[Asterisk-Users] ASTERISK PROGRAMER
Franz
global_group_inc at yahoo.es
Sat Apr 23 10:50:56 MST 2005
PLEASE CAN SOMBODY HELP ME PROGRAMIN AN VoIP ASTERISK SERVER
Atentamente,
Franz Schuverer Arrue
GLOBAL GROUP, INC.
www.telefoniaglobal.net
gerencia at telefoniaglobal.net
Tel. (504) 221-4062 (Honduras
Tel. (507) 322-2259 (Panamá)
Tel. (866) 978-0976 (U.S.A.)
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-----Mensaje original-----
De: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] En nombre de
asterisk-users-request at lists.digium.com
Enviado el: Sábado, 23 de Abril de 2005 11:00 a.m.
Para: asterisk-users at lists.digium.com
Asunto: Asterisk-Users Digest, Vol 9, Issue 209
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Today's Topics:
1. RE: Cisco 7960 won't register as SIP device (List Receiver)
2. Re: if outgoing call fails with provider 1 then auto try
provider 2 (Thomas Miller)
3. Re: if outgoing call fails with provider 1 then auto try
provider 2 (Thomas Miller)
4. RE: Cisco 7960 won't register as SIP device (Robert Webb)
5. RE: Re: Hotel billing in IPSwitchBoard (Mathew McKernan)
6. Re: Hotel billing in IPSwitchBoard (Steve Rawlings)
7. RE: Cisco 7960 won't register as SIP device (Robert Webb)
8. RE: Cisco 7960 won't register as SIP device (List Receiver)
9. Re: Quadbri & bristuff: can * respond only to 1 MSN and
leave
1 number to other ISDN phones ? (Michiel van Baak)
10. Re: Hotel billing in IPSwitchBoard (tgj)
11. RE: Hotel billing in IPSwitchBoard (Chris Mason (Lists))
12. [Fwd: FW: [Asterisk-Users] IAX help] (Michael DiMartino)
13. Re: Re: Hotel billing in IPSwitchBoard (tgj)
14. Re: OctoBRI and 2.6kernel (Michael Bielicki)
15. Re: [Fwd: FW: [Asterisk-Users] IAX help] (Peter Bowyer)
16. Re: Re: Hotel billing in IPSwitchBoard (David John Walsh)
----------------------------------------------------------------------
Message: 1
Date: Sat, 23 Apr 2005 08:23:32 -0700
From: "List Receiver" <listreceiver at mastermindpro.com>
Subject: RE: [Asterisk-Users] Cisco 7960 won't register as SIP device
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<DC7C0457603D8D4989F0560F617DBFA24051A8 at exch1.redwest.mastermindpro.com>
Content-Type: text/plain; charset="us-ascii"
The DNS servers are valid. I configured the phone via .cnf files. The
following are the sip.conf and sipMAC.cnf files.
[tycisco]
type=friend
username=username
secret=secret
qualify=200 ; Qualify peer is no more than 200ms
away
nat=yes
;insecure=no
host=dynamic ; This device registers with us
;defaultip=24.18.147.95
canreinvite=no
context=fullaccess
dtmfmode=inband
;mailbox=101
disallow=all
allow=ulaw
allow=alaw
allow=g729
.cnf:
# SIP Configuration File (start)
# Proxy Server
proxy1_address: "asterisk.mastermindpro.com"
proxy2_address: ""
proxy3_address: ""
proxy4_address: ""
proxy5_address: ""
proxy6_address: ""
# Line 1 Settings
line1_name: "tycisco" ; Line 1 Extension\User ID
line1_displayname: "101" ; Line 1 Display Name
line1_authname: "username" ; Line 1 Registration Authentication
line1_password: "secret" ; Line 1 Registration Password
# Line 2 Settings
line2_name: "" ; Line 2 Extension\User ID
line2_displayname: "" ; Line 2 Display Name
line2_authname: "UNPROVISIONED" ; Line 2 Registration
Authentication
line2_password: "UNPROVISIONED" ; Line 2 Registration Password
# Line 3 Settings
line3_name: "" ; Line 3 Extension\User ID
line3_displayname: "" ; Line 3 Display Name
line3_authname: "UNPROVISIONED" ; Line 3 Registration
Authentication
line3_password: "UNPROVISIONED" ; Line 3 Registration Password
# Line 4 Settings
line4_name: "" ; Line 4 Extension\User ID
line4_displayname: "" ; Line 4 Display Name
line4_authname: "UNPROVISIONED" ; Line 4 Registration
Authentication
line4_password: "UNPROVISIONED" ; Line 4 Registration Password
