Hi,
after I upgraded from 1.2 to 1.4.13 the CLI does not show DTMF anymore,
even at high debug level. Do I need to activate that?
Regards
Volker
--
Volker Sauer * Poststrasse 1/601 * 64293 Darmstadt * Germany
E-Mail/Jabber: volker(at)volker-sauer.de * http://www.volker-sauer.de
Balu Raman wrote:
Omar,
I am hoping that there may be some temp sensor interface that can be
routed to a pc and if the temp falls out of a range, I can have this
event call someone. I know what to do in asterisk to make a call. I
have to do some research. may be, someone has already done a
Tzafrir Cohen wrote:
snip /... is required to properly start zaptel. It will also run
ztcfg. Otherwise
users run into issues where misconfigured zaptel.conf fails loading of a
module. That is a buggy behaviour.
If your card is an analog one, take a look at http://bugs.digium.com/7613
and
Hi,
I am learning Asterisk for a small project.
At this stage I have an AsteriskNOW system running locally. I can call
SIP phone to SIP phone fine, the operator and voice mail work fine,
except some stuttering (probably caused by it running in MS-VPC)
What I need to figure out is...
Try this: http://astrecipes.net/index.php?n=42
l.
On Thu, 18 Oct 2007 06:43:11 +0200, satish patel
[EMAIL PROTECTED] wrote:
I am new in asterisk world can u shortly explian how to create queue and
how to work this ?
David Gomillion [EMAIL PROTECTED] wrote:
On 10/17/07, satish patel
This have been discussed a couple of weeks ago in this list.
You should find useful and detailed answers in archives.
Regards
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To UNSUBSCRIBE or update
Hi,
From personal experience, Asterisk 1.4 IMAP storage seems broken and
unusable.
Is anyone using it successfully ?
This kind of poll would be very useful to estimate is a code rewrite has a
chance to disturb a running system.
I we get no successful report, it would help developers to consider
On Thursday 18 October 2007 04:47:14 Jean-Denis Girard wrote:
Hi list,
Last Friday, an Asterisk server became unresponsive after ~8,5 months of
smooth operation (~32 calls). Server did reply to pings, but no ssh,
no more console login. Also Asterisk no longer took calls, but ISDNguard
There's a large number of gadgets one can buy that work with Skype
through the API. One of the things I'm interested right now is the
ability to properly use a mobile phone headset with a SIP/IAX softphone.
Is there an softphone that emulates the Skype API?
Are there legal implications in
Hi All,
sorry if I post again this e-mail but I think the first one was lost.
I don't know if this is OT but I'm working in my spare time at a small
hardware project that match to what's requested below.
It's a board with Input/Output capabilities and 10Mbps ethernet interface. It
has
I've got someone sat in a home-office with an SPA921 behind NAT, and
most probably a firewall. I've got a STUN-server running, and calling
in from the SPA921 to our Asterisk box works fine - though I had to
open the firewall for UDP traffic on port 1-2.
Calling from our Asterisk to the
Hello Marco,
could you explain how you did the interfacing to the Asterisk PBX? does
your prototype speak SIP to receive commands?
Thanks
l.
On Thu, 18 Oct 2007 12:27:33 +0200, marcotasto [EMAIL PROTECTED]
wrote:
Hi All,
sorry if I post again this e-mail but I think the first one was
Hi list.
I was googling about this subject and found a message to this list
asking the same thing I want, but see no response. So, I'm reposting
it in case someone has a solution for this.
I'd like to know if there's some way to detect (not block) a collect
call in a ISDN E1? I need this cause I
On Thu, 18 Oct 2007 00:32:44 -0400, Brian Capouch wrote:
shadowym wrote:
Size/Speed/write cycles have gone way up, price has gone way down. More
common than CompactFlash and no need for an adapter. So is it feasible to
run an Asterisk server on something like this? With a MTBF of 1million
I'm in process of transitioning a number of offices to a hosted virtual
pbx from Junction Networks. It's a combination of OpenSER and Asterisk.
