- Original Message -
From: Jean-Michel Hiver
If I understand, you are doing.
Extension (VoIP) - (VoIP) Asterisk (VoIP) - (VoIP) Zyxtel (FXS) -
(FXS) GSM
Unless I've missed something, it seems that you are trying to plug a
telephone adapter (Zyxtel) to another telephone adapter
Searching through wiki and google.
http://www.2n.cz/products/gsm_gateways/voip/voiceblue.html
but there are also other products on the
market.
---
Wondering if its possible to connect as
follows:
Extension - Asterisk -
ZyxelAnalogTelephoneAdapter - GSM gateway.
The best way would be to
Did you remember to push 1# to confirm ?
Could it be so easy that the password is set to your phonenumber ?
as posted on some forums;
It is possible for service providers to restrict access to this feature. If
you are using a SPA that has been provisioned by a provider, you may need
the
Hi,
We are trying to make the http://sourceforge.net/projects/asteriskbilling/
work properly.
When we call the prepaid extension, I get these results:
-- Executing Goto("SIP/03-5ac7", "prepaid|s|1") in new
stack -- Goto (prepaid,s,1) --
Executing Answer("SIP/03-5ac7", "") in new stack --
Is there a way to debug, more debug than already
than the option - does ?
Like...
when "Answer" is executed, can I get more info from
where this app is run, what data is processed.. ?
(is there a monitoring app. which I can use ?)
Or is this just asking for alot of unusable info ?
/
I'm running Trustix 2.1, and there is no telnet daemon. It is replaced by
sshd.
Your flash operator looks good!
I'll will go for this.
/ Stig Henning
- Original Message -
From: Nicolás Gudiño [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL
Title: Zyxel Prestige 2002/2002L sound quality
Hi,
This ATA works fine for us.
Haven't had any problem with sound issues, other than 1-way audio, but that
is another problem which is fixed.
But checked with different analog phones, and there seems to be a
difference. good and bad quality.
This sounds odd.
We use the same adapter.
I will check this more..
Are u sure you have set the phone up correctly ?
And also - have to checked the ring phone1 or phone2 on incomming calls ?
/ Stig Henning
- Original Message -
From: Mihkel Raba [EMAIL PROTECTED]
To: Asterisk Users
In my queue.conf I have:
; Maximum number of people waiting in the queue (0
for unlimited);maxlen = 5
-
Log:
--
Executing Goto("SIP/242.112.162.21-081a8b90", "veksel|s|1") in new
stack
-- Goto (veksel,s,1)
-- Executing Answer("SIP/242.112.162.21-081a8b90", "") in new
Followed this;
#cd /usr/src
#export
CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
#cvs login (password is
anoncvs)
#cvs
checkout zaptel libpri asterisk
#cd zaptel
; zaptel equipment
#make clean; make install
#cd ../libpri
; isdn
#make clean; make install
#cd
../asterisk
#make
NOTICE[98310]: chan_sip.c:6638 handle_response:
Failed to authenticate on INVITE to
'sip:[EMAIL PROTECTED];tag=as0f1d3429'
sip.conf
register =
1234:[EMAIL PROTECTED]
extension.conf
--
;; Own extensions;exten =
You have installed asterisk = ok.
You try starting asterisk:
#safe_asterisk
errors ??
if none, then try
#asterisk -r
you enter the consol, and get the CLI command prompt.
/ Stig Henning
- Original Message -
From: Joao Carlos Moura [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
Make sure you don't use mp3 with variable bitrate.
Record mp3 at 128 - constant should work.
/ Stig Henning
- Original Message -
From: Altus Snyman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Monday, September 13, 2004 1:53 PM
Followed; http://www.voip-info.org/wiki-Asterisk+Agents
agents.conf
[agents]
agent=1001,4321,BenDover
queues.conf
[queue1]
member=Agent/1001
extensions.conf
exten=28,1,AgentLogin(1001)
exten=29,1,Queue(queue1)
But when I call number 28, I get:
"Please enter your password followed by
Have you configured;
_ sip.conf_
..add this line:
dtmfmode=inband
..also you have uncomment the right line that
matches your dhcp setup:
localnet=192.168.0.0/255.255.0.0; All RFC 1918
addresses are local networks
;localnet=10.0.0.0/255.0.0.0 ; Also
RFC1918;localnet=172.16.0.0/12 ;
Asterisk is up running.
SIP phones (SJPhone, xlite i.e) can call an
extension and get a voice message...
..then expect the customer to push 1, 2 or 3 to be
sent to other extensions.
This works for SIP phones.But when mobiles or ordinary phones
(what is the tech word for this?) call the *
Is there a file I can tail, or do print out of
?
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My asterisk server has an external IP.
Also my Ericsson DRG22 has an external
IP.
Tried setting up the Ericsson DRG22
device.
*SIP Server IP (primary): MYIPNUMBER
User name: drg22-2
password: password
Caller ID name: drg22
Telephone number: 12345
*Telephone domain name:?? (I only have
I'm new to this list.Reading the asterisk
handbook pdf (good work)but but still have some questions.
Using Trustix 2.1 and installed Asterisk via CVS,
zaptel and libpri.
We have a dedicated server which is connected to
our telephone company.
It makes us able to call ordinary phones via VOIP
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stig Thune
Sent: Monday, August 23, 2004 6:39 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] routing telephone calls via
switchboard/asterisk.
I'm new to this list.
Reading the asterisk handbook pdf (good work
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