# Line 5 Settings
line5_name: "" ; Line 5 Extension\User ID
line5_displayname: "" ; Line 5 Display Name
line5_authname: "UNPROVISIONED" ; Line 5 Registration
Authentication
line5_password: "UNPROVISIONED" ; Line 5 Registration Password
# Line 6 Settings
line6_name: "" ; Line 6 Extension\User ID
line6_displayname: "" ; Line 6 Display Name
line6_authname: "UNPROVISIONE" ; Line 6 Registration
Authentication
line6_password: "UNPROVISIONE" ; Line 6 Registration Password
# Emergency Proxy info
proxy_emergency: ""
proxy_emergency_port: "5060"
# Backup Proxy info
proxy_backup: ""
proxy_backup_port: "5060"
# Outbound Proxy info
outbound_proxy: ""
outbound_proxy_port: "5060"
# NAT/Firewall Traversal
nat_enable: "1"
nat_address: "24.18.147.95"
voip_control_port: "5060"
start_media_port: "16384"
end_media_port: "32766"
nat_received_processing: "1"
# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "Ty's Phone " ; Has no effect on SIP messaging
# Time Zone phone will reside in
time_zone: PST
# Enable_VAD (1-enabled, 0-disabled)
enable_vad: "0"
# Network Media Type (auto, full100, full10, half100, half10)
network_media_type: "auto"
#user_info: phone
# SIP Configuration File (stop)
When the phone tries to register, all I get in the Asterisk console is
this:
Apr 23 08:22:29 NOTICE[26568]: chan_sip.c:8804 handle_request_register:
Registration from '<sip:tycisco at asterisk.mastermindpro.com;user=phone>'
failed for '24.18.147.95'
...but the phone can make a call to any destination in the dialplan...
:^/
Where's my stupidity? Am I confused on all the "names" in the .cnf
file?
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
> Henry Devito
> Sent: Saturday, April 23, 2005 6:11 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Cisco 7960 won't register as SIP device
>
> It can use DNS if the DNS servers are valid. Can you post
> your SIP.conf?
> Didi you configure the phone manually or did you use the cnf
> files? If you used cnf files can you post those also?
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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------------------------------
Message: 2
Date: Sat, 23 Apr 2005 08:25:29 -0700 (PDT)
From: Thomas Miller <thomasamillergoogle at yahoo.com>
Subject: Re: [Asterisk-Users] if outgoing call fails with provider 1
then auto try provider 2
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <20050423152529.12664.qmail at web53304.mail.yahoo.com>
Content-Type: text/plain; charset=us-ascii
Rich- wouldn't Andrew K's solution work? That seems to
make good sense.
>
> There are no real examples that would address your
> points. The
> primary reason is that your * can dispatch a call to
> a provider
> and the provider will accept that handshaking call.
> But, if
> they are having internal call-completion issues,
> there is no
> way for you to know that. You could get some sort of
> busy,
> dead air, etc.
>
> You could probably design some sort of timer-based
> timeout,
> but what indication would you use to indicate the
> call was
> successful vs unsuccessful?
>
> There are several ways to address whether your * is
> successful
> in reaching your provider's equipment, but that's
> about it.
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>
>
http://lists.digium.com/mailman/listinfo/asterisk-users
>
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------------------------------
Message: 3
Date: Sat, 23 Apr 2005 08:26:25 -0700 (PDT)
From: Thomas Miller <thomasamillergoogle at yahoo.com>
Subject: Re: [Asterisk-Users] if outgoing call fails with provider 1
then auto try provider 2
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <20050423152625.38297.qmail at web53309.mail.yahoo.com>
Content-Type: text/plain; charset=us-ascii
Thanks Andrew for the great example! Anybody else have
any input?