They have a nice click-to-call extension for Firefox, but I need the
equivalent for IE so that it can work with our CRM system. Junction
told me that they
Just for fun.
http://news.bbc.co.uk/1/hi/magazine/7049642.stm
Dave
*
This email is intended solely for the use of the individual to whom
it is addressed and may contain confidential and/or privileged
Kevin,
What kind of device are you using on the fridge ?
Dovid
- Original Message -
From: Kevin Withnall
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, October 18, 2007 12:04 AM
Subject: Re: [asterisk-users] Refrigerator Alarms
We use similar
Hi Balu,
http://www.digitemp.com/ has all the info you need. Cost of hardware
around US$70. Will give you pretty fridge temperature graphs too!
You can easily hack some of Brian's scripts to do different levels of
temp alarm and trigger calls in Asterisk.
regards,
Drew
Balu Raman wrote:
Hi Lenz.
What I did to interface asterisk with the door opener was to implement, in the
board, a custom embedded server that receives and parses a set of UDP packets
containing a known data in it.
In the dialplan I then call a perl AGI script that sends UDP packets in the
correct sequence and
On Thu, 2007-10-18 at 09:25 +0200, Volker Sauer wrote:
after I upgraded from 1.2 to 1.4.13 the CLI does not show DTMF anymore,
even at high debug level. Do I need to activate that?
You need to enable DTMF debugging in logger.conf, then type logger
reload at the Asterisk CLI for those changes to
Hi,
I've been reading all I can on Google (and Asterisk TFOT book) looking for
ideas on how to implement an automated blacklist feature.
I would like to automatically blacklist a incoming number based on timestamp
and count information.
For example, if I get a prank call from the same number 5
Just a note for those of you trying to install Asterisk 1.4.x under
Mandriva 2008.
I wasn't able to get make menuselect to work. It kept telling me that I
didn't have ncurses installed (I have ncurses-5.6-1) even though the
configure script said it found it. I eventually went into the
On Thu, 2007-10-18 at 16:02 +0300, Brian Hutchinson wrote:
I would like to automatically blacklist a incoming number based on
timestamp and count information.
For example, if I get a prank call from the same number 5 times within
15 minutes, I want my dialplan to automatically blacklist
Balu Raman wrote:
Hi,
I want asterisk to call a person on the phone for monitoring the
refrigerator storing vaccines.
I am clueless where to look. Can someone clue me in ?
OWFS for sure! Here is a screenshot of a program I created a couple
years ago to monitor refrigerators/warmers. It
We have had a few different times when a user has forwarded their phone
to himself. This has overloaded the communications to our operator panel
(FOP). One user should not be able to effect the whole phone system!
Is there a way that the number of times that a call can be forwarded
could be
[Sorry if this arrives more than once. I have sent this twice and it
never arrived, despite other messages getting to the list O.K.]
---
Hello,
I would like an incoming caller to be able to choose from the menu
options in my extension.conf below. Once They have dialled the
appropriate
Is there a function to write the timestamp of the first call? I started
thinking about AGI and PHP/MySQL since that is what I'm familiar with. I
couldn't find methods to write timestamp info to AstDB or if I could ... how
to read it back and compare it to time now to decide to increment my
Sir,
I am having runing asterisk 1.4 server which is runing without any problem now
i want to receive fax over sip extension. how it is possible and what the
change i have make in extensions.conf.
Thanks
Rajeev.
___
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On Thu, 2007-10-18 at 16:51 +0300, Brian Hutchinson wrote:
Is there a function to write the timestamp of the first call?
You can use the built-in channel variable ${EPOCH}, which will give you
the current time in Unix timestamp format (number of seconds since Jan
1, 1970). Then you can do
On Thu, Oct 18, 2007 at 01:04:06PM +0100, Cartwright, Dave wrote:
Just for fun.
http://news.bbc.co.uk/1/hi/magazine/7049642.stm
It's Asterix != Asterisk. Though named after *.
