It just works. Simply transfer your call to the desired extension and
let Asterisk take care of the details.
--
Cosmin Prund
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of DiegoF
Sent: Friday, May 25, 2007 12:04 AM
To: Asterisk Users Mailing List - Non-Commercial
On Fri, 25 May 2007, Cosmin Prund wrote:
It just works. Simply transfer your call to the desired extension and
let Asterisk take care of the details.
Indeed. A key appeal of Asterisk does lie precisely in that it
abstracts, to a considerable degree, the chore of dealing with the
Just add
include = featuremap
in extensions.conf
i think this should help.
On Wed, May 23, 2007 at 12:59:39PM +, khawla khawla wrote:
I am trying call transfer with asterisk. blind transfer (#) is working
perfectly, but attended transfer doesn't fonction (*2).
I don't know
You may also want to have a look at our suite QueueMetrics, that is
deployed in hundreds of CCs worldwide, is very flexible and is free for
small CCs. See http://queuemetrics.com
I hope this helps
l.
On Fri, 25 May 2007 02:02:18 +0200, Senad Jordanovic [EMAIL PROTECTED]
wrote:
bilal
Well, I've run out of ideas :)
On 5/22/07, Vincent [EMAIL PROTECTED] wrote:
Must be one of those problems that are solved in 2 seconds with the
right click or line in a configuration file... when you know what
you're doing :-)
___
--Bandwidth and
This is correct. To download firmware from cisco.com you need an account
with the respective service agreement.
When buying phones make sure you buy them with the respective firmware
already present.
AFAIK this agreement for a single phone is affordable though.
Andreas
2007/5/25, [EMAIL
Quick reminder that this exists and is today.
* see http://x2z.eu for instructions
Maybe JerJer (aka Put down the crack pipe) will be there to comment
on the about Nufone and their plans in Canada and elsewhere?
___
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Good morning,
We are in the process of setting up a similar combination - Mitel 3300 ICP
+ Asterisk. We chose to use SIP for interconnectivity for ease of
configuration inhouse as getting the local Mitel support rep is tough and
they balk at any configuration beyond the basics and you know we
Hello All
This or similar topics have already been mentioned but without any
solution yet.
I have built a oneway conference system for a client using one caller's
input
and broadcast it to all the other participants using app_meetme. E.g. one
talker
multiple listeners.
Unfortunately some of
Atlanticnynex wrote:
You can specify different options to start meetme with (announcements,
etc.)
in the dialplan by having a separate extension for the person who wants to
here the sounds. I've never tried this, but I think it should work.
Tried that, problem is that it plays no sounds to
Hi,
I followed the how-to from
http://www.alcatelunleashed.com/viewtopic.php?f=44t=840
All works fine except for Asterisk-Alcatel calls.
Actually, calls from Asterisk to analog extensions on
the Alcatel work.
However, calls from Aserisk to digital extensions on
the Alcatel 4400 do NOT work.
I
On Fri, 2007-05-25 at 11:13 +0100, Joesph wrote:
Good morning,
We are in the process of setting up a similar combination - Mitel
3300 ICP + Asterisk. We chose to use SIP for interconnectivity for
ease of configuration inhouse as getting the local Mitel support rep
is tough and they balk
Hi try put Speaq
speaQ is a VoIP softphone which runs on either Windows Mobile 5.0 or Sharp
Zaurus Linux. It can be used to make and record Internet phone calls using
any SIP compliant Internet Phone Server. The free Beta Trial Version which
can be downloaded from this page, lets you record
Hello,
I'm trying to match a number in international format, like +49...
The regexp string ^\+49 doesn't work. Both in $[+49... : ^\+49]
and ${REGEX(^\+49 ${NUMBER})}.
The error is: WARNING[12486]: func_strings.c:138 regex: Malformed input
REGEX(): Invalid preceding regular expression.
Hi,
I want to use festival with asterisk to play a text with sable tags.
have some body any idea about it
Nasir Iqbal
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Oops here is the link
http://qtechinc.com/speaq_download.htm
--Giridhar Bandi
On 5/23/07, ram [EMAIL PROTECTED] wrote:
On 5/23/07, Philipp von Klitzing [EMAIL PROTECTED]
wrote:
Hi!
