Hi Steve,
Thank you for your response. As you said, i tried. But, no result. Here I am
sending my configuration file.
Contents in Zapata.conf:
[trunkgroups]
[channels]
language=en
context=from-zaptel
signalling=fxs_ks
rxwink=300
usecallerid=yes
relaxdtmf=yes
dtmfmode=rfc2833
Still failing :(
2006/11/25, Leo Ann Boon [EMAIL PROTECTED]:
Jesus Jimenez wrote:
Hi ,
I have a problem with a X100, i do a external call to the
asterisk server . The dialplan its simple answer and hangup..
when it's done , the telephone which i did the call , is in line but
On Thu, Nov 23, 2006 at 01:59:25PM -0500, Paul wrote:
I have not created my final web site, but rather put together a quick one
which will contain more free Asterisk software and tips as time permits.
http://www.siliconvp.us
For those who didn't notice it, this is a glorified 'asterisk -rx
On Sat, Nov 25, 2006 at 01:38:42AM +0100, Jesus Jimenez wrote:
Hi ,
I have a problem with a X100, i do a external call to the asterisk
server . The dialplan its simple answer and hangup..
when it's done , the telephone which i did the call , is in line but
asterisk server is finish.
The right syntax should be externip=${ENV(MYIP)} but I **think** variables
are only allowed in extensions.* and not in sip.conf.
---
Dominique Dartois
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Larry Alkoff
Envoyé : samedi 25 novembre 2006 04:01
À
On Sat, Nov 25, 2006 at 11:58:01AM +0100, Dominique Dartois wrote:
The right syntax should be externip=${ENV(MYIP)} but I **think** variables
are only allowed in extensions.* and not in sip.conf.
Right, they are.
As a workaround, use a trivial shell script (with sed -i) to rewrite the
IP
Hi
I had some backlog on asterisk-users. Anyway: my answer from the users
list at xorcom:
http://xorcom.com/pipermail/users_xorcom.com/2006-November/000328.html
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406
Hi, I mean that the server finish the action
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
I'm trying to design a mobile-parking infrastructure, (It's for a Finish
University Project)
I made more test ..
When I make a call from a mobile to
I have a need to use a standard analog modem to call out in where asterisk
and a TDM400P are in use. Thru asterisk and the TDM400P, in other words.
Is this even possible? There seem to be some differing opinions. Or is it
only reliably possible to run separate copper for this modem, and
The valet system gets us partway from what I read, but it still uses the
arbitrary number slots. It still requires the user know to transfer a
call to the valet.
no you can park to a specific number (lotname)
exten =
_6XX,1,ValetParkCall(auto|8${EXTEN:1:2}|180|${CALLEDEXTEN}|1|internal)
;
Hello Guys
We are looking for VOIP Cosultants who can successfully build A Scalable
ITSP Architecture Using OpenSource Softwares something like
http://www.skyyconsulting.com/itsp_voip_asterisk.php.
we are looking for some body who can design build a scallable highly
redundant sollution with
Hello Users
I'm planning to do Call Barging and Call snooping , I saw this Feature in
asterisk.org.
This Barging and Snooping are test for is Agents are replying the Answer
or not that I'm guessing
Can anybody help me... this Feature ..
How to do Call Barging and snooping in SIP
I will be out of the office until Tuesday December 5th. , I will checking
my email late in the evenings and will try to respond the next day.
Thank you,
Doug Leber
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
Anselm:
Try using smartCID (www.generationd.com). You'll get the benefit of ranges
of numbers mapping to single ID's (good for corporate blocks), action field
for blocking/accepting calls, etc).
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I use some custom scripts to do database lookups and rewrite CallerID
information. Everything works fine with regard to the CID name, however
my Cisco 7960 and Linksys SPA-942 phones do not display the calling
number. Instead, they display the called number. This makes the phone's
call return
Thanks Steve, that's helpful.
I use Cologne HFC card to connect 2-channel ISDN2e to my PBX. Do I just use
the same card and give it 30 channels instead?
Neil
safeharbour IT Ltd
Your IT Department
tel: 0845 644 3607
fax: 0845 867 2891
mob: 07812 114784
voip: [EMAIL PROTECTED]
email: [EMAIL
On 25 Nov 2006, at 13:34, Neil Tancock wrote:
Thanks Steve, that's helpful.
I use Cologne HFC card to connect 2-channel ISDN2e to my PBX. Do I
just use
the same card and give it 30 channels instead?
Neil
No, you will need an E1 capable card. I use one from Digium, but
there are
Hi,
I have installed Asterisk with a 4 port digium card.
It is working fine but eventhough the sound is clear, the volume is not
loud enough. I have tweaked the txgain and rxgain values but it did not
make much difference.
Please let me know if there are any settings that could help.
Regards
Hello,
The X100P, don't support reverse polarity, I have same problem, then I
bougth a TDM.
Regards
On 11/25/06, txus [EMAIL PROTECTED] wrote:
Hi, I mean that the server finish the action
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
I'm
I am using DIDx.net as my DID provider but they don't seem to get their act
together. A lot of times the phone numbers don't work. How can provide my own
DID, my asterisk server is being hosted at a Data center and has a reliable
vendor that does my termination and do SIP to SIP and have no T1
Hi all,
I have a VOIP phone with the PA1688 chip; my firmware is V1.42.028.
This IP phone is registered in an Asterisk PBX and I've a problem when I
dialing internal number.