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406
On Thu, 2007-10-18 at 09:08 -0400, Doug Lytle wrote:
I wasn't able to get make menuselect to work. It kept telling me that I
didn't have ncurses installed (I have ncurses-5.6-1) even though the
configure script said it found it. I eventually went into the
menuselect directory and did a
It's not technically complex to do - you can probably use the astdb for
that, or store all incoming numbers with timestamp in MySQL and run
something like:
SELECT count(*) 5 AS blacklisted
FROM incoming_calls
WHERE callerid = 12345
AND timestamp DATE_SUB( NOW(), INTERVAL 15 MINUTE )
you
Hi List,
I am from Peru, I have installed an asterisk server in my company with
digium card E1 TE120P, I am having issues when i make calls, here the
error from my server
[Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No
channel type registered for 'Zap'
[Oct 18 09:13:50]
Not using IMAP storage here, although that was one of the primary
drivers for upgrading to Asterisk 1.4. Why?
The short answer is that it's too confining. There are too many caveats
that don't fit into our existing IMAP structure and make the entire
project rather iffy. Security is a
Make sure chan_zap.so is loaded.
/b
On Oct 18, 2007, at 9:34 AM, Pablo Almido wrote:
Hi List,
I am from Peru, I have installed an asterisk server in my company with
digium card E1 TE120P, I am having issues when i make calls, here the
error from my server
[Oct 18 09:13:50]
I feel more comfortable with MySQL ... just need to learn how to get the
dialplan to use it.
Also figure out the pro's/con's to MySQL vs AstDB. If I used MySQL then I
could put myphpadmin and get a pseudo GUI to manipulate the blacklist
database for almost no effort so that is another reason for
On Thu, 2007-10-18 at 09:34 -0500, Pablo Almido wrote:
[Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No
channel type registered for 'Zap'
[Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full:
Unable to create channel of type 'Zap' (cause 66 - Channel not
implemented)
How do you know that the call is a prank call, an not just someone
that likes calling your company alot... ?
If you just want a database of callerid's to block, here is what I
have used, I hope it helps some
My SQL table looks has 4 columns id (autoincrement), callerid,
blockenabled
On Thu, Oct 18, 2007 at 08:35:01AM +0100, Alan Lord wrote:
Tzafrir Cohen wrote:
snip /... is required to properly start zaptel. It will also run
ztcfg. Otherwise
users run into issues where misconfigured zaptel.conf fails loading of a
module. That is a buggy behaviour.
If your card is
Tzafrir Cohen wrote:
On Thu, Oct 18, 2007 at 01:04:06PM +0100, Cartwright, Dave wrote:
Just for fun.
http://news.bbc.co.uk/1/hi/magazine/7049642.stm
It's Asterix != Asterisk. Though named after *.
In Britain, it's called humour :-)
regards,
Drew
--
Drew Gibson
Systems
Yeah I would use MySQL as well for more or less the same reasons. Using
MySQL right from the dialplan is not very elegant but it's pretty simple -
see http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MYSQL
Thanks
l.
On Thu, 18 Oct 2007 16:49:48 +0200, Brian Hutchinson
[EMAIL
Why would a config error stop the module from loading? That seems
like a suboptimal behavior.
/b
On Oct 18, 2007, at 9:50 AM, Jared Smith wrote:
That would seem to indicate that the chan_zap.so module isn't being
loaded. What happens if you type module unload chan_zap.so and then
module
On 10/18/07, Olivier [EMAIL PROTECTED] wrote:
Hi,
From personal experience, Asterisk 1.4 IMAP storage seems broken and
unusable.
Is anyone using it successfully ?
I've read blog entries that indicate that people have used it successfully
but I have not been able to get it to connect to the
Yes, the module is load
# asterisk -r
ippbx*CLI module show like chan_zap.so
Module Description
Use Count
chan_zap.soZapata Telephony
0
1 modules loaded
ippbx*CLI
ippbx*CLI
2007/10/18, Brian West [EMAIL PROTECTED]:
Make sure chan_zap.so is
Apart from religious grounds (!), is there any pros or cons why I should
choose one over the other for a new install of asterisk ?
Julian
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asterisk-users mailing list
To
On 10/18/07, Olivier [EMAIL PROTECTED] wrote:
Hi,
From personal experience, Asterisk 1.4 IMAP storage seems broken and
unusable.
Is anyone using it successfully ?
This kind of poll would be very useful to estimate is a code rewrite has a
chance to disturb a running system.
