Hi All,
This is our 2nd E1 client that we try to use crc4 either with the e100p
or with the e405p without luck.
After some trials, we ask the telco to switch off crc4 on their side and
everything works flawlessly.
Is there anything in the crc4 calculation that may be broken? We took a
look at
Title: Message
usecallerid=no in zapata.conf
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Senad
JordanovicSent: terça-feira, 18 de maio de 2004 13:24To:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] X100P
answer in first Ring
I
I may have downloaded an old CVS snapshot, but the following line seems
to be missing at channels/chan_iax2.c/load_module
ast_mutex_init(waresl.lock);
PauloHM
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi Folks,
Can someone tell me how much computing power I need on a machine running
60 channels with GSM compression?
The machine will not be doing anything else but compressing 60 channels
and sending them over an IAX2 trunk.
Best,
PauloHM
___
Hi, I'm having some problems using an IAX2 connection (using GSM) with
an ALAW endpoint.
Seems that the translation path GSM-SLIN-ALAW is working fine (I can
hear the IAX2 party on my ALAW side perfectly), but the path
ALAW-SLIN-GSM yields an distorted voice.
Any clue of what can be going on?
Hi, we are setting a 120-channel conference server and would like to
learn if someone already did this (hardware, problems, etc...)
Best regards,
PauloHM
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi All,
We are compiling an Asterisk interoperability list.
If you have connected Asterisk to either a PBX or another voice/Voip
device (gateway, gatekeeper, etc ...) please drop me an email. I will
compile it and make it available to the list and on the wiki.
Please make sure to send
Hi All,
We have successfully finished implementing R2 support for *.
Drop me an email off-list if you want to test it.
Best,
PauloHM
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Hi All,
I have a e100p that is not receiving any interrupts. My /proc/interrupts
look like
CPU0
0: 87288 XT-PIC timer
1:104 XT-PIC keyboard
2: 0 XT-PIC cascade
8: 1 XT-PIC rtc
10: 814092
Hi Daniel,
AFAIK, As R2 idle bits change between countries, you may put in
zaptel.conf what is the default for your locale.
Something like ...
cas=1-31:1001
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Bichara
Sent: segunda-feira, 12 de
What about you drop your beer, stand up from your couch (if your fat
belly allows you to), turn off the damn TV and try to learn some basic
C programming. Then maybe you can help us in solving those frequent
segmentation faults (if any).
-Original Message-
From: [EMAIL PROTECTED]
Hi All,
I was able to track down what I believe is a bug when using AGI
services. This bug may crash your system if your extensions.conf script
is intensive in using AGI services. Depending on your system's ulimit, *
keeps opening files until it reaches the system limit and then stops
responding.
: [Asterisk-Users] AGI and broken pipe
On Thu, 18 Dec 2003 11:48:59 -0300
Paulo Mannheimer [EMAIL PROTECTED] wrote:
Hi All,
I was able to track down what I believe is a bug when using AGI
services. This bug may crash your system if your extensions.conf
script is intensive in using AGI services
Title: Message
Hi
Miklos,
try
starting * with -vvvc and see if there is any
warning
also,
try connecting to your mysql server by issuing mysql asteriskcdrdb then
show tables;
select * from cdr;
best,
PHM
-Original Message-From:
[EMAIL PROTECTED]
Hi,
I'm trying to use iaxcomm. I can place a call from the softphone, but
when I place a call to it, when I answer I get ...
NOTICE[16401]: File channel.c, Line 1094 (ast_read): Dropping
incompatible voice frame on IAX2[paulohm]/3 of format GSM since our
native format has changed to ALAW
My
: [Asterisk-Users] pridump
the two dchannels.
mark
On Wed, 10 Dec 2003, Paulo Mannheimer wrote:
Hi All,
Can anyone tell me what are the dev1 dev2 parameters that I should
use to run pridump? I took a look at the source code but couldn't
figure this one out.
Best,
PauloHM
Talking to myself ... ;-)
Solved this by ...
disallow=all
allow=gsm
;allow=ulaw
;allow=alaw
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paulo
Mannheimer
Sent: quinta-feira, 11 de dezembro de 2003 09:02
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users
Hi All,
Can anyone tell me what are the dev1 dev2 parameters that I should
use to run pridump? I took a look at the source code but couldn't figure
this one out.
Best,
PauloHM
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi, not sure if this is your case, but a got rid of my error 500
messages today by changing the machine's motherboard.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Markus Mayer
Sent: quarta-feira, 10 de dezembro de 2003 15:18
To: [EMAIL PROTECTED]
Please drop me an email off-list if you can provide IAX termination in
the Netherlands.
Best regards,
PauloHM
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi All,
We are just starting to deploy a new PRI IVR system, and the incoming
calls sometimes get random short 'bips' while navigating our IVR menu.
