
analog lines, aprox. 15 agents and usingAsterisk 1.2beta1. We have10
sipura 3000s connected to the PSTN and a few linksys PAP2s.
The ports connected to phones are configured as
SIP/200s and SIP/300s and the ones connected to the PSTN as
all
I cannot get ChanIsAvail to work with sip or iax on
v1.0.3. It does work fine on a zap channel. I am trying with Sipuras and
PAP2s.
It appears I am not the only one having this
problem. Has anyone gotten it to work?
___
Asterisk-Users mailing
Folks!
I discovered some serious problem with several
Sipuras 3000 but I don't know if the problem is with them or Asterisk.
Basically, if I call a Sipura PSTN line, when there is a call already in
progress, generally I get a 503 Sevice Unavailable, but if I try hard enough, I
am able to
I have 4 sipuras 3000in a small
callcenterconnected to the PSTN receiving calls and forwarding them to
Asterisk and viceversa.
In addtiion I have some x100s, linksys FXSs,
etc
Strange things are happening with the Sipura and
Asterisk which I cannot seem to figure out. During off hours at
I am using AgentCallbacklogin to logon agents. I am
trying to avoid agents being logged in more than once in different extensions
(is this a bug?) by passing the callerid to the AgentCallbacklogin funtcion as
an option. The problem is thatby doing this, agents are not askedfor
an extension
I can send calls from asterisk to a Sipura FXO
interface (SIP/300) from any SIP phones including SIP/205 which is the Sipura
3000 FXS interface.
The problem I have is when a call from the PSTN is sends to Asterisk. On
extnesion conf I dial all the SIP clientsI get a 302 Moved temporarily
For some strange reason, Asterisk is telling giving
me the following error when I leave a voicemail
"E-mail address missing for mailbox
[1005]"
on voicemail.conf I have
1005 = 1234,Dan
Fernandez,[EMAIL PROTECTED]
Anyone?
Alejandro
Why can't you use IAX? I'd love to test your
termination.
Saludos
Daniel
- Original Message -
From:
Alejandro Sosa
To: [EMAIL PROTECTED]
Sent: Tuesday, July 20, 2004 2:54
PM
Subject: [Asterisk-Users] FREE (305) and
(786) termination. Anyone
I cannot get Festival to work with asterisk. I have
the following:
exten = 555,1,Answerexten =
555,2,Festival(mary has a little lamb)exten = 555,3,Hangup
I get the following from asterisk: "Festival returned ER" and the festival logs shows the
following:
client(1) Fri Jul 16 15:35:54 2004
On Wed, 12 May 2004, Dan Fernandez wrote:
Asterisk should answer the call, playback a message, dial another PBX
extension and if no one answers dial another extension (via IAX).
The first problem I ran into was that the Flash application doesn't
really work. To get around this I added
-
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, May 18, 2004 5:56 PM
Subject: Re: [Asterisk-Users] problems with analog interface to PBX
On Tue, 2004-05-18 at 15:45, Dan Fernandez wrote:
Steve,
Thanks for your respnose. The flash does seem to work. If I
Folks,
For the last few days I've been trying to
experiment with a Panasonic PBX and an X100P but have run into quite a few
problems which I am not sure if they can be solved with this type of card (how
about TDM01B?)
1) I wanted to use *'s IVR capabilities,so I
routed the calls to the
I have an X100P connected to an extension of
aPanasonic PBX.When a call from the PSTN comes in,it is routed
directly to theextension where the x100p is .I want* to answer
the call, play amessage and then transfer the call to another extension
via the Zap channel where the call was received
I have Suse 9.0 with gcc3.3.1 (didn't have any
problem with the previous version of gcc )and when I run make install I get the
following error:
/usr/lib/gcc-lib/i586-suse-linux-/3.3.1/../../../.../i586-suse-linux/bin/ld:
cannot find -lz
Any help would be appreciated.
Dan
I finally figured it out. Had to install zlib-devel
package.
sorry for the posting, but it was driving me
nuts.
- Original Message -
From:
Dan Fernandez
To: [EMAIL PROTECTED]
; [EMAIL PROTECTED]
Sent: Monday, February 23, 2004 8:07
PM
Subject: [Asterisk
I am having problems with early dialing and
chan_phone. In extensions.conf Ihave:
exten = _41.,1,Dial,IAX
If I dialvia a SIP or ZAP channels it works
fine.With chan_phone it start dialing right after the 3rd number.