I we get no
I have unload and load the module, it is output
ippbx*CLI module unload chan_zap.so
== Unregistered application 'ZapSendKeypadFacility'
ippbx*CLI module load chan_zap.so
== Registered application 'ZapSendKeypadFacility'
== Parsing '/etc/asterisk/zapata.conf': Found
[Oct 18 10:46:38]
On Thu, Oct 18, 2007 at 12:22:24PM -0400, [EMAIL PROTECTED] wrote:
Just 5 months ago CENTOS started to use Linux 2.6
Centos 4 (based on RHEL4) used kernel 2.6 as well.
It was released over two years ago (it has 2.6.9).
--
Tzafrir Cohen
icq#16849755
I'm sorry I call bullshit on this one. CentOS has been 2.6 for some
time.
/b
On Oct 18, 2007, at 11:22 AM, [EMAIL PROTECTED] wrote:
Just 5 months ago CENTOS started to use Linux 2.6 one of the
reasons I'd abandoned for SuSE a while back.
Julian Lyndon-Smith wrote:
Apart from religious grounds (!), is there any pros or cons why I
should choose one over the other for a new install of asterisk ?
I doubt it. A distro is a distro.
/Per Jessen, Zürich
We use only openSUSE.
___
On Wed, 2007-10-17 at 15:09 -0700, shadowym wrote:
I have read all the wiki's and blogs and how to links about Bluetooth but so
far no luck. I can confirm that CentOS5 sees my Bluetooth adapter and my
cell phone. No Joy on Asterisk 1.4. The information out there is kind of
confusing as
On 10/18/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
Apart from religious grounds (!), is there any pros or cons why I should
choose one over the other for a new install of asterisk ?
Julian
SuSE is known for using the latest packages with each release, and
RHEL/CENTOS are known for
On Thursday 18 October 2007 09:49:48 Brian Hutchinson wrote:
I feel more comfortable with MySQL ... just need to learn how to get the
dialplan to use it.
See func_odbc.conf.
--
Tilghman
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On Thu, Oct 18, 2007 at 01:05:16PM +0300, Cosmin Prund wrote:
There's a large number of gadgets one can buy that work with Skype
through the API. One of the things I'm interested right now is the
ability to properly use a mobile phone headset with a SIP/IAX softphone.
Is there an
On Thu, Oct 18, 2007 at 10:53:15AM -0500, Pablo Almido wrote:
I have unload and load the module, it is output
ippbx*CLI module unload chan_zap.so
== Unregistered application 'ZapSendKeypadFacility'
ippbx*CLI module load chan_zap.so
== Registered application 'ZapSendKeypadFacility'
On 10/18/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Oct 18, 2007 at 12:22:24PM -0400, [EMAIL PROTECTED] wrote:
Just 5 months ago CENTOS started to use Linux 2.6
Centos 4 (based on RHEL4) used kernel 2.6 as well.
It was released over two years ago (it has 2.6.9).
Never trust
Hi Carlos,
The method commonly used is flash E1 to block this calls.
A Better way could be detect the category of call.
I don´t know if there is a way to get the call category in extensions as we
can get the CALLERID.
Collect calls have B-8 category in Brazil.
Thanks.
Luis A P Barbosa
Hello I have a question about incoming calls on TDM400P cards. I want to
know why an incoming call appear in a sorpresive way on a phone that I
pickup to call out. I am using ChanIsAvailable to check those lines ( Zap
channels )that are free. I have four lines connected to my TDM400P card and
when
Drew Gibson wrote:
Tzafrir Cohen wrote:
On Thu, Oct 18, 2007 at 01:04:06PM +0100, Cartwright, Dave wrote:
Just for fun.
http://news.bbc.co.uk/1/hi/magazine/7049642.stm
It's Asterix != Asterisk. Though named after *.
In Britain, it's called humour :-)
regards,
Drew
On Thu, 2007-10-18 at 16:02 +0300, Brian Hutchinson wrote:
Hi,
I've been reading all I can on Google (and Asterisk TFOT book) looking
for ideas on how to implement an automated blacklist feature.