Tom
--- Andrew Kohlsmith <akohlsmith-asterisk at benshaw.com>
wrote:
> On April 22, 2005 10:38 pm, Thomas Miller wrote:
> > When someone teminates a call with my softphone to
> m
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------------------------------
Message: 4
Date: Sat, 23 Apr 2005 11:42:29 -0400
From: "Robert Webb" <asterisk at ropeguru.com>
Subject: RE: [Asterisk-Users] Cisco 7960 won't register as SIP device
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>, "List Receiver"
<listreceiver at mastermindpro.com>
Message-ID: <63cb01deced67c4d86cc1b902bef3ef5 at mail.ropeguru.com>
Content-Type: text/plain; charset="us-ascii"
<SNIP>
> #user_info: phone
>
> # SIP Configuration File (stop)
>
> When the phone tries to register, all I get in the Asterisk
> console is this:
>
> Apr 23 08:22:29 NOTICE[26568]: chan_sip.c:8804
> handle_request_register:
> Registration from
> '<sip:tycisco at asterisk.mastermindpro.com;user=phone>'
> failed for '24.18.147.95'
I am unfamiliar with the Cisco configs but I am just comparing your
error message to what you have in the config to make this suggestion. In
the error it has "user=phone" and in your config commented out there is
"#user_info: phone". What if you tried uncommenting that line and
putting in "username"? It could be that when thatline is commented out,
it uses "phone" by default.
Robert
------------------------------
Message: 5
Date: Sun, 24 Apr 2005 01:50:39 +1000
From: "Mathew McKernan" <mat at dwonline.com.au>
Subject: RE: [Asterisk-Users] Re: Hotel billing in IPSwitchBoard
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<B655C646916F3D459EFA79BE650C07C903AD26 at dwserver.intrl.dwonline.com.au>
Content-Type: text/plain; charset="iso-8859-1"
Hi,
Have a look at http://www.voip-info.org/wiki-CallingCard+Applications
I recently used this in a hospital for the same concept. Can charge on
caller ID etc. Works really well.
Ties to a MySQL database, so a PHP interface can be coded to view the
call charges etc on a room. It works on a card system, but all the SQL
commands are customisable, so it does the job.
Also, the destination charges are managable through the tables and
different charges for different prefixes can be a applied. Also it
supports LCDial (least cost routing dialler). So it will choose the
carrier (if you box will use it) based on the cheapest rate (for the
hotel, still charges the customer the same). In the application I used
it for, it puts International Calls through our IP Provider and local
calls/mobiles through our carrier as it was cheaper.
Hope this might help,
Thanks
Mathew
________________________________
From: asterisk-users-bounces at lists.digium.com on behalf of Chris Mason
(Lists)
Sent: Sat 23/04/2005 23:03
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Re: Hotel billing in IPSwitchBoard
Also needed is a way to title and logo the print out so it looks like an
invoice. A tempplate would work, and if can use HTML templates that
would be
easy to customise. Consider making the data a table that is substituted
into
the html template.
Chris Mason
www.anguillaguide.com
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of tgj
> Sent: Saturday, April 23, 2005 7:55 AM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Re: Hotel billing in IPSwitchBoard
>
> > Exactly what I am looking for also. Because we have
> multiple phones in
> > one villa, I would need the ability to group extensions and
> produce an
> > overall bill, and I would, of course, need the ability to set the
> > charge rate versus the cost, i.e., the cost is $.02/min,
> but we might
> > charge $.50/min regardless of destination, a flat fee for all long
> > distance and international.
> > This is so cool.
>
> Hi Chris
>
> Grouping is a good idea, will not be in the first release, but later.
>
> There will only be a charge rate in the first release. You
> can charge depending on the destination.
