- Original Message -
From: Dan Austin [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, August 19, 2007 7:58 PM
Subject: Re: [asterisk-users] Change Packetization Time
Dovid wrote:
Does anyone know if it
On Sun, Aug 19, 2007 at 09:00:33PM -0400, Matthew Brothers wrote:
Questions:
1. Is the wiki DUNDi example and the dundi.conf file too difficult to
follow for new users?
I wouldn't exactly say that it is too difficult but that the target
audience for the default examples is not the
You may want to start from here: http://astrecipes.net/index.php?n=204
l.
On Sun, 19 Aug 2007 00:46:45 +0200, Ade Vickers
[EMAIL PROTECTED] wrote:
Hi...
I have what is, I am sure, a relatively common straightforward problem
(no, NOT that kind of problem!)... I'm trying to hook two
2007/8/13, Paul Hayes [EMAIL PROTECTED]:
It's not currently possible but Siemens are working on new firmware for
at least the S450IP model which will support auto-config using http.
I'm not sure when it's due for release though.
Thanks for the tip !
Directly asking to Siemens (
Did you try T.38 ?
These PAP2 boxes should be able to benefit from Asterisk T.38 pass through
capabilities.
You would then have to install a T.38 termination device, such as Linksys
3102 :
PSTN Linksys 3102 --- LAN - PAP2 --- Fax
machine
Cheers
Hi List;
I am using Firefly softphone Version 1.9.9 Build 4521
and I select IAX protocol and did the configuration in
Network1 (and I checked the Active checkbox) as
following:
Server: 192.168.8.4
username: iax2user1
password: password
In the Asterisk, I did the following configuration on
the
Dears
Any one succeeded to make Redundancy / Failover with asterisk 1.4.9 on
centos with kernel 2.6.9-55.EL.
Can you please send me the documentation link on how to or write down how to
.
Regards
*
No employee or agent is
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Kashif Naeem wrote:
Hello All
We have developed an application for Home Delivery Restaurants using
Asterisk, Java (JSP/ JSF) and MySQL. Here is listing of its features. If
someone is interested then we can provide him more details.
- POP
On Mon, 20 Aug 2007, bilal ghayyad wrote:
Hi List;
I am using Firefly softphone Version 1.9.9 Build 4521
and I select IAX protocol and did the configuration in
Network1 (and I checked the Active checkbox) as
following:
Server: 192.168.8.4
username: iax2user1
password: password
In the
I am running r79979 of Asterisk Trunk, and I am having problems trying to use
app_queue.so.
I want to use the extension 510 to be a line where users can call technical
support.
Extensions 511 and 512 are used by the operators to dynamically make
themselves a Queue Member or not.
So,
On 8/20/07, Tim Groeneveld [EMAIL PROTECTED] wrote:
I am running r79979 of Asterisk Trunk, and I am having problems trying to use
app_queue.so.
I want to use the extension 510 to be a line where users can call technical
support.
Extensions 511 and 512 are used by the operators to
2007/8/13, Eric Chamberlain [EMAIL PROTECTED]:
What you describe is doable; we have a number of device configuration
wizards.
But it is generally easier to use the device's bulk provisioning methods,
like https an XML configuration file to the device. The provisioning
settings a pretty
Thanks Steve and Mitcheloc,
in fact i was think in something more obsolet like connect via serial/usb
cable the cell to the asterisk box. Never thought in the SIP stack of new
Nokia's but i will start looking for info about this. If you [Steve] know of
a good written material of interest please
Hello All,
Is CALLERID() setting broken in 1.4.4?
My small dialplan :
[testclid]
exten = _0.,1,Set(CALLERID(all)=Ben Jacob 988077)
exten = _0.,n,Dial(SIP/${EXTEN})
Correct me if I am wrong, Set(CALLERID(all) above supposed to change the
display name as above(Ben Jacob) and change the From URI
Well chan_bluetooth is really amazing (especially if your phone does not
support SIP).
