I think RegEvents in SIP is the keyword for such feature.
But unfortunately, it's not implemented in Asterisk, AFAIK.
2008/8/6 Stefan Gofferje [EMAIL PROTECTED]
Hi,
is there meanwhile the possibility for some actions besides dialling in *?
Namely, I would like that if a remote IAX or SIP
have done it, and its working fine. but still expecting to receive some new
ideas.
On Wed, Aug 6, 2008 at 2:12 PM, Rizwan Hisham [EMAIL PROTECTED]wrote:
hi all,
i just finished developing some incoming call features in a macro. that
macro gets executed everytime an incoming call is received
Hi all,
[snip]
For this reason I set all the 4 ports of Digium's card in NT mode
(Philips can not do this). Then i opportunely
edited /etc/misdn-init.conf and /etc/asterisk/misdn.conf. In fact, when
I run the command misdn shows stacks in * CLI, I can see all ports in
NT (PTP) mode.
have
I agree bandwidth is the limit, however the reason to use IAX is it is
saving bandwidth.
I am runging 2 Trixbox CE with IAX over a 2 Mbit line.
I have never had any isues with the IAX trunk.
I wish that I could get son good IAX phones to the office, tan would we skip
on Trixbox and run the phones
Hello All,
I'am doing a video call between two Video Phones, and i see
that Asterisk is stripping the fmtp parameters for the h263 video line in
SDP.
For example a line similar to the below is stripped,
a=fmtp:xx CIF=4;QCIF=2;F=1;K=1
Asterisk is configured NOT to
Mattias Andersson wrote:
I agree bandwidth is the limit, however the reason to use IAX is it is
saving bandwidth.
I am runging 2 Trixbox CE with IAX over a 2 Mbit line.
I have never had any isues with the IAX trunk.
I wish that I could get son good IAX phones to the office, tan would we
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED],
altrimenti vi risponderò al mio rientro.
Dimitri Osler
I
Hi All,
On a 1.4.15 system, I've a context as below, where I need to catch some
specific US ranges and dial direct via SIP rather than a PSTN trunk.
But the logic always goes via the International Trunk and I cant see
why...
[local]
exten = _00165011091[45]0-9],1,NoOp(I AM HERE)
exten =
On Thu, 7 Aug 2008, Mattias Andersson wrote:
I agree bandwidth is the limit, however the reason to use IAX is it is
saving bandwidth.
I am runging 2 Trixbox CE with IAX over a 2 Mbit line.
I have never had any isues with the IAX trunk.
I wish that I could get son good IAX phones to the
On Thu, 07 Aug 2008 00:28:29 -0500, Karl Fife wrote:
Example: Last week there was talk about Polycom's HDVoice
technology, and the term was being used interchangeably with G.722. In
fact there are important distinctions, but someone listening might
presume that the information was correct and
On Wed, 6 Aug 2008, Fidel Garcia wrote:
Guys I have been reading for days on how to get this to work with asterisk
and for some reason every time I call the call goes to intercom. I know I
must be doing something wrong with the way I am adding the steps to my call;
I am not familiar with
Hi!
I´m using cmd Dial from an EAGI script
My problem is that I cant hear the warning sound
EAGI script:
SET VARIABLE LIMIT_WARNING_FILE beep
SET VARIABLE LIMIT_PLAYAUDIO_CALLEE yes
SET VARIABLE LIMIT_PLAYAUDIO_CALLER yes
EXEC DIAL Zap/g1/676354|20|HL(132000:3000:3000)
CLI:
-- AGI Script
Hi,
I am trying to install a Voicemail on PRI after a redirection on an away or
a busy (a normal call which is redirected to voicemail in fact) but I can't
find the function in Asterisk which allow me using the phone number of the
callee (because I have only the number of asterisk and of the
I am using asterisk 1.4.21 with outgoing call files.
I am call a line that is busy as you can see below.