>
> Thorben
>
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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------------------------------
Message: 6
Date: Sat, 23 Apr 2005 16:48:25 +0100
From: "Steve Rawlings" <steve at rawlings.demon.co.uk>
Subject: Re: [Asterisk-Users] Hotel billing in IPSwitchBoard
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID: <000601c5481b$e3338b10$0c01a8c0 at SR1>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
reply-type=original
----- Original Message -----
From: "Thorben Jensen" <thorben at thorben.dk>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<asterisk-users at lists.digium.com>
Sent: Saturday, April 23, 2005 8:11 AM
Subject: [Asterisk-Users] Hotel billing in IPSwitchBoard
>I am currently working on implementing Hotel Billing in IPSwitchBoard.
>
> The idea is that a receptionist in a hotel can just right click an
> extension
> button and choose "Account"; IPS will now calculate the call charges
made
> from that extension and show all calls and charges on a form.
>
> The receptionist now has the option to close the account which will
reset
> the account.
>
> I will add a table for editing call charges, and there will be a
> possibility
> to add a fee for connection charges and also an option to charge calls
per
> xx seconds and to add/subtract a percentage to all calls.
>
> I will add a family/key to the asterisk database to indicate if the
> extension is closed, this way you can stop outgoing calls from being
made
> from a closed extension by checking the dial plan.
>
>
> Please let me know if there are any other features you would like to
see
> in
> IPSwitchBoard.
>
Hi,
As mentioned before, how about being able to search and replay
recordings
from the switchboard. With call records now searchable hopefully it
wouldn't take too much more work to enable. For example, being able to
search on extension by date and time or by cli would be very handy.
Best regards,
Steve.
------------------------------
Message: 7
Date: Sat, 23 Apr 2005 11:53:50 -0400
From: "Robert Webb" <asterisk at ropeguru.com>
Subject: RE: [Asterisk-Users] Cisco 7960 won't register as SIP device
To: "rwebb at ropeguru.com" <rwebb at ropeguru.com>, "Asterisk Users Mailing
List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>,
"List Receiver" <listreceiver at mastermindpro.com>
Message-ID: <917e0d16d1901d4992b29c4527d99e15 at mail.ropeguru.com>
Content-Type: text/plain; charset="us-ascii"
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
> Robert Webb
> Sent: Saturday, April 23, 2005 11:42 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion;
> List Receiver
> Subject: RE: [Asterisk-Users] Cisco 7960 won't register as SIP device
>
> <SNIP>
>
> > #user_info: phone
> >
> > # SIP Configuration File (stop)
> >
> > When the phone tries to register, all I get in the Asterisk
> console is
> > this:
> >
> > Apr 23 08:22:29 NOTICE[26568]: chan_sip.c:8804
> > handle_request_register:
> > Registration from
> > '<sip:tycisco at asterisk.mastermindpro.com;user=phone>'
> > failed for '24.18.147.95'
>
>
> I am unfamiliar with the Cisco configs but I am just
> comparing your error message to what you have in the config
> to make this suggestion. In the error it has "user=phone" and
> in your config commented out there is
> "#user_info: phone". What if you tried uncommenting that line
> and putting in "username"? It could be that when thatline is
> commented out, it uses "phone" by default.
>
> Robert
>
Actually after getting into the Cisco site it looks like you want a
value of "none" for that.
Configures the "user=" parameter in the REGISTER message. Valid values
are:
* none-No value is inserted.
* phone-The value user=phone is inserted in the To, From, and
Contact Headers for REGISTER.
* ip-The value user=ip is inserted in the To, From, and Contact
Headers for REGISTER.
The default value is none.
It says the default value is "none" but you may want to hard code it as
it looks like that is not what it is doing.
------------------------------
Message: 8
Date: Sat, 23 Apr 2005 09:09:29 -0700
From: "List Receiver" <listreceiver at mastermindpro.com>
Subject: RE: [Asterisk-Users] Cisco 7960 won't register as SIP device
To: <rwebb at ropeguru.com>, "Asterisk Users Mailing List -
Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<DC7C0457603D8D4989F0560F617DBFA24051AE at exch1.redwest.mastermindpro.com>
Content-Type: text/plain; charset="us-ascii"
Aye...that was it...
Thanks a billion!