Googling arround I found a number of pocket pc softphones. Of those I
was only able to install
Today I was speaking with my telephony provider. They said that they are
sending to my asterisk a 183 message and that should be enough to hear
the ring-back tone. Do I have to change something in the configs to
have this option interpreted?
Thanks in advance
On Thursday 24 May 2007 09:44, dima
Indeed, but I can't access the page... very strange!
do I need to send the config files?
2007/5/25, Russell Bryant [EMAIL PROTECTED]:
Tim Verscheure wrote:
yes!!
2007/5/21, Guilherme Góes [EMAIL PROTECTED]:
Did you acess the page at port 8088 ? I.e.: http://192.168.0.1:8088
For the first time on Wednesday, I noticed SIP-SIP echo...very weird.
Normally, I run G729 between all my Grandstream GXP2000 phones, but I tried X-Lite to call one of my Grandstream. This of course switched my codec over to GSM.
I had headphones on the PC and the mic muted. When I spoke in the
On 25 May 2007, at 04:57, Carlos Chavez wrote:
We have an installation with around 50 sip phones but only 5
of those are
hardware. There are three Grandstream phones, one Snom and one
PAP2T. We are
running Asterisk 1.2.8 with an E1 (R2). Only the hard phones are
having
problems
Doug, thanks, can you send me vm-callout.sh as I cannot find it using
google.
Regards,
Paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Thursday, May 24, 2007 9:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I am trying to test the call parking, but It doesn't fonction :(these are my
config files.extensions.conf:include=parkedcallsexten =
4000,1,Dial(SIP/4000,60,tT)exten = 4001,1,Dial(SIP/4001,60,tT)exten =
4002,1,Dial(SIP/4002,60,tT)In features.conf:[general]parkext = 700
Parking a call is a transfer to a parked extension. You need to flash, dial the extention 700 and listen for the parked number. You cannot just press 700 during the call.
I am trying to test the call parking, but It doesn't fonction :(
these are my config files.
extensions.conf:
Alex thank you for your response.
In this case we are USING INBAND, though I have tried both. Nothing works.
Yes ser is configured with mediaproxy.
Thank you,
-JK
JK,
In-band or RFC2833 DTMF signaling?
Also, unless you have SER configured with a media proxy, the actual call
is not
On Fri, 2007-05-25 at 13:23 +0100, Tim Panton wrote:
On 25 May 2007, at 04:57, Carlos Chavez wrote:
Who is hearing the echo ? Your users or the party at the far end ?
Actually they say that both sides of the conversation hear echo and/or
distortion.
--
Telecomunicaciones Abiertas
On Thu, 2007-05-24 at 11:37 -0700, pedro noticioso wrote:
hi guys, is it possible to do a basic 3-or-more-way
conference call when the phones dont support it? I am
fully aware of this concept on expensive phones like
this one:
Grandstream GXP 2000 -Conference call 3-way
Eugen Rogoza wrote:
Hello,
I'm trying to match a number in international format, like +49...
The regexp string ^\+49 doesn't work. Both in $[+49... : ^\+49]
and ${REGEX(^\+49 ${NUMBER})}.
The error is: WARNING[12486]: func_strings.c:138 regex: Malformed input
REGEX(): Invalid preceding
On Fri, 2007-05-25 at 08:22 -0600, Anthony Francis wrote:
Eugen Rogoza wrote:
Hello,
I'm trying to match a number in international format, like +49...
The regexp string ^\+49 doesn't work. Both in $[+49... : ^\+49]
and ${REGEX(^\+49 ${NUMBER})}.
The error is: WARNING[12486]:
Paul Aviles wrote:
Doug, thanks, can you send me vm-callout.sh as I cannot find it using
google.
That's just a script that I created. Nothing special. Attached below:
#!/bin/sh
cd /usr/local/bin
/bin/touch /usr/local/bin/$1.out.call
/bin/touch -r /usr/local/bin/$1.out.call -F 150
Anthony Francis wrote:
Eugen Rogoza wrote:
Hello,
I'm trying to match a number in international format, like +49...
