Hi,
my problem in short:
I have a problem reaching a co-worker with the asterisk calling from Germany
to Brazil. With a chance of about 90% I get a chanunavail message from the
asterisk. Normally I try calling him in the afternoon Germany, when he is
awake in Brazil.
so I tried to make calls
Is there not a way to manually configure these phones or at least configure
them to use a diffrent tftp server rather than it attempting to ask the
dhcp/bootp server? For users at home with dinky linksys/dlink modems you
cant set a tftp/bootp server.
--
~Shaun
try to remove manually all parts of old spandsp-installations below /usr/
and /usr/local/ and reinstall both spandsp app_rtxfax
it's likely that you have some parts of the spandsp-0.0.3 left from prior
install which is incompatible to the 0.0.2-versions
[EMAIL PROTECTED] (Robert G. Ristroph)
Hello.
Problem is: Current configuration: (PSTN (UkrTeleCom)) - Е1 -
(TE210P) - T1 - (My own Lucent *MAX 4000*).
I am testing different modem calls:
(My own Lucent *MAX 4000*) - T1 - TE210P - T1 - (My own Lucent
*MAX 4000*) - we have maximum quality of modem connection
(My own Lucent *MAX
I know may be I am disturbing you ,but I am
too thanks full for your help
But can you explain in detailed steps how
to do that
What I understand from you I that I should
put this line at asterisk.conf and its already exist
And create a bash script
#!/bin/bash
digits=$1
In the phones.cfg find the below, you can
change the 8500 to your voicemail exten in extensions.conf of
asterisk
phones.cfg (for polycom)
msg msg.bypassInstantMessage="1"
mwi msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact"
msg.mwi.1.callBack="8500"
/msg
The below will use the
Has somebody done that with a Grandstream GXP-2000 or a BudgetTone
100/101 ?
Has somebody even a list which SIP phones have this funtion?
Regards
Kai
It's called hotline or Private Line Auto Ringdown (PLAR).
SIP: It's a function of the phone, look for hotline in phone docs
Zap:
cd /var/lib/asterisk/agi-bin
touch dtmf2txt.sh
chmod +x dtmfivr.sh
edit dtmf2txt.sh by Your favorite text editor
i'm using mc -e dtfmivr.sh
(this is onlu an example)
#!/bin/bash
digits=$1
number=$2
time=`date`
echo "$time : $1" /home/dtmf2txt/$2.txt
in extensions conf
The setup looks fine, I will run through what I did and the version, there
might be an easier way.
cd /usr/src
svn checkout
http://svn.digium.com/svn/asterisk/team/bweschke/polycom_acd_functions/
asterisk-poly -r 30432
this will checkout the 30432 release and put in the the asterisk-poly
Have you tried this?
2000 = 1234,User Nametz=eastern24
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Hi all,
I just want to setup new voiceprompts for serveral queues in our
asterisk pbx (Version 1.2.41.2.4)
The Problem is, that I don't hear the start (or the first part) of the
voiceprompt.
It makes no differece if I use the Playback or Background Command.
But it makes a difference if the
Okay, i will be one of the 100 answering this question.
what about a wait (2) before the background()?
That should manage your problem.
Mein Name schrieb:
Hi all,
I just want to setup new voiceprompts for serveral queues in our
asterisk pbx (Version 1.2.41.2.4)
The Problem is, that I
Sebastian,
This is possible and most likley the reason. To make sure, check the
location code of the cause IE in your ISDN disconnect message.
You have two options:
1) call your provider and describe your problem.
2) Change your provider
Best regards
Hans
Sebastian Reitenbach schrieb:
Hi,
antonio wrote:
I have a problem: when i make i call from a device h323 to sip, i have no
cdr, and i don't see cdr variables for the channnel ooh323.
Anyone can help me ??
Thanx
On my system, this lives in /var/log/asterisk/cdr-csv/ast_h323.csv.
Regards,
Richard
go here
http://www.voip-info.org/wiki/view/Asterisk+call+forwarding
and look this
*sterisk 1.2*
[macro-stdexten]
;
; Standard extension macro (with call forwarding):
; ${ARG1} - Extension(we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
---BeginMessage---
At /var/lib/asterisk/agi-bin/dtmfivr.sh for example
After that what should I do
read this book?