Any hint on what this can be?
Best regards,
PauloHM
___
Asterisk-Users mailing list
[EMAIL
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Strage bip on ISDN/PRI
On Tue, 2003-12-09 at 11:20, Paulo Mannheimer wrote:
Hi All,
We are just starting to deploy a new PRI IVR system, and the incoming
calls sometimes get random short 'bips' while navigating our IVR menu.
Any hint on what
=pri_cpe
relaxdtmf=no (yes doesn't seem to help)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paulo
Mannheimer
Sent: terça-feira, 9 de dezembro de 2003 16:33
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Strage bip on ISDN/PRI
Sorry for the short
Hi All,
Please drop me an email if you can provide Iax termination in India.
PauloHM
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi all,
We are about to make our first channel bank install. This will be a one
PRI outside connection and up to 70 extensions.
As the schedule (and the budget) is pretty tight, I would like to learn
a little bit more about general experiences with channel banks, like
echo cancellation
-10 at 05:54, Paulo Mannheimer wrote:
Thanks Steven.
I'll have to find a way to use bandwidth only when the call to the
PSTN is completed on the other side.
Why does that matter? are you on a metered connection for bytes?
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Try safe_mysqld --skip-grant-tables
and configure your password and your allowed hosts
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of DIPAK PAUL
Sent: segunda-feira, 10 de novembro de 2003 04:45
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]; [EMAIL
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Iax2 channel usage
On Sun, 2003-11-09 at 14:01, Paulo Mannheimer wrote:
Hi all,
In a forthcommming project, I'll have one * server tentatively calling
10 PSTN numbers through IAX2 and an * gateway.
Can someone tell me if bandwidth is being used
Hi all,
In a forthcommming project, I'll have one * server tentatively calling
10 PSTN numbers through IAX2 and an * gateway.
Can someone tell me if bandwidth is being used for each of these
calls/channels even while my gateway tries to call and connect the
destination numbers?
Best,
PauloHM
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: quinta-feira, 30 de outubro de 2003 10:24
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Sip bandwidth usage
Paulo Mannheimer wrote:
That's weird. I've done some testing both with GS and Xten products,
and my iptraf readings show
Hi All-
I'm working on a project that will have remote (internet)access to an *
server through SIP phones, either soft or hard ones.
Does anyone have any experience to share about which SIP product they
are using under similar conditions, as well as which codec is being used
and bandwidth usage?
Hi, thanks for you reply. I'll send you till the end of the week more
info on how to download and use it.
Best regards,
Paulo Mannheimer
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lal, Deepak
(Contractor)
Sent: sexta-feira, 17 de outubro de 2003
Hi, thanks for you reply. I'll send you till the end of the week more
info on how to download and use it.
Best regards,
Paulo Mannheimer
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas
Wienecke
Sent: sexta-feira, 17 de outubro de 2003 17:43
Hi, thanks for you reply. I'll send you till the end of the week more
info on how to download and use it.
Best regards,
Paulo Mannheimer
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Coberly
Sent: sábado, 18 de outubro de 2003 14:49
To: [EMAIL
Hi, thanks for you reply. I'll send you till the end of the week more
info on how to download and use it.
Best regards,
Paulo Mannheimer
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josh
Roberson
Sent: sábado, 18 de outubro de 2003 01:21
To: [EMAIL
Hi All,
We've been developing for a while an IDE for Asterisk, and the time has
come to open it for beta testers.
You can check at www.instant.com.br/viv.html for a snapshot of the
application.
Current modules are Dialplan and VoiceMail configuration. As you may
see, it is all-visual, with
Here is a patch that I posted to Mark a couple of days ago. Haven't
tested it too much.
It basically implements the system command through the manager
interface. Due to security issues, you have to create a system.conf file
at /etc/asterisk with the commands that you wish to allow.
-Original
Take a look at zaptel/zonedata.c, I guess you have to change it.
Greetings from Rio de Janeiro ;-)
PHM
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andre
Lomonaco
Sent: quarta-feira, 15 de outubro de 2003 16:40
To: '[EMAIL PROTECTED]'
Subject:
Hi folks,
I'm still having the following problem, maybe someone can help me out of
it.
Two IDENTICAL MACHINES (same motherboard, same RH 7.2, same *)
communicate through IAX2. Everything works ok on machine 1. On machine
2, if I try to use 4 fxo's from a TDM400 card, sound gets lousy. If I
Hi all,
I'm looking for the following functionality: if my queues reach a
certain threshold, I would like to disable any available zap / PRI
channels, so my telco doesn't try to connect more people. After a while,
I will enable them again.