> -----Original Message-----
> From: Robert Webb [mailto:rwebb at ropeguru.com] On Behalf Of Robert Webb
> Sent: Saturday, April 23, 2005 8:54 AM
> To: rwebb at ropeguru.com; Asterisk Users Mailing List -
> Non-Commercial Discussion; List Receiver
> Subject: RE: [Asterisk-Users] Cisco 7960 won't register as SIP device
>
>
>
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf
> Of Robert
> > Webb
> > Sent: Saturday, April 23, 2005 11:42 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion; List
> > Receiver
> > Subject: RE: [Asterisk-Users] Cisco 7960 won't register as
> SIP device
> >
> > <SNIP>
> >
> > > #user_info: phone
> > >
> > > # SIP Configuration File (stop)
> > >
> > > When the phone tries to register, all I get in the Asterisk
> > console is
> > > this:
> > >
> > > Apr 23 08:22:29 NOTICE[26568]: chan_sip.c:8804
> > > handle_request_register:
> > > Registration from
> > > '<sip:tycisco at asterisk.mastermindpro.com;user=phone>'
> > > failed for '24.18.147.95'
> >
> >
> > I am unfamiliar with the Cisco configs but I am just comparing your
> > error message to what you have in the config to make this
> suggestion.
> > In the error it has "user=phone" and in your config commented out
> > there is
> > "#user_info: phone". What if you tried uncommenting that line and
> > putting in "username"? It could be that when thatline is commented
> > out, it uses "phone" by default.
> >
> > Robert
> >
>
>
> Actually after getting into the Cisco site it looks like you
> want a value of "none" for that.
>
> Configures the "user=" parameter in the REGISTER message.
> Valid values
> are:
>
> * none-No value is inserted.
> * phone-The value user=phone is inserted in the To, From,
> and Contact Headers for REGISTER.
> * ip-The value user=ip is inserted in the To, From, and
> Contact Headers for REGISTER.
>
> The default value is none.
>
>
> It says the default value is "none" but you may want to hard
> code it as it looks like that is not what it is doing.
>
>
>
>
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------------------------------
Message: 9
Date: Sat, 23 Apr 2005 18:17:59 +0200
From: Michiel van Baak <michiel at vanbaak.info>
Subject: Re: [Asterisk-Users] Quadbri & bristuff: can * respond only
to 1 MSN and leave 1 number to other ISDN phones ?
To: asterisk-users at lists.digium.com
Message-ID: <20050423161758.GB20321 at vanbaak.info>
Content-Type: text/plain; charset=us-ascii
> >
> >Works for me too.
> >We have an old fax machine sitting on the same NT1 as
> >asterisk. In asterisk I ignored the MNS by setting the line
> >exten => my_fax_msn,1,wait(30)
> >
> >
> Doesn't it work without the wait() in .nl? I just didn't mention the
fax
> MSNs in my incoming context...
>
I tried, but my default context only has a line:
exten => s,1,Congestion
I did that to prevent usage from outside, since my asterisk
box is open for outside sip phones. My folks connect to it
etc. So without the wait, the incoming call will search for
an exten=> line in the incoming context, won't find one so
falls back to default,s,1
That way faxes wont arrive on my fax machine cause asterisk
is playing the congestion tone.
--
Michiel van Baak
http://lunteren.vanbaak.info
michiel at vanbaak.info
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D
"Two of the most famous products of Berkeley are LSD and BSD. I don't
think that this is a coincidence."
------------------------------
Message: 10
Date: Sat, 23 Apr 2005 18:25:24 +0200
From: "tgj" <thorben at thorben.dk>
Subject: [Asterisk-Users] Re: Hotel billing in IPSwitchBoard
To: asterisk-users at lists.digium.com
Message-ID: <d4dski$ife$1 at sea.gmane.org>
> Hi,
>
> As mentioned before, how about being able to search and replay
recordings
> from the switchboard. With call records now searchable hopefully it
> wouldn't take too much more work to enable. For example, being able
to
> search on extension by date and time or by cli would be very handy.
>
> Best regards,
> Steve.