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
this webpage
http://www.voip-info.org/
regards
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I have setup 3 Linksys SPA-3000 devices to pass/send our analog voice calls
into/out of asterisk. The inbound calls work fine as I have set the
spa-3000's to forward all calls to an extension. I have added them to the
sip.conf as spa-3k1, spa-3k2, and spa-3k3. Is there a way for when some
picks up
Hi,
Just a suggestion,
Why dont you use GROUP function to limit the calls??
Regards
2006/7/10, alexandre - aldeia digital [EMAIL PROTECTED]:
Hi,
I set the sip.conf parameter call-limit=1 to limit outbound calls and
'disable' call waiting.
But internally, I want to enable
Hello all,Is it possible to forward a call received by the asterisk server to a mobile phone? If yes, how? a link or reference is highly appreciated.thanksLito
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To
Get an GSM Gateway from cyber-telecom.net
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lito Lampitoc
Sent: Tuesday, July 18, 2006 4:57
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [asterisk-users] call
forwarding to mobile phone
You need two commands before playing back audio over a line:
Answer()
Wait(2)
On Tuesday 18 July 2006 4:15 am, Mein Name wrote:
Hi all,
I just want to setup new voiceprompts for serveral queues in our
asterisk pbx (Version 1.2.41.2.4)
The Problem is, that I don't hear the start (or the
is there a way I can do call forwarding to mobile phone without using a gsm gateway? my landline is capable of calling a gsm network.On 7/18/06, Sam Tam
[EMAIL PROTECTED] wrote:
Get an GSM Gateway from
cyber-telecom.net
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
]
Hi,
I have two iax server, one is asterisk on ip 192.168.18.8, other is
freeswitch on ip 192.168.18.180. I wanna asterisk iax users could accept
or call freeswitch iax users. How could i do it in asterisk configure?
Bests,
Bobber Cheng
___
I need information / documents or configurations of asterisk with other Telephonic head offices(plants), for your help , thank
sorry for my english, i speek spanish only.
atte,Rodrigo M
On 7/18/06, Lito Lampitoc [EMAIL PROTECTED] wrote:
is there a way I can do call forwarding to mobile
On 7/18/06, Rodrigo Mercado [EMAIL PROTECTED] wrote:
I need information / documents or configurations of asterisk with other Telephonic head offices(plants), for your help , thank
sorry for my english, i speek spanish only.
atte,Rodrigo M
On 7/18/06, Lito Lampitoc
[EMAIL PROTECTED]
Hi,
Does anyone know the usage of ast db?
Does ast db will be useless if I use ARA in asterisk?
unplug
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How do I program the dialplan in extensions.conf to:
(a) try multiple provider to make an outgoing call based on current
latency between my * box and the different providers ?
(b) have if provider 1 goes down, then someone can still call me at
number xxx- but now come in through provider 2
(b) have if provider 1 goes down, then someone can still call me at
number xxx- but now come in through provider 2 or provider 3 ?
Oops, misread this one, yes you can have fallthrough numbers but this
must happen at the provider end, not yours.
___
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Warren (mailing lists) wrote:
Olivier Picquenot wrote:
Zeeshan Zakaria a écrit :
It is CentOS 4.3 and kernel is 2.6.9-34.0.1-smp-i686
Then you might want to use yum to install the apropriate package, the
one that contains the kernel source,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
From: http://www.sineapps.com/news.php?rssid=1377
ISS Xforce has published details of two security issues in Asterisk 1.x
which were fixed in the recently release 1.2.10 version.
Asterisk IAX2 Protocol Denial of Service Attack
Summary:
ISS X-Force
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
trixter aka Bret McDanel wrote:
On Mon, 2006-07-17 at 19:21 +1200, Matt Riddell (NZ) wrote:
It will sometimes tell you that there are modules inside
/var/lib/asterisk/modules which were not compiled for the version you
are compiling. If these are
somebody know a good
way howto select datas from * oracle database inside the
extensions?
for mysql there are
functions. are there for oracle similar ways?
regards
rene
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asterisk-users
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Where can I find information's about maximum data that I can store in
internal * database?
According to the Wiki:
The Asterisk database uses version 1 of the Berkley DB
So, you'd need to look up the information on the Berkeley
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
When asterisk receives those messages you hear when calling an
unreacheable cellular phone it sends a 'connect' over the terminating
PRI line (digium TE410P), making the call seen as billed from customer's
perspective.
Yes, this is
Can someone send me link with instructions how to install Zaptel 1.2.7 on
CentOS 4.3? So far I have used Fedora Core 4 distribution and I didn't have any
problems. I'm planning to use CentOS from now on.