Any hints on how to implement this? Should I be looking
Hi Folks,
I'm trying to interface * with a PBX, but seems that his ring cadence is
somewhat different, and my T100 doesn't show any call coming in.
I've tried to change zaptel to new values but still couldn't make it
work.
Is there any other place where I should be changing some parameter? Is
Hi folks,
As none of the SIP softphones that I tested can disable more than one
incoming call, I decided to implement it by software ;-) I'm attaching a
patch that does it.
To make it work, modify your sip.conf file and include callwaiting=[0|1]
at the general section, or for each peer that you
Damn. Seems to implement what I was looking for ... ;-(
Does anyone know if the incominglimit works if the call is being
generated from a queue?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: September 17, 2003 2:19 PM
To: [EMAIL
Hi Rich,
We have done this before. We basically developed a small client that
sits on every machine and communicates with * to get information about
an incoming call. Contact me off-list and I will be glad to tell you
more about the entire solution.
-Original Message-
From: [EMAIL
: September 16, 2003 4:09 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] call center design question
Yes, Please share.
On Tue, Sep 16, 2003 at 03:05:33PM -0400, Yifang Dai wrote:
On Tue, Sep 16, 2003 at 03:27:44PM -0300, Paulo Mannheimer wrote:
Hi Rich,
We have done this before. We
I'm not sure I understood your question.
As far as I know, listening to the manager interface wouldn't give me
enough information. At the moment where the call is transferred, the
client has already browsed through a couple of menus, setting some
variables. The AGI sends the content of these
Is there anyone out there with a custom client softphone and is
interested in integrating both solutions?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of TC
Sent: September 16, 2003 3:53 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] call center
] On Behalf Of John Todd
Sent: September 11, 2003 8:20 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP busy
[message re-ordered]
- Original Message -
From: Paulo Mannheimer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, September 11, 2003 4:32 PM
Subject: [Asterisk-Users
Me too. I sent Steve an email about this, but didn't get a reply.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of LQ
(Asterisk)
Sent: September 11, 2003 10:19 AM
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Is there any MFC-R2
Hi,
I would like * to treat a SIP extension as a normal extension, when it
comes to the busy functionality. In other words, if someone tries to
call the SIP phone and there is already an ongoing conversation, the new
caller should get a busy message/tone
Is there any parameter that I can set? Is
Hi,
I have an installation connecting two machines through IAX2. Each
machine has 3 FXS and 4 FXO ports.
Everything seems to work fine, except on one FXO port, where I
constantly get a strange locomotive noise when I use it to terminate
an IAX2 incomming call. Usually after a while the strange
Hi,
My installation that was working flawlessly for 2 weeks stopped working
when I installed a g729 codec license.
Now, if I try to start * I get a File size limit exceeded error and
the program aborts.
Any clue of what's going on?
___
Found what was going on ...
My debug file at /var/log/asterisk was greater than 2 gigs (don't ask me
why ...)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paulo
Mannheimer
Sent: September 08, 2003 8:47 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk
Hi All,
Does anyone have any experience with the ArrayCom VoIP phone?
I bought one a couple of weeks ago, it used to work quite well with *
until I misconfigured one option.
I now cannot make it work anymore, because the phone boots up, doesn't
find a valid SIP gateway, resets itself and keeps
Hi,
I'm testing an E1 with EM signaling. Some of the problems I'm running
into are the following:
1) if I try to configure any channel above channel 15, I start
getting a multiframe alignment error on my telco test equipment. So I
have my zaptel file only configured for 15 channels, like
Am I crazy or do you have a Goto just before your Record command?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kaku ustaad
Sent: August 25, 2003 8:33 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Why doesnt anyone reply me ?
I have posted soo
Has anyone used * and IAX in a gateway to a videoconferencing
application?
Best,
PauloHM
Answering myself,
It seems that my zaptel
service script wasnt loading the wct1xxp module.
Should I load something else? Torisa? Tor2?
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paulo Mannheimer
Sent: August 12, 2003 11:23 AM
Hi, Im installing my first E100P.
My zaptel reads the following:
Span=1,0,0,ccs,hdb3,crc4
Em=1-31
My Zapata.conf reads the following:
Signaling = em_w
Channel =1-15
Channel =16-31
After starting the zapter service
I get:
ZT_SPANCONFIG failed on span 1: No such device
Hi folks, where can I find the R2 beta code for Asterisk?
Best,
PauloHM
Hi folks,
Im having problems accessing my voicemail files
through the web interface.
I remember that this was discussed on the list, and it seems
to be a permission problem, but I couldnt find any answer by searching
the archives.
Any hint?
PauloHM
Thanks!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: July 30, 2003 4:06 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] voicemail file access problems
On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote:
Hi folks
/vmail.cgi
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Paulo
Mannheimer
Sent: Wednesday, July 30, 2003 3:23 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] voicemail file access problems
Thanks!
