As for "doing something better", I would hope to see two
things happening ...
1) you begin to use queue_app for your call centre
requirements and if you need any assistance, you ask about
it here
2) having experienced Asterisk's superior queue management
system,
Olle E. Johansson wrote:
Agreed. I wasn't clear enough. Asterisk have users in
many
places, but no centralized view of a "user".
Agreed.
I haven't said that parallel forking is my recommended
way of doing this.
Fair enough.
I've stated several times that
On Wed, 2004-07-14 at 07:55, Sunrise Ltd wrote:
Thats where we should go. [peer]s and [user]s being
devices
(IMEI) and a new user architecture representing the IMSI.
We have accountcode now. It's not enough.
It may well be worth while implementing (parts of) the GSM
IMSI specification into
Andrew Kohlsmith wrote:
I wasn't talking about bandwidth but rather lengthy
Dial() commands...
exten = s,1,Dial(SIP/someuserSIP/someuserSIP ..
kind of thing... seems awfully unwieldy.
That's why you would stick the members into a global
variable
Kannaiyan Natesan wrote:
I hope you clearly understand that everyone here
**KNOWS**
to use alternative software such as SER, what is needed
here is
the same facility in asterisk.
You have not shown us ANY example yet for which this
facility is *NEEDED*.
You have
Duane wrote:
We're running SER and Asterisk on the same system with
Like2Fone.com and we just stuck Asterisk on a different
port then redirect calls as needed, although I doubt it
would
be as difficult as your making out, if you stick SER on
an
alternative port and then
Kannaiyan Natesan wrote:
I hope you clearly understand that everyone here
**KNOWS**
to use alternative software such as SER, what is needed
here is
the same facility in asterisk.
You have not shown us ANY example yet for which this
facility is *NEEDED*.
Based upon the analysis I think we need to modify two things,
1. chan_sip.c (Registrar)
2. app_dial.c (Dial Command for simultaneous dialling, as of now it
supports simultaneous dialling too)
The registrar of SIP need to collect the array of registrants and the Dial
command
You have not shown us ANY example yet for which this
facility is *NEEDED*.
Well, I have users that get an account on my PBX.
I really don't care how many phones they want to use, hardware phones on
their desktop or soft phones on their laptop while travelling. It's still a user
with one account.
Also, you can use the callgroup feature in sip.conf
[111]
...
callgroup=1
callerid=Member 112345
[112]
...
callgroup=1
callerid=Member 212345
[113]
...
callgroup=1
callerid=Member 312345
then in your dialplan
exten = 12345,1,Dial(SIP/111) ; dialling one member
rings them all
: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
You have not shown us ANY example yet for which this
facility is *NEEDED*.
Well, I have users that get an account on my PBX.
I really don't care how many phones they want to use, hardware phones on
their desktop or soft phones
Hello,
From: Sunrise Ltd [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Date: Tue, 13 Jul 2004 16:31:58 +0900 (JST)
snip
If Asterisk is directed to speak SIP on port 5061 and SER
remains on port 5060, then how do you get Asterisk to talk
to SER and vice versa
I can see the point of the discussion somewhere, but let's take it the
other way around (comments though mail):
On Tue, 2004-07-13 at 08:53, Olle E. Johansson wrote:
You have not shown us ANY example yet for which this
facility is *NEEDED*.
Well, I have users that get an account on my
On 13/07/2004 at 11:48 Martin List-Petersen wrote:
I can see the point of the discussion somewhere, but let's take it the
other way around (comments though mail):
On Tue, 2004-07-13 at 08:53, Olle E. Johansson wrote:
You have not shown us ANY example yet for which this
facility is *NEEDED*.
On Tuesday 13 July 2004 03:07, Sunrise Ltd wrote:
exten = s,1,Dial(SIP/someuserSIP/someuserSIP ..
That's why you would stick the members into a global
variable
You global variable is still unwieldy. All you did was move the problem.
Also, you can use the
On Tue, 2004-07-13 at 03:54, Holger Schurig wrote:
Also, you can use the callgroup feature in sip.conf
[111]
...
callgroup=1
callerid=Member 112345
[112]
...
callgroup=1
callerid=Member 212345
[113]
...
callgroup=1
callerid=Member 312345
then in your dialplan
] New Asterisk bounty: SIP simultaneous
If Asterisk is directed to speak SIP on port 5061 and SER
remains on port 5060, then how do you get Asterisk to talk to
SER and vice versa?
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http
Kannaiyan Natesan wrote:
Have you used 5 welcome service in fwd?
If not try that. You are invited to join as a volunteer
to provide support and talk to new people on fwd.
Asterisk can do that much better than SER because it has
got a queue management system built-in.
