Hello, I removed the two lines:
nationalprefix = 0
internationalprefix = 00
And I run bri debug span 1:
*CLI bri debug span 1
Enabled debugging on span 1
1 Timed out looking for release complete
1 Protocol Discriminator: Q.931 (8) len=8
1 Call Ref: len= 1 (reference 2/0x2) (Originator)
1
Hello everybody.
I have a problem with my Billion ISDN card.
When I run Asterisk (asterisk -vvvc) on five minutes (aprox.) it puts in the
screen this:
zaphfc: empty HDLC frame or bad CRC received (framelen = 3, stat = 0xff,
card = 0).
in the framelen it change 3 and 2.
Anyone knows something
Wow... how did you get it to work with Asterisk?
I bought one a few months ago... played with it for 2-3 days... couldn't
get it going... so I stuck it in the basement on my pile of projects to
work on when I have a bunch of free time to waste, hoping it doesn't
become a doorstop.
Did you have
Hi all,
How can i make voicemailmain application to not ask user mailbox. it should
only ask for password. In the prev versions i used the 'u' option for this
purpose but now its gone. So is there still a way to do this?
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
Darryl Dunkin wrote:
What are the end devices? That seems to have
been lost here. The real issue is the handsets as those are the devices
introducing the echo (the only analog players here). Most likely a
volume or gain issue on those handsets, what SIP devices are the echo
issues
On Wed, 13 Jun 2007, Rizwan Hisham wrote:
Hi all,
How can i make voicemailmain application to not ask user mailbox. it should
only ask for password. In the prev versions i used the 'u' option for this
purpose but now its gone. So is there still a way to do this?
Read The Fine Manual (or
Hi all
Asterisk is logging every call in
/var/log/asterisk/cdr-csv/Master.csv in a format like this::
,frisi,s,last extension,005572908060frisi
friessnegger,SIP/sebastianstrasse.at-1be7fab0,,BackGround,
loccata/6308384b6bc39696f05ef4c9ad1e7ea0,2007-02-16 15:56:07,2007-02-16
On 6/13/07, Harald Friessnegger [EMAIL PROTECTED] wrote:
Hi all
Asterisk is logging every call in
/var/log/asterisk/cdr-csv/Master.csv in a format like this::
,frisi,s,last extension,005572908060frisi
friessnegger,SIP/sebastianstrasse.at-1be7fab0,,BackGround,
2007/6/13, Darrick Hartman [EMAIL PROTECTED]:
Andrew Latham wrote:
Oliver
The thing you missed about Gigabit enabled SIP hardphones is the
demand for them.
Not true. I can think of several places where I have or would like to
install phones where the end users currently have Gigabit
Eric,
Before using this version, did you have audio clipping issues (this version
is supposed to solve) ?
Weeks ago, we installed HPEC and got such serious audio clipping issues we
had to roll back and use standard Echo Cancellation.
We can't afford to meet these issues again.
Unfortunately, I
Wednesday, June 13, 2007, 9:44:08 AM, Nick wrote:
I'd love to get it working... if you could share a sample config or other
advice, I'd appreciate it.
http://www.voip-info.org/wiki/view/Linksys-Cisco+SPA400
http://voxilla.com/voxilla-tips/voxilla-tips/the-linksys-spa400-and-asterisk-886.html
Try florz patch, when installing your Bristuff, for me it worked.
Ciao
Mauro
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hi all,
I am running asterisk 1.2.18, zaptel 1.2.18, libpri 1.2.4. on a suse 10.2,
running kernel 2.6.18.2-34-default.
The zaptel drivers are loaded on boot via /etc/init.d/zaptel, but Asterisk
is unable to start. It ends with the following message:
[chan_zap.so] = (Zapata Telephony w/PRI)
where can I download that patch
thanks for respons
2007/6/13, Mauro Zanin [EMAIL PROTECTED]:
Try florz patch, when installing your Bristuff, for me it worked.
Ciao
Mauro
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
On Wed, 13 Jun 2007, Olivier wrote:
Today, buying extra ports for stations having extra bandwidth requirements
is acceptable as 10/100 LAN access is the norm.
But it could be painful to explain executives, every IP Phone you bought
during 2007 will not keep up with 1GE LAN.
There is one other
Sebastian Reitenbach wrote:
Hi all,
I am running asterisk 1.2.18, zaptel 1.2.18, libpri 1.2.4. on a suse 10.2,
running kernel 2.6.18.2-34-default.
The zaptel drivers are loaded on boot via /etc/init.d/zaptel, but Asterisk
is unable to start. It ends with the following message:
Today, buying extra ports for stations having extra
bandwidth requirements
is acceptable as 10/100 LAN access is the norm.
But it could be painful to explain executives, every IP
Phone you bought
during 2007 will not keep up with 1GE LAN.