-Original Message-
From: [EMAIL
Try increasing busycount (a hidden parameter) at Zapata.conf
Mine works like a charm with
busydetect=yes
busycount=6
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerk Face
Sent: July 29, 2003 9:03 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
] busydetect and random hangups
increase busycount in zapata.conf
busycount=6 is ok for me.
the default is 3 , I think, and sometimes
it hangsup on speaking (or some other moh ;) )
Matteo.
Il mar, 2003-07-22 alle 22:11, Paulo Mannheimer ha scritto:
Hi,
I'm having random hangup problems
Hi,
Im having random hangup
problems with zap channels.
If I turn busydetect
off in Zapata.conf, * fails completely to detect a
user hangup in the middle of a script.
On the other hand, if I turn it on, everything works much
better, but long calls tend to be hung up without a
Hi,
Im localizing the voicemail messages to Portuguese. To
make it possible for another person to translate it, Ive set up a couple
of extensions that call the following macro for each message on the system. After
recording, I can perfectly hear each message using Playback.
When I
Hi folks,
There was a bug with the GotoIfTime
built-in command, under certain circumstances a variable contained garbage,
screwing up correct time identification.
Im submitting now a patch to Mark so this can be
fixed.
PauloHM
on the bitfield logic.
PauloHM
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: July 01, 2003 1:24 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] gotoiftime error
On Tuesday 01 July 2003 09:08 am, Paulo Mannheimer wrote:
Hi folks
Im getting this intermittent problem, sometimes a zap
channel gets stuck after a call. Below is a snapshot of the channel. Any ideas
what can be happening?
Name: Zap/1-1
Type: Zap
UniqueID: 1056988772.10
Caller ID: (N/A)
DNID Digits: (N/A)
State: Up (6)
Rings: 0
I just noticed that app_queue here
rings together all available extensions, which may not be the best for a call
center.
Is this the correct functionality or something specific from
my installation?
PauloHM
-
From: Paulo Mannheimer
[mailto:[EMAIL PROTECTED]
Sent: Monday, June 23, 2003 6:36 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] dynamic
queue channels
Hi, Im trying to build a call
center application that allows attendants to come in the morning and dial a
certain extension to make
: Benjamin Miller [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
[EMAIL PROTECTED]
Date: June 26, 2003 11:45 AM
Subject: RE: [Asterisk-Users]
dynamic queue channels
Nice work! :-)
Thanks
Cant wait to see it in
cvs.
-Original Message-
From: Paulo Mannheimer [mailto:[EMAIL PROTECTED]]
Sent
but that pacth is not
against current cvs
-Original Message-
From: Paulo Mannheimer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
[EMAIL PROTECTED]
Date: June 26, 2003 2:13 PM
Subject: RE: [Asterisk-Users]
dynamic queue channels
Sure, here it goes. PLEASE
READ THE DISCLAIMER BELOW
I think it's a good start, and would be willing to work on expanding the
concept.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dylan
VanHerpen
Sent: June 26, 2003 6:29 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Web interface for Asterisk
Hi
Hi,
Can someone provide information about IAX termination in the
US and other
countries?
I tried Google but nothing showed up ;-(
PauloHM
Hi, Im trying to build a call center application that
allows attendants to come in the morning and dial a certain extension to make
their extension available.
I wouldnt like to use the AgentLogin
app because their line would need to stay off-hook (is this correct?)
Is there any SET
Hi,
Im working on a call center application where callers
input some information and get transferred to an attendant, or waits in a queue
until one is available. The operator is using a PC-based system that needs to
have access to the information previously input by the caller. I was
PROTECTED] On Behalf Of Martin Pycko
Sent: June 11, 2003 3:51 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] lost variables
Why do you think so?
Local variables get lost only when the call gets hanged up.
Martin
On Wed, 11 Jun 2003, Paulo Mannheimer wrote:
Hi,
Seems that my local variable
Hi,
Seems that my local variable content get lost when I call an
AGI program. Is this the correct functionality?
Thanks,
Paulo H. Mannheimer
Hi,
I have an incoming call that I would like answered every
time by a different SIP phone (out of 50).
Also, some of the phone may not be available (may be turned
off and thus unregistered with Asterisk).
Any way of doing this?
Paulo H. Mannheimer
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: June 05, 2003 2:25 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] answering calls with SIP phones
On Thu, 5 Jun 2003, Paulo Mannheimer wrote:
I have an incoming call that I would like answered every time by a
different
Hi All,
Ive been working with asterisk for about two months,
and I would like to contribute to the project on the localization side, mostly
making it easier to translate text output and pre-recorded messages.
My goal is to discuss with you guys a framework for
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