I would like to automatically blacklist a incoming number based on
timestamp and count
Okay, so we are planning for the future here where I work so we are
trying to do testing ahead of time as we might be setting up a
satellite campus that would need its own Asterisk phone system but
still tied into our main campus phone system. This much we have
accomplished. We have a
Atis Lezdins a écrit :
Yup, it's also a problem for me, but it haven't ever crashed server. It just
makes specific remote process unresponsive. There's a patch for 1.4, but i
guess it wouldn't be hard to backport it for 1.2
http://bugs.digium.com/view.php?id=10847
you might also want
SIP wrote:
Drew Gibson wrote:
Tzafrir Cohen wrote:
On Thu, Oct 18, 2007 at 01:04:06PM +0100, Cartwright, Dave wrote:
Just for fun.
http://news.bbc.co.uk/1/hi/magazine/7049642.stm
It's Asterix != Asterisk. Though named after *.
In Britain, it's
On Thu, Oct 18, 2007 at 06:25:39PM +0200, Per Jessen wrote:
Julian Lyndon-Smith wrote:
Apart from religious grounds (!), is there any pros or cons why I
should choose one over the other for a new install of asterisk ?
I doubt it. A distro is a distro.
Well, no.
/Per Jessen, Zürich
We
I run this command
[EMAIL PROTECTED] ~]# cat /proc/zaptel/1
Span 1: WCT1/0 Wildcard TE12xP Card 0
IRQ misses: 40
1 WCT1/0/1
2 WCT1/0/2
3 WCT1/0/3
4 WCT1/0/4
5 WCT1/0/5
6 WCT1/0/6
7 WCT1/0/7
8 WCT1/0/8
Jay R. Ashworth wrote:
On Thu, Oct 18, 2007 at 06:25:39PM +0200, Per Jessen wrote:
Julian Lyndon-Smith wrote:
Apart from religious grounds (!), is there any pros or cons why I
should choose one over the other for a new install of asterisk ?
I doubt it. A distro is a distro.
Well, no.
I doubt it.
hxxp://boycottnovell.com/2007/10/02/opensuse-103-release/
Original Message
Subject: Re:[asterisk-users] centos 5 vs OpenSuse 10.3
From: Per Jessen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: 18/10/2007 11:25 a.m.
Julian Lyndon-Smith wrote:
On Thu, Oct 18, 2007 at 02:54:36PM -0500, Perssy Llamosas wrote:
I doubt it.
hxxp://boycottnovell.com/2007/10/02/opensuse-103-release/
The above page (even with the proper protocol name) is very low on
actual facts and reasonings. It mentions nothing specific to openSUSE
10.3 .
So let's get
Glare that's what it's called, if the number you advertise as your
business number is zap/1 then use zap/G1 to dial out, otherwise use
zap/g1 to dial out. This will reduce but not eliminate the problem.
http://www.telos-systems.com/techtalk/gldefs.htm#Glare
On 10/18/07, Gustavo Gonzalez [EMAIL
Here's what I'm looking to do
exten = 10,1,Dial(SIP/1000SIP/1001,15,wW)
exten = 10,2,Voicemail(u1000)
This should ring both phones and they should keep ringing for the
alloted time before moving on. However, it appears that if one of the
phones is Busy, it returns with a busy and moves on
Hi all, i have been using asterisk for a few years but i am about to do
my first t1 setup. After terrible quality issues between two business
locations, we have decided to purchase a point to point t1 from the
local phone co. The internet is too crappy, too much lag, queing and
jitter. Most
I hope you have better success than I did, my problem was
not so much with asterisk in particular but 64-bit in general.
Examples of problems using CentOS 4.5 on x86_64
- many problems loading php5 mysql from package
repositories.
- a few asterisk functions don't work, eg STRFTIME()
Hi Luis.
On 10/18/07, Luis Antonio Prata Barbosa [EMAIL PROTECTED] wrote:
Hi Carlos,
The method commonly used is flash E1 to block this calls.
Yes.. But I need to detect it so I can set a billing on it.
A Better way could be detect the category of call.
I don´t know if there is a way to get
Hello all,
i would like to have references so i'm giving free help
for any project (commercial or public).
regards,
--
Your next Partner !