There is one other issue - I don't think
Steve Finkelstein wrote:
Hi all,
I have the following in my extensions.conf:
exten = s,4,GotoIf($[${CALLERID(number)} = 8585979857 |
8585970327]?15:5)
The numbers listed above are known spammer numbers. However, when I call
from any other CALLERID, it still directs me to s,15 which is
On Wed, Jun 13, 2007 at 01:35:16PM +0200, Josu Lazkano wrote:
where can I download that patch
http://zaphfc.florz.dyndns.org/
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
On Wed, Jun 13, 2007 at 12:31:53PM +0200, Mauro Zanin wrote:
Try florz patch, when installing your Bristuff, for me it worked.
Can anybody report negative experince from that patch where the original
version performed better?
--
Tzafrir Cohen
icq#16849755
Start asterisk as asterisk -vv and see what comes up. Then do a
reload and see what comes up. Also have a look at the logs. Also are you
maxing out the memory on your box ?
- Original Message -
From: Angel Luis Martinez [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
snip
... says that G279A uses slightly less CPU to do the compression at the
expense of sound quality. Digium appear to supply G279 rather than G729A.
(at least they don't mention A)
/snip
Digium actually uses G729A and not G729 Vanilla.
___
Has anyone had any experience using a modem through the Asterisk system?
I have some technical support personnel that need to use a computer
modem to connect to a remote system for troubleshooting. Is there a SIP
compliant gateway that will support a modem connection at decent speeds
(minimum of
Lutgring, Sam wrote:
Has anyone had any experience using a modem through the Asterisk
system? I have some technical support personnel that need to use a
computer modem to connect to a remote system for troubleshooting. Is
there a SIP compliant
This will probably not work for the same
Hi,
When you type ztcfg -vvv, what does it display?
After reboot it shows the following:
asterisk1:~ # ztcfg -vvv
Zaptel Configuration
==
SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
Channel map:
Channel 01: Clear channel (Default) (Slaves: 01)
Sebastian Reitenbach wrote:
Hi,
When you type ztcfg -vvv, what does it display?
How about your zaptel.conf, zapata.conf and the snip of your dial plan
that is doing the dialing.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a
Hi folks,
A user here has asked if we can display the current voicemail message's
envelope (date/time/caller id of message) in the LCD of the Polycom
phones we use (430 501). I realize this is somewhat like the many
caller-id-after-the-fact threads, but I figured maybe someone had solved
this a
Remove Answer() and try .
On 12/06/07, Rosalinda Trevino Cadena [EMAIL PROTECTED] wrote:
I'm using the Dial application in the extensions file with the G option
to execute an AGI script after the Dial (I need to track the call status) as
follows:
exten = _X.,3, Dial({DIAL_STRING},,G(_X.^4))
Hello everybody.
I am trying to configure an Asterisk on Debian with the Billion ISDN card. I
am using mISDN.
But when I call on the CLI apears this:
-- Executing Dial(SIP/101-081805b8, mISDN/1/943833473|45|tTwW) in new
stack
-- Called 1/943833473
P[ 1] empty_chan_in_stack: cannot empty
Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com wrote:
Sebastian Reitenbach wrote:
Hi,
When you type ztcfg -vvv, what does it display?
How about your zaptel.conf, zapata.conf and the snip of your dial plan
that
The latest has the correction of the audio clipping issues. They
should be taken care of now. However, if you still see ANY sort of
problem with the HPEC with the latest version, be sure to let Digium
support know so that we can have it fixed. From the time of getting
audio samples to
Community,
We have released a new patch por mfcr2 chan_unicall
Regarding CID presentation via SetCallerPres() in Asterisk for R2
This files involves legal regulations impact and general support
To dynamic adjustment in the ANI presentation to the user.
They proved to be functional in a telco
Do you just passthrough from FXO to FXS on the channel bank? Does asterisk do
the passthrough or the channel bank itself?
I ask because we're considering an Adit 600 internally and that's one of my
pending questions.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Actually, uncompressed HDMI is 10.2Gbps. I don't think there are any
1080p IP phones yet, but there could be VoIP HD TVs coming. While there
are some MPEG-2 apps with each stream consuming over 20Mbps
( http://en.wikipedia.org/wiki/MPEG-2#Profiles_and_Levels ), there could
indeed be IP
I have done PRI -- Asterisk w/2 Port Card -- TNT to handle modems and
voice on the same PRI line. It basically did a Dial(Zap/g2/${EXTEN}) for
DID numbers that were for the modem bank.
This worked very well even for our ISDN customers. Note, there was no
VOIP involved in that process..
Has anyone seen this before? These phones are behind an edgewater.
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
0.0.0.0:5060;branch=z9hG4bKaf87f1c9f;received=xxx.xxx.xxx.xxx.