If tried different combinations like (41., ... or
_41X., ) and still
Any news on this regard?
If this is not implemented yet, what alternatives do we have? A channel
bank?
- Original Message -
From: Paulo Mannheimer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, September 11, 2003 10:23 AM
Subject: RE: [Asterisk-Users] Is there any MFC-R2
Last week I did aCVS update and since then I
haven´t been able to run asterisk normally.The strange thing is that I
have even go back to previous versions (0.5.0) andI am seening the same
problems.
Basically, when I try to load the zap module I get
the following error:
WARNING[16384]:
Last week I did aCVS update and since then I
haven´t been able to run asterisk normally.The strange thing is that I
have even go back to previous versions (0.5.0) andI am seening the same
problems.
Basically, when I try to load the zap module I get
the following error:
WARNING[16384]: File
] problem loading chan_iax2.so and
chan_zap.sofrom latest CVS
On Tue, 2003-09-16 at 20:27, Dan Fernandez wrote:
I just updated to the new CVS and now I am getting the following error
from chan_zap (modprobe wcfxo works fine):
WARNING[16384]: File chan_zap.c, Line 577 (zt_open): Unable
Yes, setting callprogress=no fixed the problem.
Thanks to everyone.
- Original Message -
From: Martin Pycko via RT [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 16, 2003 6:43 PM
Subject: [digium.com #860] Fw: x100P: Ring/off-hook in strange state 6 on
channel1
it
I just updated to the new CVS and now I am getting the following error from
chan_zap (modprobe wcfxo works fine):
WARNING[16384]: File chan_zap.c, Line 577 (zt_open): Unable to
specify channel 1: Device or resource busyERROR[16384]: File
chan_zap.c, Line 4781 (mkintf): Unable to open
I´ve been having this same problem for a few weeks now.
WARNING[262160]: File chan_zap.c, Line 2857 (zt_handle_event):
Ring/Off-hook in strange state 6 on channel 1
I get this message and then the Zap channel hangs up and it does not Answer
the call. I have no problems dialing out.
This used
All of a sudden I am getting the following warning
"Ring/off-hook in strange state 6 on channel1" from chan_zap.c and I cannot
answer calls. I can place calls out without a problem though.
Any ideas what can be the problem. I have checked
zapata.conf and zaptel.conf and they both seem fine.
- Original Message -
From: Michiel Betel [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, May 19, 2003 11:53 AM
Subject: RE: [Asterisk-Users] CDR-Event on AstManager
The manager inteface currently sends the following events with the
associated parameters:
Event: Newexten Channel
I am having problems using the manager even though
I am following the instructions from the Manager.rtf doc.
In manager.conf I have the
following
[general]
enabled=yes
port=5038
[fred]
username=fred
secret=fred
read=system,call,log,verbose,command,agent
Ricardo
Have you tested g729 between two endpoints (SIP) for over 5mins?
My experience has been that after 3-4 mins both ends begin to get huge
delays and after a few minutes is impossible to continue the conversation.
HAve you done any testing similar to mine?
- Original Message -
On your sip.conf for each sip endopoint set canreinvite = yes.
That way the rtp stream won´t go through *. The only problem though is for
ATA 186. They need canreinvite = No when they are in a NAT environment.
- Original Message -
From: Low, Adam [EMAIL PROTECTED]
To: [EMAIL
Is there a way in iax to have to endpoints talk to
each other directly (after the call is setup by *) without going through
*. In sip, with * you can do it by
configuring sip.conf with canreinvite = yes.
, Dan Fernandez wrote:
I would like to run an agi script (to calculate the cost of a long
distance or international call) right after I execute a Dial app.
Can this be configured in extensions.conf? It seems the entries
It cannot. If the Dial app succeeds in getting a connected channel
://www.junghanns.net/asterisk
Am Sam, 2003-07-26 um 01.28 schrieb Dan Fernandez:
John
Thanks for the response. This seems to be what I am looking. However, I
have discovered a problem with a simple perl script triggered from the h
extension.