How can my AGI ask what the status of the last call was
so I can tell if there was NO ANSWER or it was BUSY?
Thanks,
Jerry
-- Attempting call on SIP/401 for
[EMAIL PROTECTED]:1 (Retry
Jerry Geis wrote:
I am using asterisk 1.4.21 with outgoing call files.
I am call a line that is busy as you can see below.
How can my AGI ask what the status of the last call was
so I can tell if there was NO ANSWER or it was BUSY?
Thanks,
Jerry
-- Attempting call on SIP/401 for
On Wednesday 06 August 2008 22:35:01 Positively Optimistic wrote:
We've searched but thus far have not successfully found a solution for
this…
We're looking for a way to set a variable using get digits for a DISA
application. Sometimes we're away from the office and get a voicemail
that I
Jerry Geis wrote:
Jerry Geis wrote:
I am using asterisk 1.4.21 with outgoing call files.
I am call a line that is busy as you can see below.
How can my AGI ask what the status of the last call was
so I can tell if there was NO ANSWER or it was BUSY?
Thanks,
Jerry
-- Attempting
Thanks for your reply!
Just so you have a better understanding of what I am trying to accomplish.
The distinctive ring is working fine with Family, however, the intercom
configuration that I am currently testing makes all my calls and intercom
call. It does not matter if I call using Dial or Page
Call files spawn a completely new channel that your AGI most likely
isn't going to be able to track. Since the call is a completely new
channel, the DIALSTATUS variable for this channel will not be visible to
your original channel. You may want to look at using the Originate
action
This problem was fixed when I upgraded my box to version 1.4.21.1
Thanks everyone,
Daniel
On Mon, Jun 30, 2008 at 2:01 PM, Chento Arohuanca [EMAIL PROTECTED]wrote:
I forgot it!, I'm using Asterisk 1.4.19.1 version.
On Mon, Jun 30, 2008 at 1:47 PM, Chento Arohuanca [EMAIL PROTECTED]
wrote:
Hello--
Why do I target chan_sip for so much effort? Because,
it seems to me, chan_sip is probably the most used channel
driver in the asterisk community!! (and, of course,
the zap/dahdi driver, is also pretty popular)
I haven't had time to follow up on chan_sip, and I probably
won't for
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED],
altrimenti vi risponderò al mio rientro.
Dimitri Osler
I
Jason Parker wrote:
I just wanted to post this so that it was out there and Googleable. Hopefully
it will save other people a bit of time.
If you have a Cisco phone (I was testing with a 7970, though presumably it
would
affect 7960 and others as well) that is looping trying to fetch the
On Thu, 7 Aug 2008, Fidel Garcia wrote:
Thanks for your reply!
Just so you have a better understanding of what I am trying to accomplish.
The distinctive ring is working fine with Family, however, the intercom
configuration that I am currently testing makes all my calls and intercom
call.
On 10:59, Thu 07 Aug 08, Fidel Garcia wrote:
Thanks for your reply!
Just so you have a better understanding of what I am trying to accomplish.
The distinctive ring is working fine with Family, however, the intercom
configuration that I am currently testing makes all my calls and intercom
I added the configuration as you suggest but now the phone does not do
intercom. I tried Dial and Page in the gxp2000 but everything goes out as
Dial.
Here is the extensions.conf now
exten=s,1,SIPAddHeader(Alert-Info: http://127.0.0.1\;info=Family)
Steve Murphy wrote:
Hello--
Why do I target chan_sip for so much effort? Because,
it seems to me, chan_sip is probably the most used channel
driver in the asterisk community!! (and, of course,
the zap/dahdi driver, is also pretty popular)
I haven't had time to follow up on chan_sip,
Try this:
[local]
exten = _00165011091[45][0-9],1,NoOp(I AM HERE)
exten = _00165011091[45][0-9],n,Macro(setcli)
exten = _00165011091[45][0-9],n,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _00165011091[45][0-9],n,Hangup
The [ before 0-9] is needed.