From: 7408 sip:[EMAIL PROTECTED];tag=23943befc9dc103
To: 7408 sip:[EMAIL PROTECTED];tag=as2c0b7dcd
Call-ID: [EMAIL
Do you want this as they are checking their messages or in general ? If you
want this as they are checking their messages you will probably need to
modify the code and then send it to the phone (which I am unsure on how to
do).
If you want it to be in the mini browser you will need to write a
Summarizing my previous problem:
I had troubles to execute dialplan commands (for example: AMD) on the
called party call-leg after issuing a Dial().
Today I played around with the Dial-flag M(), which allows to execute
macros on the called party call-leg, before the link (bridging) takes
place
Jeremy Mann wrote:
Do you just passthrough from FXO to FXS on the channel bank? Does asterisk do
the passthrough or the channel bank itself?
The Adit hooks up to the Asterisk via a T1 cable, so you'd need a Dual PRI card
in your Asterisk box. Our channel bank is on channels 25-48.
So you're doing PRI-Channel bank?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Wednesday, June 13, 2007 11:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Using Modems with Asterisk
Doug Lytle wrote:
Steve Finkelstein wrote:
Hi all,
I have the following in my extensions.conf:
I use the mysql addon and create a subroutine that checks for black
listed numbers. I then call it at each entry point (For faxes as well):
; **
; Auto attendant
;
Hi,I have been using wav files with sample rate of 8khz and 8 bits and I find
the sound quality really poor.What is the best sound quality I can achieve on
Asterisk?Responses would be appreciated.Rgds,Akpome
_
Express yourself
Hello
I make one call:
Account:userid
Channel:Local/966190700
Callerid:fromid 966198098
Context:default
MaxRetries:2
RetryTime:5
WaitTime:30
Application:mcc2
Data:fromid|callwithid=966198098|accountcode=userid|direct|SIP/[EMAIL
PROTECTED]|60)
When user answer call, push dtmf numbers, for make
On Wed, 13 Jun 2007, Gergo Csibra wrote:
Wednesday, June 13, 2007, 9:44:08 AM, Nick wrote:
I'd love to get it working... if you could share a sample config or other
advice, I'd appreciate it.
http://www.voip-info.org/wiki/view/Linksys-Cisco+SPA400
Jeremy Mann wrote:
So you're doing PRI-Channel bank?
Yes, for inbound:
PRI-Asterisk-Chanel Bank-Modem/Fax/Cheapy Phone
For outbound:
Modem/Fax/Cheapy Phone-Chanel Bank-Asterisk-PRI
Before we moved to a PRI, it was:
Phone Lines--Chanel Bank- Tellabs Echo
does astribank from xorcom do the same for me?
asterisk-astribank-Modem/Fax
On 6/13/07, Doug Lytle [EMAIL PROTECTED] wrote:
Jeremy Mann wrote:
So you're doing PRI-Channel bank?
Yes, for inbound:
PRI-Asterisk-Chanel Bank-Modem/Fax/Cheapy Phone
For outbound:
Modem/Fax/Cheapy Phone-Chanel
The Digium mailing list server (lists.digium.com) is being replaced with
a new system that will provide better performance and more reliability.
Our IT staff will begin the first step (and hopefully complete the task)
of this migration during the evening tomorrow, June 14th, from 5PM to
8PM
Anthony Francis wrote:
Has anyone seen this before? These phones are behind an edgewater.
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
0.0.0.0:5060;branch=z9hG4bKaf87f1c9f;received=xxx.xxx.xxx.xxx.
From: 7408 sip:[EMAIL PROTECTED];tag=23943befc9dc103
To: 7408 sip:[EMAIL PROTECTED];tag=as2c0b7dcd
Lutgring, Sam wrote:
Has anyone had any experience using a modem through the Asterisk
system? I have some technical support personnel that need to use a
computer modem to connect to a remote system for troubleshooting. Is
there a SIP compliant gateway that will support a modem connection
MR == Matthew Rubenstein [EMAIL PROTECTED] writes:
MR And if you've got GigE installed, not 10/100Mb, and your LAN
MR doesn't have a switch that can handle a phone's lower bitrate
MR without bringing down the whole LAN's rate.
I bet you can't find a switch which acts that way. It existed
On Wed, Jun 13, 2007 at 09:31:30PM +0330, Paradise Dove wrote:
does astribank from xorcom do the same for me?
asterisk-astribank-Modem/Fax
How are calls coming in?
If from an Astribank BRI or FXO module, this should work well.
If from any other Zaptel device, then it should work with the
Hello Asterisk-Users,
I'm trying to install addons 1.2.6 on Asterisk 1.2.16 (is that OK?), but my
MySQL server is installed on a different sever, so the MAKE of the addons fails
with the following (truncated) error message: app_addon_sql_mysql.c:23:19:
mysql.h: No such file or directory.