I am using perl-Asterisk and if I call the script from
I am having someproblems with g729 with SIP
and ZAP channels.
1)
I have two g729 licences. Very frequetnly (I don´t
know what triggers the error) I get the followingwarnings and
errorwhen I try to place a call via SIP to my X100P. The only way to get
out of this is through a restart of *.
Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 23, 2003 4:47 PM
Subject: Re: [Asterisk-Users] Problems with g729
Try the new_codec_binary/codec_g729b.so from the digium ftp site.
regards
Martin
On Wed, 23 Jul 2003, Dan Fernandez wrote:
I am having
I would like to run an agi script (to calculate the
cost of a long distance or international call) right after I execute a Dial app.
Can this beconfiguredin extensions.conf? It seems the entries right after a Dial app get executed only if the Dial
app was executed unsucessfully.
Would I have
: Re: [Asterisk-Users] executing an agi script after a successful
Dial
On Wednesday 23 July 2003 05:04 pm, Dan Fernandez wrote:
I would like to run an agi script (to calculate the cost of a long
distance or international call) right after I execute a Dial app.
Can this be configured
I have discovered a problem when using g729 under
the following setup:
SIP call between a Budgetone 102 and ATA 186
(configured without silence suppresion). Both ends have a ADSL 64kbps. Both ends
are behind Linksys routers. The pings between them are aprox. 100ms. No other
local users on
I have a Budgetone 102 with the latest firmware 1.0.3.72 and using
dtmfmode=rfc2833
With g711 I have no problem with Voicemail or Voicemail2.
With g729 it always repeats digits and it is impossible to check my
voicemail (or any other apps that require digits)
- Original Message -
From:
and even works with g723.1
:)
On Wednesday 09 Jul 2003 11:15 pm, Dan Fernandez wrote:
I have a Budgetone 102 with the latest firmware 1.0.3.72 and using
dtmfmode=rfc2833
With g711 I have no problem with Voicemail or Voicemail2.
With g729 it always repeats digits and it is impossible
Steve
Can you please give us some guidance on how to make call progress work
outside the US or UK?
Thanks
Dan
- Original Message -
From: Stephen Davies [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, June 21, 2003 4:51 AM
Subject: Re: [Asterisk-Users] Billsec on CDR
On Fri,
John,
Thanks for the detailed guide.
As you mentioned, the situation where two ATAs behind NAT want to establish
a direct connection is not resolved yet. In that case, the canreinvite would
have to be set to no and some other solution outside of * would have to be
used to traverse the NAT. Have
I have a Budgetone and an ATA but none of them
support GSM. I´d like to place call to the PSTN with my X100P viaa WAN
(64kbps). g711 is out of the question. Can * transcode from g723.1
to GSM? How costly is it? I have tried different configurations on
sip.conf and extensions.conf but have
I have an X100P and when I place calls to the PSTN
which are not answered, the Billsecfield of the CDR still logs the seconds
that the phone rang.
Can someone please confirm that this has to do with
the ringcadance of the indications.conf file? Is there anything else I need to
check ?
Will look into this once someone can help me with the configuration behind
NAT (without NAT I have no problem)
I am using v1.0.3.53 and a linksys router (the phone IP is 192.168.1.2)
I´ve tried in my sip.conf with and without NAT=1.
In the phone, if I set the outbound proxy to the linksys it
Is there a way to reduce the number of rings if
there is a message on the mailbox. That is I set the Wait app to 10 secs but
then want it to pick up a call right away after someone leaves a message (ie I
am not at home, office)
How can i do this?
Thanks in advance
Dan
yast2', find the Install Packages option, and
search for newt.
Hope that helps,
-BAK
On Mon, 2003-03-10 at 16:26, Dan Fernandez wrote:
Can astman be compiled without newt? I have Suse 8.1 and it doesn´t
have newt. If needed, where can I get it?
Thanks in advance
--
Ben Klang [EMAIL
, 2003-03-10 at 23:01, Jim Archer wrote:
--On Monday, March 10, 2003 4:47 PM -0300 Dan Fernandez
[EMAIL PROTECTED] wrote:
Iconnect uses codecs g723 and g711 that can be configured for each
account (you can change them by the prefix)
I tried adding the in front
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