Thank you!
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.:
On Tue, Jul 18, 2006 at 10:20:12AM +1200, Matt Riddell (NZ) wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Warren (mailing lists) wrote:
Olivier Picquenot wrote:
Zeeshan Zakaria a écrit :
It is CentOS 4.3 and kernel is 2.6.9-34.0.1-smp-i686
Then you might want to use yum
AGI() using php, perl, c, bash, python or almost enything You like.
somebody
know a good way howto select datas from * oracle database inside the
extensions?
for
mysql there are functions. are there for oracle similar ways?
regards
rene
Wilson Pickett wrote:
How do I program the dialplan in extensions.conf to:
(a) try multiple provider to make an outgoing call based on current
latency between my * box and the different providers ?
To do this, you need a seperate application that would run something
like fping on all your
On Tue, 2006-07-18 at 06:02 +1200, Matt Riddell (NZ) wrote:
:)
Which applications exist that have been disclaimed, well coded, are
patent unencumbered and are not accepted?
res_js for example, which in my experience on a more or less fair
comparison (the javascript dialplan has more error
Hi,I've connected a GSM gateway to a WellTech 3702A FXO port and created an extension in Asterisk which mobile calls are forwarded to. When a call is made, the GSM module generates a tone instead of picking the digits from the Voip gateway and processing the calls. The VOip gateway keypad type is
(b) have if provider 1 goes down, then someone can still call me at
number xxx- but now come in through provider 2 or provider 3 ?
The way to do this is to use a PSTN based DID provider such as Kall8 and
use a rollover list to route to your DID provider. If your VOIP provider
goes dead,
Hi all
I am facing a strange problem related to CDRs.
I am using asterisk 1.2.4 and AMP
When I setup a 3 way call from a phone, the CDRs are
generated quite strange.
e.g. Phone A calls phone B and Phone C
in CDRs it should appear that Phone A called phone B
and Phone A called phone B
Instead,
Hello everybody,
I is possible to manage multiple call parked per line
.
I mean a caller (agent) have to park more than two
call . It is possible to retrieve caller one ,two
,three, ... with a aplliction which one display the
calling parked to a PC screen or a screen phone .
Regards
Harry
voiplist wrote:
On 7/17/06, Luki [EMAIL PROTECTED] wrote:
We have 6 or 7 SPA-2000's which all work with other installs of
Asterisk but can't get a single one to receive calls using Asterisk
1.2.4.
Ha! You're right. I just got some too and didn't even think of testing
the ringer. Outgoing
On Mon, 2006-07-17 at 21:18 -0700, Luki wrote:
We have 6 or 7 SPA-2000's which all work with other installs of
Asterisk but can't get a single one to receive calls using Asterisk
1.2.4.
Ha! You're right. I just got some too and didn't even think of testing
the ringer. Outgoing calls work
Patrick wrote:
On Mon, 2006-07-17 at 21:18 -0700, Luki wrote:
We have 6 or 7 SPA-2000's which all work with other installs of
Asterisk but can't get a single one to receive calls using Asterisk
1.2.4.
Ha! You're right. I just got some too and didn't even think of testing
the ringer. Outgoing
Hi,
Johann Steinwendtner [EMAIL PROTECTED] wrote:
Sebastian,
This is possible and most likley the reason. To make sure, check the
location code of the cause IE in your ISDN disconnect message.
I have a PRI interface, here ISDN with 30 channels. I am a bit unsure what you
mean with the
I've just found that picking up another phones call via *8# gives me the call
but the other phone keeps ringing. Anyone else seeing this on svn head (updated
last Sunday).
Chris
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Kai Ober wrote:
Has somebody done that with a Grandstream GXP-2000 or a BudgetTone
100/101 ?
Has somebody even a list which SIP phones have this funtion?
SIPura supports it, Cisco ATAs support it. I assume that Cisco phones
support it.
I don't know about Grandstream devices since they
Shaun wrote:
Is there not a way to manually configure these phones or at least configure
them to use a diffrent tftp server rather than it attempting to ask the
dhcp/bootp server? For users at home with dinky linksys/dlink modems you
cant set a tftp/bootp server.