>
Hi Steve,
I will implement that too, but in a later release.
thorben
------------------------------
Message: 11
Date: Sat, 23 Apr 2005 12:26:35 -0400
From: "Chris Mason (Lists)" <lists at masonc.com>
Subject: RE: [Asterisk-Users] Hotel billing in IPSwitchBoard
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<asterisk-users at lists.digium.com>
Message-ID: <20050423163315.AC03092C3AB at mercury.mason.home>
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Now that makes me very excited. I have implemented a pbx in a datacenter
for
a online stock exchange and they want all calls recorded. I am uncertain
how
to handle recovery of the calls, though. This would be wonderful.
Chris Mason
www.anguillaguide.com
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
> Steve Rawlings
> Sent: Saturday, April 23, 2005 11:48 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Hotel billing in IPSwitchBoard
>
> ----- Original Message -----
> From: "Thorben Jensen" <thorben at thorben.dk>
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Sent: Saturday, April 23, 2005 8:11 AM
> Subject: [Asterisk-Users] Hotel billing in IPSwitchBoard
>
>
> >I am currently working on implementing Hotel Billing in
> IPSwitchBoard.
> >
> > The idea is that a receptionist in a hotel can just right click an
> > extension
> > button and choose "Account"; IPS will now calculate the
> call charges made
> > from that extension and show all calls and charges on a form.
> >
> > The receptionist now has the option to close the account
> which will reset
> > the account.
> >
> > I will add a table for editing call charges, and there will be a
> > possibility
> > to add a fee for connection charges and also an option to
> charge calls per
> > xx seconds and to add/subtract a percentage to all calls.
> >
> > I will add a family/key to the asterisk database to indicate if the
> > extension is closed, this way you can stop outgoing calls
> from being made
> > from a closed extension by checking the dial plan.
> >
> >
> > Please let me know if there are any other features you
> would like to see
> > in
> > IPSwitchBoard.
> >
> Hi,
>
> As mentioned before, how about being able to search and
> replay recordings
> from the switchboard. With call records now searchable hopefully it
> wouldn't take too much more work to enable. For example,
> being able to
> search on extension by date and time or by cli would be very handy.
>
> Best regards,
> Steve.
>
> _______________________________________________
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> Asterisk-Users at lists.digium.com
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------------------------------
Message: 12
Date: Sat, 23 Apr 2005 12:31:35 -0400
From: Michael DiMartino <mdm at bigmtnskier.com>
Subject: [Fwd: FW: [Asterisk-Users] IAX help]
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <426A7867.5080709 at bigmtnskier.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Peter thanks for the response.
I put the user name and password in but i still get the same error.
;Extentions at telx-nyc
exten => _70XX,1,Dial(IAX2/telx-nyc:telx-nyc at telx-nyc/${EXTEN})
Apr 23 12:30:35 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected
connect attempt from 192.168.0.251
What else could it be?
-----Original Message-----
From: Peter Bowyer [mailto:peeebeee at gmail.com]
Sent: Saturday, April 23, 2005 4:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAX help
On 23/04/05, Michael DiMartino <mdm at bigmtnskier.com> wrote:
> 3. Extensions.conf (telx-NY17S)
> ;Extentions at telx-nyc
> exten => _7XXX,1,Dial(IAX2/telx-nyc/${EXTEN})
exten => _7XXX,1,Dial(IAX2/username:password at telx-nyx/${EXTEN})
where username:password is the credientials you need to authenticate
with the other server.
The username/secret in iax2.conf is for inbound, not for outbound calls.
Peter
--
Peter Bowyer
Email: peter at bowyer.org
Tel: +44 1296 768003
VoIP: sip:peter at bowyer.org
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------------------------------
Message: 13
Date: Sat, 23 Apr 2005 18:26:28 +0200
From: "tgj" <thorben at thorben.dk>
Subject: [Asterisk-Users] Re: Re: Hotel billing in IPSwitchBoard
To: asterisk-users at lists.digium.com
Message-ID: <d4dsmi$ikd$1 at sea.gmane.org>
> Also needed is a way to title and logo the print out so it looks like
an
> invoice. A tempplate would work, and if can use HTML templates that
would
> be
> easy to customise. Consider making the data a table that is
substituted
> into
> the html template.