You connect your phone via bluetooth to your asterisk box and it becomes
a channel type. You can use it as an extension(FXS) or a phone line
(FXO). I believe you can send and receive SMS through the
Also,
. if I use Remote-party-id header, can it be different from the 'From' URI?
. If yes, how do you achieve this in Asterisk?
. What(From or Remote-party-id) is used by clients to show as the CLI of
the caller?
if I am not mistaken, Remote-party-id is for network elements to confirm
Dear Gordon;
Thanks a lot for your email.
I need one more tracing tool, how can I know the used
port of the IAX on teh Asterisk and wethor the
listening on that port is successully done (ready to
receive on that port)?
About the firewall, actually the client PC and
Asterisk on the same LAN (my
On Mon, 20 Aug 2007, bilal ghayyad wrote:
Dear Gordon;
Thanks a lot for your email.
I need one more tracing tool, how can I know the used
port of the IAX on teh Asterisk and wethor the
listening on that port is successully done (ready to
receive on that port)?
Use
netstat -lnveep
to
On Monday 20 August 2007 8:16:32 pm Atis wrote:
On 8/20/07, Tim Groeneveld [EMAIL PROTECTED] wrote:
Can you also provide output of show queues and show channels? Plus
the logfile of dial to 511.
I'm using QueueAdd after AgentCallbackLogin (trough manager API).
Maybe you need to use
JR Richardson wrote:
Questions:
1. Is the wiki DUNDi example and the dundi.conf file too difficult to
follow for new users?
2. Does the complexity of the DUNDi setup discourage you from using it
or even attempting to configure it?
3. If there was a simple tutorial, step by step guide
On 8/20/07, Tim Groeneveld [EMAIL PROTECTED] wrote:
On Monday 20 August 2007 8:16:32 pm Atis wrote:
On 8/20/07, Tim Groeneveld [EMAIL PROTECTED] wrote:
Can you also provide output of show queues and show channels? Plus
the logfile of dial to 511.
I'm using QueueAdd after
I wouldn't exactly say that it is too difficult but that the target
audience for the default examples is not the average person/entity
that could make use of the power inherent with DUNDi. When an
average * user/admin wants to use DUNDi they will want to start out
small and local rather
On 8/19/07, Steve Turner [EMAIL PROTECTED] wrote:
I would like to send Multimedia Messaging (MMS) email (gateway) to my
cell
phone and have the from and subject be the callerid/calleridnam
information
from the voice mail message.
voicemail.conf lets you change the from and subject line,
Hi all,
I am seeing the following messages on my asterisk cli:
Got SUBSCRIBE for extension [EMAIL PROTECTED] from 192.168.0.158, but there is
no
hint for that extension.
I dont know what it means. I believe it has something to do with realtime
extensions or hints. i know about realtime
On Mon, Aug 20, 2007 at 09:26:00AM -0400, Matthew Brothers wrote:
I wouldn't exactly say that it is too difficult but that the target
audience for the default examples is not the average person/entity
that could make use of the power inherent with DUNDi. When an
average * user/admin
Hello,
I have a small LAN network connected through an Asterisk Server. When I try to
make a call between two of the user pc's on this network I get a 401
Unauthorized error.
Would anyone know how to remove the Asterisk Authorization/Authentication? I
am not sure if this can be done with
the following link show more than one methods to connect 2 asterisk servers:
http://www.voip-info.org/wiki-Asterisk+-+dual+servers
On 8/19/07, Ade Vickers [EMAIL PROTECTED] wrote:
Panic over...
I have a weird network problem (now solved), whereby incoming packets
arrived directly to the
well i have mentioned earlier that it happens to only one user. all of the
other users limits are working fine.
In asterisk 1.4.0, this was a bug as it happened to every user. But then i
upgraded to 1.4.2 and it was gone. It was working fine since then but
recently this problem again showed up
Tim Groeneveld wrote:
I am running r79979 of Asterisk Trunk, and I am having problems trying to use
app_queue.so.