Of course there is. You
Hi,
I'm using Asterisk 1.2.1 on a Debian distro. It happens that sometimes
the caller cannot hear the called party just after the called picks up
the phone. This happens with inbound calls but also with calls from a
SIP phone to another SIP phone. Asterisk and all the SIP phones are on
the
The configuration is this:
H323 -- ASTERISK --- SIP
ooh323.conf
amaflags = billing
[xxx.xxx.xxx.xxx]
type=friend
context=h323-route
ip=xxx.xxx.xxx.xxx
port=1720
allow=all
h323id=example
accountcode=5698742
rtptimeout=60
dtmfmode=rfc2833
extension.conf
...
On 6/11/06, James Harper [EMAIL PROTECTED] wrote:
[.snip.]
My dialplan in the pap2 is:
(:0S0)
Which causes it to dial a '0' to asterisk as soon as I gets picked up.
In my asterisk dialplan it then does a DISA to another context, which
means Asterisk is doing all the dialplan stuff. For what I
Hi,
I have following setup:
++ ++
| asterisk A |-| asterisk B |-- PSTN-gateways ...
++ ++
| .|
| .
=Router (NAT)=.==
|.
Define poor link quality. Are you seeing latency delays, packetization... ?AlexOn 7/18/06, Alexandr Bondar
[EMAIL PROTECTED] wrote:Hello.Problem is: Current configuration: (PSTN (UkrTeleCom)) - Е1 -
(TE210P) - T1 - (My own Lucent *MAX 4000*).I am testing different modem calls: (My own Lucent *MAX
Eric ManxPower Wieling schrieb:
I don't know about Grandstream devices since they are banned from our
network.
Banned? I didn't try any other devices, but whats wrong with the
Grandstreams??
wondering
Kai
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Kai Ober wrote:
Eric ManxPower Wieling schrieb:
I don't know about Grandstream devices since they are banned from our
network.
Banned? I didn't try any other devices, but whats wrong with the
Grandstreams??
Grandstream seems unable to produce stable firmware. They have tried
for *YEARS*
Eric ManxPower Wieling schrieb:
Grandstream seems unable to produce stable firmware. They have tried
for *YEARS* and still people have to try many different versions of
the firmware to find one that actually works in their environment.
okay, i see, thx :)
i will try to remember, if i'm
Below is what I have in my notes regarding CentOS. I am not sure exactly where I originally found this - could have been right from this mailing list actually!
Patch CentOS
spinlock.h file before installing zaptel
Rebuilding Zaptel - Every
time there is a kernel update with yum (which is the
Hi,
Running asterisk ver 1-0-9
Trying to send a call to a mobile phone and playback a
message to the user to press one to accept the call.
If 1 isn't pressed then the call needs to be re-routed
back into the asterisk dialplan.
Tried various macros etc but if one isn't pressed the
call still
Matt Riddell (NZ) wrote:
Warren (mailing lists) wrote:
The proper method is, as root, type:
yum install kernel-devel
The problem is, the kernel headers will have the name 2.6.13-15.8
whereas uname -a will report 2.6.13-15.8-smp.
You may need to create a symbolic link.
Sorry...
yum
Hey guys.
I have a little problem. My provider requires me to make the trunk name
of my SIP connection i2telecom.com (otherwise it won't register).
Unfortunately, this name also becomes the identifier for the connection.
Now, when I want to dial through it Asterisk think I am trying to dial
The show channels output is always truncated.
On 7/17/06, marek cervenka [EMAIL PROTECTED] wrote:
hi,
i have problem with showing actual channels
asteriskshow chanels
SIP/123456789-b6c4b2 [EMAIL PROTECTED] Up Busy()
(last 2 chars are NOT showed)
but the name of channel is longer
We are using an Ateus VoiceBlue to GSM gateway calls on our * 1.0.9 server.
It works perfectly fine, except at peak periods, say, 10 AM and 3 PM. At
that point, calls get dropped (not gateway'd) and Asterisk jumps to the next
priority in the dialplan. Our interpretation of this is that the local
-Original Message-
From: Nic Bellamy [mailto:[EMAIL PROTECTED]
Sent: Monday, July 17, 2006 11:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hitting # to Transfer out of a Queue
Douglas Garstang wrote:
Why? I don't want the
Does anyone know of a way to disable access to the TUI interface
(accessed via ) on the PAP2 devices? I'm looking at using these
devices for lobby and door phones and would like to remove/disable the
TUI interface if at all possible.
--
Jamin W. Collins
Douglas Garstang wrote:
Jul 18 08:27:37 VERBOSE[26274] logger.c: -- Unable to find extension '4' in
context ''
Don't know why the context is '', null.