If I dial an internal number, like for example 102, the IP phone takes 35
seconds to send the number to Asterisk; here
I cannot access my voicemail and get the following warning in my
console:
[Nov 25 10:26:43] WARNING[5628]: app.c:935 ast_lock_path: Failed to
lock path '/var/spool/asterisk/voicemail/default/8900/Old': File exists
I have also noticed that Asterisk will crash several minutes later
after
On Saturday 25 November 2006 09:38 am, Androtech wrote:
Hi all,
I have a VOIP phone with the PA1688 chip; my firmware is V1.42.028.
This IP phone is registered in an Asterisk PBX and I've a problem when I
dialing internal number. If I dial an internal number, like for example
102, the IP
I think it is wrong. You should specify the next hop with some like this
S0:[EMAIL PROTECTED]
2006/11/23, Larry Alkoff [EMAIL PROTECTED]:
Problem: SPA3000 phone does not ring for incoming PSTN call although I
can dial out.
I set up my Sipura with the Voxilla Wizard which is pretty good but
I will be out of the office until Tuesday December 5th. , I will checking
my email late in the evenings and will try to respond the next day.
Thank you,
Doug Leber
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
I'm having trouble with Cisco 7960 and Linksys SPA-942 SIP phones not
displaying the Caller-ID number. The Caller-ID name is displayed, but
not the number. Instead, the phones always display the value that's set
in the fromuser= parameter in sip.conf. If fromuser= is not set, then
the
I posted a new article on linking Asterisk Servers via SIP instead of IAX on
my web site. It is newbie driven, but I think useful for many since the
information is in one place. Just search 'Linking Asterisk Servers' and all
you will come up with is IAX configurations.
http://www.siliconvp.us
On Sat, Nov 25, 2006 at 07:17:47AM -0500, marvin horst wrote:
The valet system gets us partway from what I read, but it still uses the
arbitrary number slots. It still requires the user know to transfer a
call to the valet.
no you can park to a specific number (lotname)
exten =
Hello everyone!
I have created an background agi which responds to dtmf 0-9, each key
should playback a sound, and it does, but here is the problem.
The sound which is played is just played to the person who touches the
key, not to everyone else in the conference, does anyone know how i
can do
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 have the same problem. Also, the web interface is really awkward, they
don't
have DIDs in the countries where I need them (Chile, for example), and the
quality of the sound is from bad to unusable, even from the US phone they
provide
you for free. If I would have the chance, I would have them
Thanks Alex, I'll try the rapidvox also. I regret ever using didx.net.
-- Original message --
From: Alex [EMAIL PROTECTED]
I have the same problem. Also, the web interface is really awkward, they don't
have DIDs in the countries where I need them (Chile, for example),
I would think an external program that tried to make a sip call and try
diffrent routes etc. would be better or maybe he can add it on.
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, November 25, 2006 11:45 AM
Subject:
pls visit www.inspiresoftbd.com
--
Regards
--
M Emran
E-mail: [EMAIL PROTECTED]
[EMAIL PROTECTED]
Web: www.inspiresoftbd.com
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or
I retested this with 1.4.0-beta3 and I still can't access my
voicemail. I dial the voicemail extension and I just get silence for
a few seconds and it hangs up. HELP! I have 295 messages in my old
mailbox and I want to retrieve my new messages.
Max Bergmann schrieb:
How can i programming a Cisco 7961 to be used as busy lamp field?
my configs :
sccp.conf :
[devices]
type= 7961
tzoffset= 0
autologin = 601
speeddial = *31, Hanna -- other SIP telefon
extensions.conf :
exten = *31,hint,SIP/hanna
exten =
Hello,
Anyone saw asterisknow, ?
Regards
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi All,
I have two old S100 units (the blue ones, not the newer black ones). I
am trying to reset these to factory default using the following
instructions, but it is not working. Does anyone have any other
suggestions to reset this model of the adapter?
Tried this:
1. Remove all of the
We are moving our office, but our PRI isn't moving for a while yet.
I'd like to setup a box at the old office to receive -ALL-- PRI traffic
and send it over an IAX trunk to another Trixbox install at the new
office. Everything should go, period.
Any ideas on a simple dialplan to make
Something like this should do (assuming you get 4 digits for DIDs):
oldoffice:
exten = _,1,Dial(IAX2/whatever/${EXTEN})
exten = _,2,Busy();if you get here then something is wrong with
the connection, so busy out.
newoffice:
exten = _,1,Noop(we got this call from the old office)
On
There was a stale lock file in the mailbox directory. This is a bug
though. Asterisk should clean up all lock files on startup. Lastly,
I can't explain the intermittent crash and wasn't able to catch it
using gdb either.
___
--Bandwidth and
On Sat, Nov 25, 2006 at 08:46:27PM -0500, Darren Wright wrote:
We are moving our office, but our PRI isn't moving for a while yet.
I'd like to setup a box at the old office to receive -ALL-- PRI traffic
and send it over an IAX trunk to another Trixbox install at the new
office.
On Sat, Nov 25, 2006 at 10:57:18AM -0500, Robert La Ferla wrote:
I cannot access my voicemail and get the following warning in my
console:
[Nov 25 10:26:43] WARNING[5628]: app.c:935 ast_lock_path: Failed to
lock path '/var/spool/asterisk/voicemail/default/8900/Old': File exists
Dandling
Jason Adams wrote:
I just upgraded to the latest beta version and I am running into one
problem. We purchased g729a licenses from digium and they aren't
loading anymore. If I roll back asterisk to 1.2.10 the codecs work
fine. I've downloaded the new 1.4 version of the codec from their
44 matches
Mail list logo