The regexp string ^\+49 doesn't work. Both in $[+49... : ^\+49]
and ${REGEX(^\+49 ${NUMBER})}.
The error is: WARNING[12486]: func_strings.c:138 regex: Malformed input
I came across an issue where the user interface I was using (FreePBX?) to enter
expressions was silently swallowing backslash characters (this wasn't regexp,
but my dialplan had to add a SIP header with a semicolon in - that was falling
foul of the comment character matching for the user
JK,
On Fri, 25 May 2007, JK wrote:
Alex thank you for your response. In this case we are USING INBAND,
though I have tried both. Nothing works. Yes ser is configured with
mediaproxy. Thank you,
Depending on the exact acoustic qualities of the end-to-end path,
in-band can be problematic.
Hi All,
Has anyone gotten the polycoms or the linksys phones to accept oprtion
66 on the dhcp request for the address of the tftp config server?
We have the dhcp server issuing the proper IP of the tftp server, but
the phones just sit there and never try to contact the tftp server for
their
On Fri, 2007-05-25 at 08:14 -0700, Steve Langstaff wrote:
I came across an issue where the user interface I was using (FreePBX?) to
enter expressions was silently swallowing backslash characters (this wasn't
regexp, but my dialplan had to add a SIP header with a semicolon in - that
was
Hi List,
I am currently using the H parameter in the dial command. The issue that I am
having is that if the user is calling an ivr that requires him to press the *
key then the call gets hung up on. How would I go about changing it so that the
user will have to press say ** for the H parameter
Hi,
I want to start recording the caller automatically when the receptionist
transfers a new sales lead to 567. I don't want the receptionist to have to
press *1 manually for automon. Can someone recommend how best to accomplish
this?
exten = 567,1,Set(CALLERID(name)=SALES CALL)
exten =
JR Richardson wrote:
Hi All,
Has anyone gotten the polycoms or the linksys phones to accept oprtion
66 on the dhcp request for the address of the tftp config server?
We have the dhcp server issuing the proper IP of the tftp server, but
the phones just sit there and never try to contact
On Fri, 25 May 2007, Dovid B wrote:
I am currently using the H parameter in the dial command. The issue that
I am having is that if the user is calling an ivr that requires him to
press the * key then the call gets hung up on. How would I go about
changing it so that the user will have to
Hi:
Does anybody know of a TDM interface card for *digital Centrex* that
will work in Asterisk? We're not talking about BRI, here -- the lines
have Nortel digital sets on them, and we want to run them into an
Asterisk PBX.
Centrex is more widely used in NAm.
-Stephen-
Mandeep Singh Bhabha
Just add
include = featuremap
in extensions.conf
i think this should help.
This fixed the issue for me also. I did not realize that this was needed
to make these features work. It does not appear anywhere in
extensions.conf.sample for 1.2.18.
Don Pobanz
On
On Fri, 25 May 2007, Alex Balashov wrote:
Make it so it accumulates states of at least two contiguous
DTMF-containing frames and makes the inference if they come within a
certain interval of each other.
Or, if you're not particular about *, make it a single # or something
else instead,
Alex Balashov wrote:
JK,
On Fri, 25 May 2007, JK wrote:
Alex thank you for your response. In this case we are USING INBAND,
though I have tried both. Nothing works. Yes ser is configured with
mediaproxy. Thank you,
Depending on the exact acoustic qualities of the end-to-end path,
I have a scenario here with IP phones, on a private 192.168 network
connecting to an Asterisk box, also on the same 192.168 private network.
We'd like to have the Asterisk box also be able to send traffic to the
public IP space. For this, we would need to multi-home the box, and put
two network
Hello,
I am working on an outdial project and the Asterisk box is connected
behind a PBX via SIP. When a call from the Asterisk box is routed out
over the PRI attached to the PBX I am not getting proper call progress.
The PBX is indicating that the call is answered while it is still
ringing at
At 23:40 5/24/2007, JK wrote:
Hello asterisk-users list.
I have been scratching my head for almost a week. We are trying to
set a service with a company (ip=XXX.XXX.XXX.XXX) and dtmf is not working.
Have had better luck with SIP Info.