Girish Gopinath wrote:
[globals]
SERADDRESS=XXX.XXX.XXX.XXX:5060
[context]
exten =
yourexten,1,Dial(SIP/[EMAIL PROTECTED],20,r)
In ser.cfg:
if (method == "INVITE") {
if (uri =~ "sip:[EMAIL PROTECTED]"){
log(1, "Forwarding to Asterisk?n");
Olle E. Johansson wrote:
Well, I have users that get an account on my PBX.
I really don't care how many phones they want to use,
hardware phones on their desktop or soft phones on their
laptop while travelling. It's still a user with one
account.
Two words: self
ubject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Kannaiyan Natesan wrote:
Have you used 5 welcome service in fwd?
If not try that. You are invited to join as a volunteer
to provide support and talk to new people on fwd.
Asterisk can do that m
As I explained to you before we use it for our customer service in call
center and implemented in many call centres which really makes $.
All this stuff to do a simple call queue system??? Man, You need to read
wiki. Read agents.conf and queue.conf before to begin a war here...
Hi!
That sound unwieldy as the number of simultaneous ringers increases...
Can SIP peers be grouped like Zap channels?
Yes - use a queue.
http://www.voip-info.org/wiki-Asterisk+call+queues
Cheers, Philipp
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Asterisk-Users mailing list
[EMAIL
On Jul 13, 2004, at 5:13 AM, Andrew Kohlsmith wrote:
On Tuesday 13 July 2004 03:07, Sunrise Ltd wrote:
exten = s,1,Dial(SIP/someuserSIP/someuserSIP ..
That's why you would stick the members into a global
variable
You global variable is still unwieldy. All you did was move the
problem.
It
Sunrise Ltd wrote:
Olle E. Johansson wrote:
Well, I have users that get an account on my PBX.
I really don't care how many phones they want to use,
hardware phones on their desktop or soft phones on their
laptop while travelling. It's still a user with one
account.
Two words: self provisioning.
I wasn't talking about bandwidth but rather lengthy Dial() commands...
Variables can shorten the parameter greatly
GROUP1=SIP/1SIP/2SIP/3SIP/4
GROUP2=SIP/5SIP/6
dial(${GROUP1}${GROUP2})
I have no idea on the theoretical limit for a dial string, but I suspect it
should be quite long
Youness
On Tuesday 13 July 2004 17:16, Youness El Andaloussi wrote:
Variables can shorten the parameter greatly
GROUP1=SIP/1SIP/2SIP/3SIP/4
GROUP2=SIP/5SIP/6
dial(${GROUP1}${GROUP2})
You're missing the point. This is STILL unwieldy. I'd have to put together a
list of variables, ensuring each
Jay Milk wrote:
[general]
Port=5060
register = [EMAIL PROTECTED]:5061
[sip-vonage]
...
host=sphone.vopr.vonage.net
port=5061
very interesting, thanks.
Still need to get my head around this and see how it may
be used for running X-Lite along
Kannaiyan Natesan wrote:
Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP
simultaneous
I will close it from my side.
Thanks for your info.
Let us do something better.
Did you check out /etc/asterisk/queues.conf and
/etc/asterisk/agents.conf ???
So isn't this the problem * has? The first client registers as the address
of record, then the second client comes in with the same registration and
becomes the address of record?
I think you are making this look more complicated than it actually is.
We do this with our SER Network all the
be careful.
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andres
Sent: 12 July 2004 08:54 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
So isn't this the problem * has? The first client
Paul Mahler wrote:
Well, this is certainly getting exciting.
Yes, it is. Sorry for coming in late to this debate...
Andy, I took your advice and re-read the RFP.
It's actually RFC, not RFP. (teasing :-)
So, gentlemen, help me out here. The spec says:
The Address of record is the SIP address
* No, there's no quick fix for a 100 USD bounty
How much you estimate on quick fix?
-Kannaiyan.
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To UNSUBSCRIBE or update options visit:
Excellent Post! Very Informative. Thanks a lot Sir!
Regards, Girish
From: Olle E. Johansson [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Date: Mon, 12 Jul 2004 10:52:33 +0200
Paul Mahler wrote:
Well, this is certainly getting exciting.
Yes, it is. Sorry
Kannaiyan Natesan wrote:
* No, there's no quick fix for a 100 USD bounty
How much you estimate on quick fix?
I apologize for my Swenglish language...
I don't believe there's a quick fix at all.
If you want a quote for a fix, contact me off-list. But remember, that I believe
that fixing this is
On 11/07/2004 at 18:11 Paul Mahler wrote:
Well, this is certainly getting exciting.