Felippe Silvestre
Hello!
Does anyone tried BRI with asterisk for DATA transfer? My customer
wants BRI connection, but he wants it for the data, and I have to
bring connection to his office, so I see the connection as follows:
E1-(CORE_ASTERISK)-(IAX2)-(EDGE_ASTERISK)-BRI - so will data
work in such scenario? If
On Thu, 2008-08-07 at 18:35 +0200, Pavel Jezek wrote:
Steve Murphy wrote:
Hello--
Why do I target chan_sip for so much effort? Because,
it seems to me, chan_sip is probably the most used channel
driver in the asterisk community!! (and, of course,
the zap/dahdi driver, is also
I just received an email notice from FWD about $30 membership fee.
My question: Is the email genuine? Did anybody else receive it?
I'm just trying to be sure that it is real and not a scam.
The (FWD) does not do anything to authenticate such emails (implementing
GPG/PGP signature etc.)
If the
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED],
altrimenti vi risponderò al mio rientro.
Dimitri Osler
I
Oh for
Stared at that for ages not seeing it
Thanks Felippe...
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Felippe
Silvestre
Sent: 07 August 2008 17:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On Thu, Aug 7, 2008 at 2:04 PM, Joseph [EMAIL PROTECTED] wrote:
I just received an email notice from FWD about $30 membership fee.
My question: Is the email genuine? Did anybody else receive it?
I'm just trying to be sure that it is real and not a scam.
The (FWD) does not do anything to
Hi Felippe,
in the past we had some trouble with a specific SNOM Firmware, which did not
handle dtmf tones correctly. As a workarround, we tried to set
relaxdtmf=yes in sip.conf.
As a result we had these beep-tones generated randomly.
Not shure, if this is your problem too...
Friendly
Pavel Jezek wrote:
Steve Murphy wrote:
Hello--
Why do I target chan_sip for so much effort? Because,
it seems to me, chan_sip is probably the most used channel
driver in the asterisk community!! (and, of course,
the zap/dahdi driver, is also pretty popular)
I haven't had time to
Gonzalo Servat wrote:
On Thu, Aug 7, 2008 at 2:04 PM, Joseph [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I just received an email notice from FWD about $30 membership fee.
My question: Is the email genuine? Did anybody else receive it?
I'm just trying to be sure that it
FWD has had paid membership options for years. The paid memberships help to
improve the network and increase it's reach. As far as I've heard (and as
far as the site mentions), paid membership is not a requirement. That would
sort of go against the talk... for free... for good slogan.
AR
--
Without TICC Capital investing in Pulvermedia/VON, maybe he/they need
another revenue source to pay the bills.
Thanks,
Steve T
On Thu, Aug 7, 2008 at 3:05 PM, Alex Robar [EMAIL PROTECTED] wrote:
FWD has had paid membership options for years. The paid memberships help to
improve the network and
Dear list,
I got this scenario.
FAX Machine - FXS (tdm800) -Asterisk - SIP - OPENSER - SIP -
Asterisk - FXO(tdm400) - PSTN - FAX Machine
I' been reading a lot of Faxes and t.38 protocol... and I found that
Asterisk 1.6 has the possibility to do FAX t.38 Gateway funtion...and also
that
El jue, 07-08-2008 a las 13:31 -0600, Arturo Ochoa escribió:
Has anyone have experiencies on this kind of scenario... what
version?.. patches?... or any information regarding this goal will be
VERY helpful...
Hi Arturo,
Please ckeck the following URL (on spanish):
I badly want to roll out Asterisk at my job. Unfortunately, my boss is
dazzled by shiny objects. We had a vendor in today who showed us their
system which, honestly, didn't suck -- but boy, is it going to be
expensive! One major component of the eye candy was an end-user interface
that allowed
Druid has a user portal that might cover what you are looking for.