___
J French wrote Friday, May 25, 2007 10:54 AM
I want to start recording the caller automatically when the
receptionist transfers a new sales lead to 567. I don't want
the receptionist to have to press *1 manually for automon.
Can someone recommend how best to accomplish this?
exten
Is there any built in functionality when using Originate to retry a call
based on the DIALSTATUS? Similar to the .call file where you can set
max retries and time between them?
I've tried putting the logic in an outbound context/macro, but it just
times out if the time between retries is too
I don't think that it is true that it will only listen on the first
interface.
I've built many boxes with the configuration you describe. In many
networks the phones are on their own vlan with the PBX and the PBX is
also connected to the gateway router acting as the gateway for the phone
Hi:
Can anyone recommend a good ISDN BRI interface card for Asterisk? I know
there are a few out there.
-Stephen-
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On 25 May 2007, at 16:44, Eugen Rogoza wrote:
On Fri, 2007-05-25 at 08:14 -0700, Steve Langstaff wrote:
I came across an issue where the user interface I was using
(FreePBX?) to enter expressions was silently swallowing backslash
characters (this wasn't regexp, but my dialplan had to add a
On Fri, 25 May 2007, Barry Porch wrote:
I am using BackgroundDetect to wait for the greeting (hello, etc.)
following the answer. I just don't know how to deal with the variable
number of rings.
This problem or may not have a good solution, but if it does, it's
probably bound up in some
On 5/25/07, Douglas Garstang [EMAIL PROTECTED] wrote:
I have a scenario here with IP phones, on a private 192.168 network
connecting to an Asterisk box, also on the same 192.168 private network.
We'd like to have the Asterisk box also be able to send traffic to the
public IP space. For this,
I have a box doing this, Asterisk listens on either IP unless you bind to a
specific interface.
On 5/25/07, Douglas Garstang [EMAIL PROTECTED] wrote:
I have a scenario here with IP phones, on a private 192.168 network
connecting to an Asterisk box, also on the same 192.168 private network.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
JR Richardson
Sent: Friday, May 25, 2007 11:31 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom or Linksys phones bootp
tftp config setup
Hi All,
Has anyone gotten the
List users,
Using Asterisk in an inbound call center environment has led us to
pushing the limits of vertical scaling. In order to treat each caller
fairly and to utilize our agents as efficiently as possible, it is
desirable to configure each client as a single queue. As far as I know,
I have asterisk 1.4 ,the function that I am using in extensions.conf is not
functioning
Its was functioning on asterisk 1.2.further more
cdr_addon_mysql.so cdr_csv.so cdr_custom.so cdr_manager.so
cdr are loaded
Is there any missing module ?
Function IS
Hi there.
Just curious if you've checked out Linux clustering software such as
OpenSSI ( http://www.openssi.org/ ) and run Asterisk on it? It features
a multi-threaded cluster-aware shell (and custom kernel) that will
automatically cluster-ize any regular Linux executable (such as the main
Hi, all. I'm checking in about an issue that has been mentioned here a
few times, but to which I can't seem to find a solution for a very
present need.
The summary is that we'd like to have a queue that rings logged-in
agents in the same order every time, based on penalty, in a way that
Chris Hardie wrote:
Hi, all. I'm checking in about an issue that has been mentioned here a
few times, but to which I can't seem to find a solution for a very
present need.
The summary is that we'd like to have a queue that rings logged-in
agents in the same order every time, based on penalty,
Chris,
If it's not something that the Asterisk queuing algorithms provide out of
the box, it may be wortwhile to consider deputising that level of logic
to AGI in the dialplan.
I'm also not sure if it's possible to make AGI hooks in the queue config
directly, let alone bring them to bear on
I am not sure about the details of the DHCP protocol and what polycom want
but in a linux box using dhcp3 server this works for me:
option tftp-server-name tftp://10.102.1.1;;
Justin
On 5/25/07, Watkins, Bradley [EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL
In reference to an old post from 2002:
http://www.marko.net/asterisk/archives/0203/0103.html
How does one go about doing this?
Also, what is the present status of the OpenSS7 stack in Asterisk? What
can it do now?