> Chris Mason
> www.anguillaguide.com
Hi Chris,
I will find a solution :-)
thank you
thorben
------------------------------
Message: 14
Date: Sat, 23 Apr 2005 18:38:33 +0200
From: Michael Bielicki <cypromis at gmail.com>
Subject: Re: [Asterisk-Users] OctoBRI and 2.6kernel
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <18fec271050423093852edc0d at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
are you using udev ? If yes, check README.udev in the zaptel directory
On 4/23/05, Terry Wade <terry at isdial.net> wrote:
>
>
>
> Hi Guys
>
>
>
> I am trying to get the Junghanns card to load on Suse 9.3 and tried to
get
> it running on Fedora Core 3 (latest kernels). I have heard from a
source
> here in South Africa that this is about as hard as pulling teeth.
Could
> someone please confirm this for me and if they do have it working
properly
> is it possible to get a guide.
>
>
>
> I can get the zaptel and qozap to load the card and all the ports and
inside
> asterisk I see the zap channels. But I cannot get a line out or make
any
> incoming calls.
>
>
>
> Are there some 2.6 tweaks that I need to do in the kernel.
>
>
>
> Kind Regards
>
>
>
> Terry Wade
>
> Mobile: +27 82 802-5750
>
> Office: +27 11 784-7642
>
> Fax: +27 11 388-0855
>
>
>
> Linux is like a Wigwam - No gates, no windows, Apache inside
>
>
>
> Disclaimer and Confidentiality Warning
>
>
>
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>
--
Michal Bielicki
http://www.aefirion.org/
http://www.asterisk.com.pl/
------------------------------
Message: 15
Date: Sat, 23 Apr 2005 17:39:01 +0100
From: Peter Bowyer <peeebeee at gmail.com>
Subject: Re: [Fwd: FW: [Asterisk-Users] IAX help]
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <56152ae90504230939dc42176 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
On 23/04/05, Michael DiMartino <mdm at bigmtnskier.com> wrote:
> Peter thanks for the response.
> I put the user name and password in but i still get the same error.
>
> ;Extentions at telx-nyc
> exten => _70XX,1,Dial(IAX2/telx-nyc:telx-nyc at telx-nyc/${EXTEN})
>
> Apr 23 12:30:35 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected
> connect attempt from 192.168.0.251
>
> What else could it be?
This peer entry in telx-nyc's iax.conf:
; telx-NY17S - Incoming
[telx-NY17S]
type=peer
secret=telx-NY17S
context=from-telx-NY17S
disallow=all
allow=ulaw
Needs to match with the dial string you're calling it with above. See
the difference?
Check the presented username with iax debug enabled to confirm.
Peter
--
Peter Bowyer
Email: peter at bowyer.org
Tel: +44 1296 768003
VoIP: sip:peter at bowyer.org
------------------------------
Message: 16
Date: Sat, 23 Apr 2005 17:48:54 +0100
From: David John Walsh <davidjohnwalsh at gmail.com>
Subject: Re: [Asterisk-Users] Re: Hotel billing in IPSwitchBoard
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <eeb77e8905042309482abd5b9e at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
Taking this idea a little further.
(I apreciate there may be "legal" issues with this request)
Would it be possible for extensions to be tagged, so that if they make
and / or recive a call the call is automatically recorded each and
every time, at the end of the call the file is closed
I would imagine, that its either set in the context menu of the
extention (ie right click, select always record on active) or in the
extensions list.
A supervise (either on demand or always) would be a great help as well.
On 4/23/05, tgj <thorben at thorben.dk> wrote:
> > Hi,
> >
> > As mentioned before, how about being able to search and replay
recordings
> > from the switchboard. With call records now searchable hopefully it
> > wouldn't take too much more work to enable. For example, being able
to
> > search on extension by date and time or by cli would be very handy.
> >
> > Best regards,
> > Steve.
> >
> Hi Steve,
>
> I will implement that too, but in a later release.
>
> thorben
>
>
> _______________________________________________
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> Asterisk-Users at lists.digium.com
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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------------------------------
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