Silly question,
Has a context been defined in the queues.conf?
; A context may be specified, in which if the user types a SINGLE
;
Hi,
[EMAIL PROTECTED] wrote:
You need two commands before playing back audio over a line:
Answer()
Wait(2)
thanks a lot for answering! This solves my problem perfectly.
ciao,
morel
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On 07/18/06 04:03 Fredrik Emil Jensen said the following:
the packet too, but when the firewall/router loses its table (usually it
will timeout after xx sec/min) you will only be able to dial outgoing
can't you use qualify to get the nat device to keep the mapping ?
--
Regards,
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 18, 2006 8:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hitting # to Transfer out of a Queue
Douglas Garstang wrote:
Jul 18 08:27:37
I have problems compiling zaptel 1.2.6 on my CentOS 4.3. CentOS is
updated and I believe I have installed all the dependencies.
did you fix spinlock.h?
Go into your kernel source directory(or directories if you have more
than one kernel source on your system) and edit the file spinlock.h
Then
I think when a PSTN line says 'Ring' it's simply for aesthetics... The
line is 'answered' the instant * connects to it for two-way audio...
(well not that instant but somewhere in the connection process. When
you are hearing ringing from the PSTN through a zap card, the rings are
coming from
Yes, use a web interface to re-generate features.conf on the fly? hmm..
then tell asterisk to reload it. (Should be OK if the web interface has
some manner of a mutex to keep other instances of the web interface from
stomping on its features.conf and if reload res_features.so does what
you
Buch Bekanntmachung: Der Weg zu VoIP Asterisk von A bis Z
For english see bellow.
Als ich Oktober 2005 angefangen habe mit Asterisk zu arbeiten, gab es nur wenig
zusammenhängende Informationen zu Asterisk. Es gab bereits ein Buch zu
Asterisk, jedoch wurden dort einige Themen ausgelassen.
Has anyone noticed that doing a 'reload' on the Asterisk console clears all the
stats shown by the 'show queues' command?
I'd like to report a bug, but would probably get my head chewed off for not
testing it in the latest SVN code first.
Doug.
___
Put the extensions or a wildcard that matches them into the polycom's
digitmap (dialplan?) I find info about this on page 79 of the 1.5.2 IP
Guide. My offices are like 110, 112, 113, 114, 115, 116 so I use
something like 110|11[2-9]| to match these instantly without softkey
afterwards. or
Are you using the Non-CallManager version?
_
Mobilcom
http://www.mobilcom.net
- Original Message -
From: Tong [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, July 17, 2006 8:56 PM
K, here's something a phone coughed up the other day:
?xml version=1.0 standalone=yes?
PHONE_CONFIG
OVERRIDES reg.1.ringType=17/
/PHONE_CONFIG
and here's another chunk from another phone.
?xml version=1.0 standalone=yes?
PHONE_CONFIG
OVERRIDES reg.1.fwdContact=312
This is an override file. They are an addendum to the main sip.cfg and
phone1.cfg files.
-Original Message-
From: Mojo with Horan Company, LLC [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 18, 2006 10:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On Tue, Jul 18, 2006 at 10:13:58AM +1200, Matt Riddell (NZ) wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
From: http://www.sineapps.com/news.php?rssid=1377
ISS Xforce has published details of two security issues in Asterisk 1.x
which were fixed in the recently release 1.2.10
-Original Message-
From: Massimo Nuvoli [mailto:[EMAIL PROTECTED]
Sent: Monday, July 17, 2006 8:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hitting # to Transfer out of a Queue
Douglas Garstang ha scritto:
I have dialled into
I understand, I was helping Alex to make the connection too. I thought
this was what he was talking about.
Douglas Garstang wrote:
This is an override file. They are an addendum to the main sip.cfg and
phone1.cfg files.
-Original Message-
From: Mojo with Horan Company, LLC
Hi,
Can anyone direct me to where I might find examples of handling
interactive input from a phone using PHP and AGI. I want to have someone
dial an extension and then have the system request input from the user,
take that input and put it into a database.
Thanks
Hello,
Well I had an issue
this morning where the Asterisk process unexpectedly stopped. Below is an
output from the full log:
Jul 18 09:18:51
DEBUG[4892] chan_sip.c: (Provisional) Stopping retransmission (but retaining
packet) on '[EMAIL PROTECTED]'
Request 103: Not Found
Jul 18
Hi all,
Does anyone have any tips on how I would
accomplish a plan where if a user dials 4 digits, then prefix 6 digits, then if
there is a local extension configured for that number dial it, otherwise send
it out another sip gateway ( my pstn gateway)?