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asterisk-users mailing list
To
ZAP/G1 will dial starting from channel 1. ZAP/R1 will dial starting
from the last channel of the group.
Pablo, please tell us what version of Linux and which distribution are
you using. Maybe for the time being try the stock asterisk of your
distro or the one they provide in the buildservice?
On
On 10/18/07, Cosmin Prund [EMAIL PROTECTED] wrote:
One of the things I'm interested right now is the ability to
properly use a mobile phone headset with a SIP/IAX softphone.
Thats a function of the bluetooth stack. Go ahead and pair your
bluetooth headset to your PC it will work like any
Hi list,
I just installed 64 bit Linux, and ready to install Asterisk through
source on it. Are there any settings have to change to build 64 bit
Asterisk? Thnx a million.
___
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I doubt it.
boycottnovell.com/2007/10/02/opensuse-103-release/
Original Message
Subject: Re:[asterisk-users] centos 5 vs OpenSuse 10.3
From: Per Jessen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: 18/10/2007 11:25 a.m.
Julian Lyndon-Smith wrote:
Apart
I doubt it.
boycottnovell.com/2007/10/02/opensuse-103-release
Original Message
Subject: Re:[asterisk-users] centos 5 vs OpenSuse 10.3
From: Per Jessen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: 18/10/2007 11:25 a.m.
Julian Lyndon-Smith wrote:
Apart
Pablo - You said you have 1/2 E1 - which half???
That might be your problem. Unless 1/2 E1 means something else...
Asterisk normally dials out on the low end unless you specify
G instead of g ??? or something like that.
Brett
___
--Bandwidth and
While I am a fan of CentOS some pople just take it tooo far.
- Original Message -
From: Perssy Llamosas [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, October 18, 2007 9:54 PM
Subject: Re:
We used to use CentOS 4 here but about 6-8 months ago we found that
they were too slow with updates their repos for some of the 3rd party
software that we were developing. We switched to SuSe 10.2 and haven't
looked back. However Asterisk works equally well on both. Just pick your
[EMAIL PROTECTED] wrote:
Do you think there might be a working t.38 implementation in ABE 1.4?
Even passthru Between endpoints and gateways with Asterisk in the
middle?
ABE does not contain functionality that isn't in an open source version
of Asterisk, except for license- or
Did you set NAT Keep Alive Enable: = Yes for the line in question in
the SPA's configuration?
On 10/18/07, Per Jessen [EMAIL PROTECTED] wrote:
I've got someone sat in a home-office with an SPA921 behind NAT, and
most probably a firewall. I've got a STUN-server running, and calling
in from
Hi,
Has anyone had any great difficulties with QoS using the second ethernet
phone in these Polycom phones for desktop machines in a converged
network? I had heard that these can cause difficulties when used in
this manner. I have always tried to persuade customers to go with 2
ethernet drops
I know I can get free DID's with SIP, is anyone giving out free DID's with
IAX?
Thanks in advance,
In the words of the great jbot in the #asterisk channel on irc.freenode.net
Dovid ~ygwypf
jbot well, ygwypf is You Get What You Pay For. If the sole factor in your
decision to purchase a
The Asterisk.org development team has announced the release of Zaptel
versions 1.2.21 and 1.4.6. These releases contain many bug fixes as well
as performance enhancements (Too many to list here). A couple of major
changes: there is an update to the Octasic API version as well as a
- Original Message -
From: Don Pobanz [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, October 18, 2007 3:38 PM
Subject: [asterisk-users] Limit number of times a call can be forwarded
We have had a few
Nope it should just work. Just finished setting up 1.4 for the first
time in a while and just works. Been running 1.2 for the longest time
and same thing.
On 10/18/07, Wai Wu [EMAIL PROTECTED] wrote:
Hi list,
I just installed 64 bit Linux, and ready to install Asterisk through
source on it.
So your problem is:
-- IAX2/alanb-3 answered SIP/101-081d1050
Except the remote end didn't actually answer the call? The problem is
your remote end... its answering the call. All the IAX hardphones I've
seen don't seem to be the highest of quality honestly.
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