Is
8khz is not the best sampling rate, but that is the best you can do on the
PSTN. HOWEVER, you should be able to get fairly decent sound qualify out
of an 8khz sound file on the phone line. We have our IVR recorded at 8khz
and it sounds fine. Are you using any compression, or G711u/PSTN for
Try using Unique call id, for example to pass this parameter into agi
script you can use:
exten = s,n,agi,myscript.agi|${UNIQUEID}
Regards,
Luis Morales
On Wed, 2007-06-13 at 19:35 +0530, Jaswinder Singh wrote:
ption to execute an AGI script after the Dial (I need to track the
call status)
On 6/13/07, David [EMAIL PROTECTED] wrote:
Hello Asterisk-Users,
I'm trying to install addons 1.2.6 on Asterisk 1.2.16 (is that OK?), but my
MySQL server is installed on a different sever, so the MAKE of the addons
fails with the following (truncated) error message:
Not that I like answering my own emails, but I hate when I come across
the same problem as someone else and there isn't an answer there.
Anyway, the problem turned out to be in my sip.conf. The variable
'fromuser' was being set for all sip friends. Each friend's fromuser
variable was being set
Hello all,
The wiki has a fairly detailed description of the the issues involved
with encryption of Asterisk calls:
http://www.voip-info.org/wiki/view/Asterisk+encryption
I'm interested in hearing what is working for people today.
I think the ideal solution would be a hard phone that could
On Wed, 2007-06-13 at 00:44 +0300, Atis wrote:
On 6/12/07, Steve Murphy [EMAIL PROTECTED] wrote:
I have created an asterisk.org blog entry:
http://www.asterisk.org/node/48358
to describe what I will shortly be committing to trunk to correct the
weaknesses of CDRs, that asterisk users
Sebastian Reitenbach wrote:
Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com wrote:
my /etc/zaptel.conf
#Configuration for EuroISDN (E1)
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
loadzone = de
defaultzone=nl
Two things come to mind,
(1)
This may seem like a stupid question, but is there a way to kludge
PSTN disconnect tone detection in Asterisk, when the disconnect tone
is fed back purely in the media of a SIP trunk? In other words, no
Zapata interface. I need to troubleshoot very rare and sporadic call
failures from a SIP
Its called CPC
On 6/12/07, Kyle Vorster [EMAIL PROTECTED] wrote:
Hello,
Sorry if this is a real dumb question but when sending a fax and the end
user does not enable fax on their side and then just hangs up does not
force asterisk to end the call.
So it keeps the trunk open until its killed
This is just weird I wrote it in caps so you can read it but you still
didn't read it so here it is again: its a T1 card that does NOT have a
CSU in it, and it is working fine and yes it is a T1 providing PRI.
On 6/11/07, Alex Balashov [EMAIL PROTECTED] wrote:
On Mon, 11 Jun 2007, C F wrote:
so how to avoid CPC??
On 6/14/07, C F [EMAIL PROTECTED] wrote:
Its called CPC
On 6/12/07, Kyle Vorster [EMAIL PROTECTED] wrote:
Hello,
Sorry if this is a real dumb question but when sending a fax and the end
user does not enable fax on their side and then just hangs up does not
force
On 6/13/07, C F [EMAIL PROTECTED] wrote:
This is just weird I wrote it in caps so you can read it but you still
didn't read it so here it is again: its a T1 card that does NOT have a
CSU in it, and it is working fine and yes it is a T1 providing PRI.
sarcasm
Dang shmaltz. You've convinced us
Quoting C F [EMAIL PROTECTED]:
This is just weird I wrote it in caps so you can read it but you still
didn't read it so here it is again: its a T1 card that does NOT have a
CSU in it, and it is working fine and yes it is a T1 providing PRI.
If you are convinced that's what you have, and it
Quoting Erik Anderson [EMAIL PROTECTED]:
On 6/13/07, C F [EMAIL PROTECTED] wrote:
This is just weird I wrote it in caps so you can read it but you still
didn't read it so here it is again: its a T1 card that does NOT have a
CSU in it, and it is working fine and yes it is a T1 providing PRI.
Hello ,
Thursday, June 14, 2007, 4:00:37 AM, you wrote:
Message: 2
Date: Wed, 13 Jun 2007 09:40:08 -0600
From: Anthony Francis [EMAIL PROTECTED]
Subject: [asterisk-users] Weird sip registration problem
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Wed, Jun 13, 2007 at 06:10:04PM -0400, Doug Lytle wrote:
Sebastian Reitenbach wrote:
Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com wrote:
my /etc/zaptel.conf
#Configuration for EuroISDN (E1)
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
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