I want to use the extension 510 to be a line where users can call technical
support.
Extensions 511 and 512 are used by the operators to dynamically make
themselves a Queue
On 8/20/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
(But then again, if anybody wishes to write something, I won't say no)
So why all the verbiage? JR offered a valuable service to the
community, I see no downside to this. If anyone doesn't care for the
idea they can just ignore it. A lot of
Hi,
Has anyone experienced a tool providing system administrators with IMAP
voicemail statistics ?
The main usage is know the amount of time between the moment a message is
dropped in voicemail and the moment this message is read (heard).
I guess this is required to help management to pin point
I have a costumer with a Siemens PBX installed, and I would like to setup a
Asterisk system that would act as a kind of Proxy between the Siemens PBX and
the operator network.
The current setup is:
Siemens PBX 2*PRI - Operator
what I want is:
Siemens PBX 2*PRI - Asterisk BOX - Operator
For
Well only certain situations expose this bug. I am able to reproduce
this bug in two instances. One is with the Adtran Total Access 900
series when it receives a fax call it sends a INVITE to the Asterisk.
When Asterisk receives this INVITE it changes the call from peer to
user, so by the
On 8/20/07, Gustavo Felisberto [EMAIL PROTECTED] wrote:
I have a costumer with a Siemens PBX installed, and I would like to setup
a
Asterisk system that would act as a kind of Proxy between the Siemens PBX
and
the operator network.
The current setup is:
Siemens PBX 2*PRI - Operator
On 8/20/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
In an attempt to understand why there are no better docs inside
asterisk.
Well, we're all on the same page then :)
My opinion, summed up into a sentence would be that the people who
create the code have *mostly* commented the main conf files
I am trying to figure out how long a caller waited in queue for someone
to answer versus how long they stayed on the phone after the answer. I
am assuming that the duration is the total talk time and that the
billsecs are the total time in queue. is this correct? or should i be
deducting the
Hi List,
I wonder that how I can check that FAX is delivered successfully or not,
in my dialplan while using TxFAX.
Which variable contain fax status? as PHASEESTATUS and PHASEESTRING in
Callweaver.
Regards
Nasir Iqbal
___
--Bandwidth and
On Mon, 2007-08-20 at 11:45 -0500, Jordan Novak wrote:
I am trying to figure out how long a caller waited in queue for
someone to answer versus how long they stayed on the phone after the
answer. I am assuming that the duration is the total talk time and
that the billsecs are the total time in
On Mon, 2007-08-20 at 11:45 -0500, Jordan Novak wrote:
I am trying to figure out how long a caller waited in queue for
someone to answer versus how long they stayed on the phone after the
answer. I am assuming that the duration is the total talk time and
that the billsecs are the total time in
From: Hans Feringa [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: Re: [asterisk-users] 99
Excuse me if I recently posted on this, but I cannot find it, in my, or the
list archives.
Is it possible, when transferring a call, to set the user ID to that of the
outgoing number instead of the incoming number? I believe the answer is
(was) yes.
New twist, does it matter what the
Steve Murphy wrote:
So, the billsec field is usually SMALLER than the duration.
Usually? Are there situations when the duration is
smaller than billsec?
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not
comment the line secret=201 or 202
Then it'll not ask for 401Autherization.
Regards,
Kesh
[EMAIL PROTECTED] wrote:
Hello,
I have a small LAN network connected through an Asterisk Server. When I try to
make a call between two of the user pc's on this network I get a 401
Has anyone ever tried using a Nokia phone with SIP client as channel for
Asterisk? I mean i would like to receive calls to the mobile on
asterisk and use the Nokia phone to place calls to cell destinations.
I have enough Nokia E60's to do that and it would circumvent the need for
On 8/20/07, Carlos Chavez [EMAIL PROTECTED] wrote:
On Mon, 2007-08-20 at 11:45 -0500, Jordan Novak wrote:
I am trying to figure out how long a caller waited in queue for
someone to answer versus how long they stayed on the phone after the
answer. I am assuming that the duration is the total
I did post recently, under another subject line.