Yes I think it's something that is under utilized in the current
offerings.
At one stage I was thinking of rolling out a 3rd party user portal that
would use the ami to sit over and above any asterisk platform but I
could never
I would checkout Switchvox :)
http://www.digium.com/en/products/switchvox/
-Brandon
Ken D'Ambrosio wrote:
I badly want to roll out Asterisk at my job. Unfortunately, my boss is
dazzled by shiny objects. We had a vendor in today who showed us their
system which, honestly, didn't suck -- but
Ken,
You might want to check out our free Druid Open source unified
communications project. It is not proprietary and has open source soap
API for third party applications.
http://www.voiceroute.org
We have mobile integration with blackberry iphone that no vendors
open source or otherwise has.
Yann Derichard wrote:
Hi,
I am trying to install a Voicemail on PRI after a redirection on an away
or a busy (a normal call which is redirected to voicemail in fact) but I
can't find the function in Asterisk which allow me using the phone
number of the callee (because I have only the
Hi All;
Did anyone used AGI to do te CRM integration in the Asterisk call center?
If yes, I would like to know the overview to know from where to start?
Regards
Bilal
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
On Thu, Aug 7, 2008 at 5:55 PM, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi All;
Did anyone used AGI to do te CRM integration in the Asterisk call center?
If yes, I would like to know the overview to know from where to start?
Regards
Bilal
What CRM? FastAGI to hit a box that has logic to
[Sorry if this is a duplicate; originally sent from an address the list
doesn't know.]
Wow. Okay, Druid has my attention; I'll definitely be kicking the tires.
That being said, though, I do have a quick question (that I always have
about GUIs):
First, I assume that Druid is based on Asterisk;
Ken,
Druid is based on Asterisk and we love asterisk for the call control
functionality. We built value on top of asterisk by extending
functionalities to the unified communications space (e-fax, mobility,
sugarcrm integration, google apps integration, auto-provisioning of 5
brand phones)
Check
Thanks Memo,
I've already see that article before, the problem is that this solution is
useful when you want asterisk (via t38modem) to terminate the call...
Someone send you a Fax using t.28 and this software
(t38modem+asterisk+hylafax) will handle the incomming fax. In fact I have a
working
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED],
altrimenti vi risponderò al mio rientro.
Dimitri Osler
I
Does anyone know where I might purchase a G.722 capable SIP soft phone?
Counterpath say that they offer one, but only in the OEM versions do
they support G.722. I need only a couple of licenses.
www.qutecom.org
---
Marek Cervenka
CRM: Customer Record Module which is any kind of application.
For example, a bank has an application and the agent sit on his PC, when call
come, the application fetched with the customer information based on the card
number which is entered with the IVR,
How the application of the bank
I just received an email notice from FWD about $30 membership fee.
My question: Is the email genuine? Did anybody else receive it?
I'm just trying to be sure that it is real and not a scam.
The (FWD) does not do anything to authenticate such emails (implementing
GPG/PGP signature etc.)
We have done this several times for customers with VICIDIAL. I have
also seen companies use AGI scripts to enable this kind of application
as well. So, yes it is possible.
MATT---
On 8/7/08, bilal ghayyad [EMAIL PROTECTED] wrote:
CRM: Customer Record Module which is any kind of application.
On Wednesday 06 August 2008 04:09:13 Pavel Jezek wrote:
A week ago, I tried give realtime priority to asterisk proces using -p
switch,
asterisk was running inside astcanary,
but yestarday asterisk probably starts eating all cpu and lock any
access to computer, only ping was possible,
so,
Hello!
Does anyone tried BRI with asterisk for DATA transfer? My
customer
wants BRI connection, but he wants it for the data, and I
have to
bring connection to his office, so I see the connection as
follows:
E1-(CORE_ASTERISK)-(IAX2)-(EDGE_ASTERISK)-BRI - so
will data
work in such scenario?
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