Andy, I took your advice and re-read the RFP. Andy--I don't think you are a
Sorry, I was sleeping when these new emails came in
I've read the other responses which seem to make it pretty clear.. and
I don't think we should let these misunderstandings judge the quality of
Paul's Asterisk book. Even authors need to learn now and then :-)
Can I just point out that the reason I said what I said (see, I can't write)
was because Paul steadfastly refused to believe what we were saying, rather
This may sound like a stupid work around, but how about registering
different extensions and putting both of them in the Dial String (so they
would ring at once) and giving both extensions the same caller id?
I do something with my zaptel and x lite phones... I assign them both the
same number
in response to Olle's excellent post, ...
in particular ...
Asterisk is *not* a SIP proxy. It's a SIP registrar and
location server.
It's a very clever SIP UA. It wants to be in the middle
of the call
and wants to be in control of each device. This
device-slave view
registry possible workaround (was Re:
[Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
This may sound like a stupid work around, but how about registering
different extensions and putting both of them in the Dial String (so they
would ring at once) and giving both extensions
Jason Penton wrote:
Well Andres is right but there are numerous problems with quite a few SIP
clients that do NOT follow the the SIP RFC correctly. There is a problem
with dialog creation in a number of SIP products out there. SIP dialog
creation is the critical part of the spec that supports
: Monday, July 12, 2004 11:20 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
For those rare occasions where one would really need
multiple concurrent SIP registrations I'd say one should
consider running Asterisk in combination with a SIP proxy.
Since
On 02:44 AM 7/12/2004, Olle E. Johansson wrote:
I don't believe there's a quick fix at all.
If you want a quote for a fix, contact me off-list. But remember, that I
believe
that fixing this is chan_sip *will* cause confusion and errors to happen in
other
parts of Asterisk.
There is a sort of
On Mon, 2004-07-12 at 10:40, Youness El Andaloussi wrote:
This may sound like a stupid work around, but how about registering
different extensions and putting both of them in the Dial String (so they
would ring at once) and giving both extensions the same caller id?
That's exactly how I and
Sunrise Ltd wrote:
But if anybody has a problem that truly warrants parallel
forking, then I propose you look into sponsoring somebody
to work on the little port swapping trick to run SER
concurrently on your Asterisk box.
We're running SER and Asterisk on the same system with Like2Fone.com
.
- Original Message -
From: "Sunrise Ltd" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 12, 2004 5:20 PM
Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
in response to Olle's excellent post, ...
in particul
On Monday 12 July 2004 15:18, Eric Wieling wrote:
On Mon, 2004-07-12 at 10:40, Youness El Andaloussi wrote:
This may sound like a stupid work around, but how about registering
different extensions and putting both of them in the Dial String (so they
would ring at once) and giving both
Dude relax, take a deep breath and be zen :). From the tone of your email,
looks like you expect me to apologize for trying to be helpful :)
There is more than one way to skin a cat. You may not like my solution, but
it would work and saying I am wasting your time is a bit over reacting in
my
Not sure how many phones you need ringing at the same time, but it should
not be a major problem as only an invite is sent... and it should not
require much bandwidth just to invite.
At 23:57 12/07/2004, you wrote:
That sound unwieldy as the number of simultaneous ringers increases... Can
SIP
PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 12, 2004 5:20 PM
Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
in response to Olle's excellent post, ...
in particular ...
Asterisk is *not* a SIP proxy. It's a SIP registrar and
location server.
It's a very clever SIP
On Monday 12 July 2004 22:41, Youness El Andaloussi wrote:
That sound unwieldy as the number of simultaneous ringers increases...
Can SIP peers be grouped like Zap channels?
Not sure how many phones you need ringing at the same time, but it should
not be a major problem as only an invite is
.
- Original Message -
From: Youness El Andaloussi [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 13, 2004 3:34 AM
Subject: Re: SIP simultaneous registry possible workaround (was Re:
[Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
Dude relax, take a deep breath and be zen
When I call a SIP user, the phone should ring in more
than one
extentions. Also more than one phone should be able to
register with
asterisk. Right now it is not the case.
There is no issue here. You seem to be confused, that's
all.
A SIP account is a SIP account and
nrise Ltd" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 11, 2004 9:15 AM
Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
When I call a SIP user, the phone should ring in more
than one
extentions. Also more than one phone should be a
: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
I accept your views.
I have a specific requirements, can you help to attain the same.
In our business we have 25 employees handling customer service.
I want to add or remove employees in the customer service so does
canon" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 11, 2004 10:02 AM
Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
I was going to keep out of this (was interesting to read, as I have dealt
with simmillar situation) however I would like t
Daniel Jimenez wrote:
http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry
From the WIKI:
Contributions
Manager: Daniel Jimenez (cuban)
Bounty: $50 USD
Date opened: July 10, 2004
Contributors: cuban ($50)
Detail
Yes, Yes I know you could do all
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Kannaiyan Natesan
Sent: Sunday, July 11, 2004 1:15 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Paul Mahler wrote:
If you want to be able to more easily recognize what extension the traffic
if for, you can add additional extensions to the 7960. For example, if you
have two staff the admin monitors, add two additional extensions to the
7960. The admin can tell who is being called by the
Soren Rathje wrote:
Eh... Sort of like shadow lines ???