But would appreciate some response, as some are telling a client that this is
not possible.
joe a.
On 8/20/2007 at 1:57 PM, Joe acquisto [EMAIL PROTECTED] wrote:
Excuse me if I recently posted on this, but I cannot find it, in my, or the
Am Montag, den 20.08.2007, 13:57 -0400 schrieb Joe acquisto:
Excuse me if I recently posted on this, but I cannot find it, in my, or the
list archives.
Is it possible, when transferring a call, to set the user ID to that of the
outgoing number instead of the incoming number?
I believe the
On the SIP side of things, we have a how-to guide for the Nokia E series and
Asterisk.
http://voxilla.com/voxilla-stories/voxilla-how-to-guides/using-the-nokia-e-series-phones-with-asterisk-865.html
--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/
_
Using the phone itself as a GSM-SIP gateway is not possible with the native
VoIP application, but it looks like it should be possible with a custom
application for the phone.
--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/
-Original Message-
From:
Does anybody have realtime queue members working? Not the queues
themselves, just the members. I have realtime working for voicemail and
sippeers, but I can't get queue members to work. Here is what I have:
res_mysql.conf:
[general]
dbhost = 127.0.0.1
dbname = ASTERISK
dbuser = myuser
dbpass
John C. Wolosuk Jr. [EMAIL PROTECTED] writes:
ok this is a wired problem. when i use X-Lite - after i register with
asterisk X-lite sends a subscribe/notify request to asterisk to
determine if the account has any messages waiting.
if i create a sip.conf account using:
user 12345 with a
John C. Wolosuk Jr. wrote:
ok this is a wired problem. when i use X-Lite - after i register with
asterisk X-lite sends a subscribe/notify request to asterisk to
determine if the account has any messages waiting.
if i create a sip.conf account using:
user 12345 with a voicemail box 12345 -
Kyle Sexton wrote:
Maybe we can convince Digium to have an indemnification program for
people who purchase the business edition! :)
This is already in place. Asterisk Business Edition is delivered under a
traditional commercial software license that includes warranty
protection and patent
On 8/17/07, Andrew Ruthven [EMAIL PROTECTED] wrote:
Hi,
With more recent version of v1.2 and with v1.4 are things like the
AstManProxy still recommended if you want to have a bunch of
applications talking directly to Asterisk?
If you're looking to have a number of clients monitor events,
On my Asterisk installation, I've had to roll back to Zaptel 1.2.19.
When I use 1.2.20, I get very bad echo problems.
It seems to work OK if I use a quieter-than-normal speaking voice, but
at a sufficient sound level, the echo breaks through and then never
goes away.
The problem goes away if
Peder @ NetworkOblivion wrote:
Does anybody have realtime queue members working? Not the queues
themselves, just the members. I have realtime working for voicemail and
sippeers, but I can't get queue members to work. Here is what I have:
res_mysql.conf:
[general]
dbhost = 127.0.0.1
On Tuesday 21 August 2007 12:32:12 am Mark Michelson wrote:
When users call 510 then, it actually does ring everyone who has called
511.
The problem is when the operator comes to pick up the call. The operator
hears nothing, and the user still hears the Music on Hold. Not only that,
yep. [EMAIL PROTECTED] to be exact. it's the same in both configs, the
essentially the only things i changed is the [name] username= from
12345 to jwolosuk. i should note my version is 1.4.9 and i am serving
the configs via asterisk real time.
---
John
no dice. :-(
---
John C. Wolosuk Jr.
Unix/Linux Systems Administrator
Academic Computing Communications Center
University of Illinois @ Chicago
E-Mail: jwolosuk at uic dot edu
---
Kyle Sexton wrote:
John C.
Anthony Francis wrote:
There is no queue_members file, asterisk doesnt know hat you are
talking
about, you would have to #include queue_members from inside that queue
definition.