Perhaps more specifically, are
A pretty cool package was released on CPAN today from Leif Johansson.
http://search.cpan.org/search?query=Net%3A%3ACSTA
NAME:
Net::CSTA - Perl extension for ECMA CSTA
SYNOPSIS:
use Net::CSTA;
# Connect to the CSTA server
my $csta =
Hi
all,
I'm using spandsp
0.0.2pre21 with tiff 3.7.1 and asterisk 1.0.9
With rxfax
application, everythinghs is ok, but when i try to send a fax whit txfax
applicationthe channel got hangup. I'm testing send fax towards an ATA
grandstrem 488, but if i sent it to a normal fax i got the same
Dan Brummer wrote:
Hello,
Well I had an issue this morning where the Asterisk process
unexpectedly stopped. Below is an output from the full log:
Jul 18 09:18:51 VERBOSE[30430] logger.c: == Parsing
'/var/spool/asterisk/voicemail/default/2195/Old/msg.txt': Jul 18
09:18:51
I think asterisk will take care of this for you. Asterisk will take the most complete match in a pattern match. For example if you have a local extension 1234567890 and a pattern match _XX then Asterisk will match the 1234567890 to the exact match if it exist, if not then it will got the
I'm not completely sure if it will.
As I under stand the qualify options will request a SIP OPTIONS call
every minute from the phone. This solved my NAT problem with one phone
through a linux firewall running ipfilter, I am going to test more phone
through the firewall as I can see it uses the
Thank you Doug for the response. Do you know if the 1.2.10 release
fixes the warm transfer issue I experienced in 1.2.9.1?
Thank you,
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Tuesday, July 18, 2006 10:23 AM
To: Asterisk
Hi all,
I'm trying to setup asterisk and spandsp to recieve fax transmissions. I
got Asterisk to detect fax calls, it even tries to communicate, but the
other side doesn't seem to send the main data. Instead it ends the
communication with hangup. Have anybody got an idea?
Thanks a lot.
Jan
Dan Brummer wrote:
Thank you Doug for the response. Do you know if the 1.2.10 release
fixes the warm transfer issue I experienced in 1.2.9.1?
I have no idea. I would suggest reading the change log at:
http://ftp.digium.com/pub/telephony/asterisk/releases/ChangeLog-1.2.10
Doug
--
Ben
Anyone notice that tf.voipmich.com (ENUM for US toll free service) will
connect you successfully, but then disconnect after what seems like 30
seconds or so? Anyone know what might be going on here? I googled the hell
out of voipmich and did not get very far.
Sincerely,
Brent A. Torrenga
On Tue, 2006-07-18 at 10:29 -0600, Douglas Garstang wrote:
[snip]
exten = oe_ccare,1,NoOp(*** Incoming call from ${CALLERID}
to queue oe_ccare)
exten = oe_ccare,n,Set(TIMEOUT(response)=5)
exten = oe_ccare,n,
call-limit and problem with freezy phones. also freezy zap channels
with x101p card.
Hello all.
I have installed asterisk 1.2.9.1 and zaptel 1.2.6.
I have such configuration:
I have some phones with planet vip-156 with configuration in sip.conf:
[036] ; planet 222
type=friend
host=dynamic
-Original Message-
From: Patrick [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 18, 2006 12:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Hitting # to Transfer out of a Queue
On Tue, 2006-07-18 at 10:29 -0600, Douglas Garstang
DRi == DRi [EMAIL PROTECTED] writes:
DRi
DRi try to remove manually all parts of old spandsp-installations below /usr/
DRi and /usr/local/ and reinstall both spandsp app_rtxfax
DRi it's likely that you have some parts of the spandsp-0.0.3 left from prior
DRi install which is incompatible to
On Monday 17 July 2006 15:14, Joshua Colp wrote:
No, this will set the variable cid_agent to the value 80014054. The
spaces are considered part of the variable name and variable value.
This is insane!
setvar is no different. In the future you can use sip show peer to see what
is happening.
Hello,
I get this error message when trying to route an incoming fax from a packet
based T1 to an EICON board that is connected to an external fax voice mail
server.
Voice calls route to this external server with no error. Both fax and voice
calls that come in a channelized T1 also route
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