And is there any possibility in the future of developing a DS3 card
for it,
On 5/25/07, Alex Balashov [EMAIL PROTECTED] wrote:
In reference to an old post from 2002:
http://www.marko.net/asterisk/archives/0203/0103.html
How does one go about doing this?
I think what mark was referring to there is dynamic spans. They
actually work over a standard ethernet network.
On Fri, 25 May 2007, William Moore wrote:
I think what mark was referring to there is dynamic spans. They
actually work over a standard ethernet network. They are configured in
zaptel.conf and zapata.conf just like any other zaptel device.
Interesting! So Zaptel does have native TDMoE
I have just upgraded my Polycom 501's from 1.6.2.0041 to 2.1.0.2708 to
get the microbrowser.
Almost everything is fine except when receiving calls from a BT200
(1.1.14 and earlier) the Polycom rings but when answered, drops out and
the BT200 gets a busy tone.
I have many PAP2T's and SPA3000's
Can you attach the trace, or at least let me know what DHCP server you
are using? The Polycoms, at least, require that DHCP option 66 use the
Microsoft-style DHCP behavior and actually encode it as a DHCP option
(rather than a BootP header). On certain DHCP servers (Nortel at least
I can say
Hi All,
Call comes into Asterisk
Asterisk answers and Dials SIP Phone
SIP phone has call forward enabled to a long distance number
Asterisk receives a SIP response 302 Moved Temporarily back from phone
Asterisk then forwards inbound call to 'Local/[EMAIL PROTECTED]' thanks to phone
2 problems
List users,
This post contains the benchmarks for Asterisk at low call volumes on
similar single and dual-core servers. I'd appreciate it greatly if you
took the time to read and comment on it.
Thank you,
Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
Sean M. Pappalardo wrote:
Just curious if you've checked out Linux clustering software such as
OpenSSI ( http://www.openssi.org/ ) and run Asterisk on it? It
features a multi-threaded cluster-aware shell (and custom kernel) that
will automatically cluster-ize any regular Linux executable (such
Hi JR -
Has anyone gotten the polycoms or the linksys phones to accept oprtion
66 on the dhcp request for the address of the tftp config server?
Yes. I've gotten this to work successfully using Polycom phones with
DHCP from Cisco routers and firewalls (I generally don't use ISC's
DHCP).
http://www.thetechgeek.com/content/product.php?pid=25311cid=
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
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Hello,
We bought some Linsys WRTP54G-NA boxes which have WIFI, 4-port, 2 SIPs...
The two SIP ports work on A* if you call one line to talk to the other in
the same box.
When we pick up a line, dial to another phone via the A* server, this will
ring at the other end... But, when you pick up
Hi Marco -
We bought some Linsys WRTP54G-NA boxes which have WIFI, 4-port, 2 SIPs...
The two SIP ports work on A* if you call one line to talk to the other in
the same box.
When we pick up a line, dial to another phone via the A* server, this will
ring at the other end... But, when you pick
Doug,
I have tried that. I am testing this with verizon DID. Any have done the
setup with them??
I am still dead in water,
PLEASE PLEASE PLEASE..
Thank you,
-Jai
--
Message: 3
Date: Fri, 25 May 2007 12:03:40 -0500
From: Doug [EMAIL PROTECTED]
Subject: Re:
On 5/25/07, Matthew J. Roth [EMAIL PROTECTED] wrote:
List users,
This post contains the benchmarks for Asterisk at low call volumes on
similar single and dual-core servers. I'd appreciate it greatly if you
took the time to read and comment on it.
Are you recording memory figures as well and
Doug,
I have tried that. I am testing this with verizon DID. Any have done the
setup with them??
I am still dead in water,
PLEASE PLEASE PLEASE..
Thank you,
-Jai
--
Message: 3
Date: Fri, 25 May 2007 12:03:40 -0500
From: Doug [EMAIL PROTECTED]
Subject: Re:
Very good... by the way, I'm studing electrical engineering and I've
chosen asterisk scalation as my final graduation project. I hope do a
similar work within and asterisk cluster.
On 5/25/07, William Moore [EMAIL PROTECTED] wrote:
On 5/25/07, Matthew J. Roth [EMAIL PROTECTED] wrote:
List
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