Huh? How is including a file going to make realtime access the
queue_members database via mysql?
Russ Price wrote:
On my Asterisk installation, I've had to roll back to Zaptel 1.2.19.
When I use 1.2.20, I get very bad echo problems.
You should be trying 1.2.21.1
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve
I think that revision 80086 in the 1.4 subversion branch would fix this.
Julian.
Peder @ NetworkOblivion wrote:
Does anybody have realtime queue members working? Not the queues
themselves, just the members. I have realtime working for voicemail and
sippeers, but I can't get queue members
I have a TE405/TE410P card that was working on 1.0.X
I upgraded the OS to Centos 4.5, Updated asterisk to 1.4 and zaptel to
1.4.5 and libpri.
I copied all the zaptel and zapata and extensions.conf files from 1.0
I did update extensions.conf from 1.0 to 1.4 commands.
I cannot get the card to
Is there a way, other then recoding the entire voicemail application, to
pass dialplan variables to the voicemail application and to the email
notifications of new voicemail.
For example in our small tech support queue i would like to pass the ticket
number with the email notification that a new
While I don't have an answer on how to access channel variables from
voicemail.conf, for the problem you mention this should help.
Change CALLERID(name) to your ticket number and then use VM_CIDNAME in
the subject line.
If you don't want to lose the original CIDNAME then just add your
ticket
After rethinking.
I'm not sure if this works, but please report back after testing.
The idea would be that the CIDNAME should not be in the subject just
the ticket number, and the ticket number should not be in the email
body just the CIDNAME.
Please try the following and report back.
exten =
Dear All,
I would like to get community's feedback with regard to RAID1 ( Software or
Hardware) implementations with asterisk.
This is my setup
Motherboard with SATA RAID1 support
CENT OS 4.4
Asterisk 1.2.19
Libpri/zaptel latest release
2.8 Ghz Intel processor
2 80 GB SATA Hard disks
256 MB RAM
AHH lol i can't believe i didn't see/think of that :) thanks .. it's a quick
hack but it works for what i need right now. Maybe this can be a feature
request for the voicemail app
On 8/20/07, C F [EMAIL PROTECTED] wrote:
While I don't have an answer on how to access channel variables from
I have done some work with Siemens hipath systems in the past - just
watch out the pridialplan and it's friends.
PaulH
On Mon, 2007-08-20 at 16:17 +0100, Gustavo Felisberto wrote:
I have a costumer with a Siemens PBX installed, and I would like to setup a
Asterisk system that would act as a
While hardware RAID tend to be more reliable, it is not always
possible to properly monitor hardware raid in a linux system, unless
you write your own code.
Consider this:
~# cat /proc/mdstat
Personalities : [raid1]
md0 : active raid1 sdb2[2](F) sda2[1]
76139968 blocks [2/1] [_U]
unused
Thanks, that fixed it. I just looked up the bug and then patched my
1.4.10.1 source with it and it appears to work as there are now queue
members listed.
http://bugs.digium.com/view.php?id=10424
I can't believe nobody else ran into this. Basically the issue was that
you couldn't use
Okay for a quick report back, that all seems to work...
Thanks a lot.
Not much to report back other then that :)...
On 8/20/07, C F [EMAIL PROTECTED] wrote:
After rethinking.
I'm not sure if this works, but please report back after testing.
The idea would be that the CIDNAME should not be
Hello,
A client has asked for Two B channel Transfer capability (known as
TBCT or 2BCT, similar to other features such as ECT, RTL and Q,SIG
Path Replacement) in a new Asterisk system and so I researched the
capability and came up with quite a few gaps in documentation.
From what I've gathered,
Steve Murphy wrote:
How about this one: from an extensions.conf that someone posted on the
internet, I think, and I converted to AEL; I'm sorry, but I can't find
the original author.
(If anybody can find his post, I'd love to give him credit.) I did test
this out,
and it works; just put a
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