Remember that Dial(SIP/1 H323/1 ZAP/1,[timeout],[options],[URL]) will dial all 3
extensions simultaneously (regardless of channel choice) and with a little tinkering in your dialplan you can even
activate/deactivate this from the Manager
Kannaiyan Natesan wrote:
I hope this helps.
Since I feel this is a great feature, I will topup up to $100/-
-.Kannaiyan
http://www.goods2world.com -- Your Only VoIP
Thank you, I updated the wiki with your $25 addition.
--
Daniel Jimenez djimenez[at]pobox[dot]com
Paul Mahler wrote:
If I have the requirement right, you could accomplish this by ringing the
staff extension and the admin extension at the same time. The Dial command
allows you to ring multiple extensions simultaneously.
Paul
Did you even read the bounty?
Yes, Yes I know you could do all sorts
On 11/07/2004 at 08:42 Paul Mahler wrote:
You are confused about what a SIP session is and what a SIP session does.
SIP, session initiation protocol, controls an RTP, real time protocol,
session between two IP endpionts. The end points have to have unique IP
addresses for the session to run.
-
From: usedcanon [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 11, 2004 10:02 AM
Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
I was going to keep out of this (was interesting to read, as I have
dealt with simmillar situation) however I would like
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andy Powell
Sent: Sunday, July 11, 2004 9:57 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
On 11/07/2004
] On Behalf Of
Kannaiyan Natesan
Sent: Sunday, July 11, 2004 9:58 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
As Daniel Says, Bounty stands.
I cannot explain to you anymore. I'm sorry.
Please read more what SIP can do with SER
Daniel Jimenez wrote:
Soren Rathje wrote:
Eh... Sort of like shadow lines ???
Remember that Dial(SIP/1 H323/1
ZAP/1,[timeout],[options],[URL]) will dial all 3 extensions
simultaneously (regardless of channel choice) and with a little
tinkering in your dialplan you can even
Did you even read the RFC? Section 10.2.1 clearly talks about adding
multiple bindings to the same address-of record.
On Sun, 2004-07-11 at 12:31, Paul Mahler wrote:
The whole point of a SIP registration is to identify a UNIQUE device. You
CAN'T HAVE multiple devices registered as the same
On 11/07/2004 at 12:31 Paul Mahler wrote:
The whole point of a SIP registration is to identify a UNIQUE device. You
CAN'T HAVE multiple devices registered as the same SIP device. That's WHY
the last device that registers gets the traffic.
WRONG!
This doesn't have ANYTHING TO DO WITH ASTERISK.
Mike Machado wrote:
On Sun, 2004-07-11 at 12:31, Paul Mahler wrote:
The whole point of a SIP registration is to identify a UNIQUE device. You
CAN'T HAVE multiple devices registered as the same SIP device. That's WHY
the last device that registers gets the traffic.
This doesn't have ANYTHING
Paul Mahler wrote:
You should spend your money on getting a copy of each of the
two books that are now available and learn *. Then it will be clear to you
that you don't really want what you are asking for.
Shameless plug? It's offical you are trolling.
--
Daniel Jimenez
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Nicholas Bachmann
Sent: Sunday, July 11, 2004 12:41 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Mike Machado wrote:
On Sun, 2004-07-11
http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry
From the WIKI:
Contributions
Manager: Daniel Jimenez (cuban)
Bounty: $50 USD
Date opened: July 10, 2004
Contributors: cuban ($50)
Detail
Yes, Yes I know you could do all sorts of fun with the dialplan to
produce a
http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry
Updated,
Allow a SIP device to register more than once so a single extension may
exist in multiple locations.
Upped total to $75.
Daniel...
Daniel Jimenez wrote:
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Daniel Jimenez
Sent: Saturday, July 10, 2004 3:05 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP
simultaneous registry
http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+s
: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry
I'm not sure I understand what you are trying to do.
You have an administrative assistant and several other staff. You want the
administrator to be able to take calls directed to the staff extensions?
If I have the requirement
On Sun, 11 Jul 2004, Kannaiyan Natesan wrote:
When I call a SIP user, the phone should ring in more than one
extentions. Also more than one phone should be able to register with
asterisk. Right now it is not the case. The last phone which register will
be receiving the calls. This type of
up to $100/-
-.Kannaiyan
http://www.goods2world.com -- Your Only VoIP
- Original Message -
From: Paul Mahler [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 11, 2004 5:44 AM
Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
registry
I'm not sure I
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