PLEASE CAN SOMBODY HELP ME PROGRAMIN AN VoIP ASTERISK SERVER
Atentamente,
Franz Schuverer Arrue
GLOBAL GROUP, INC.
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Enviado el: S�bado, 23 de Abril de 2005 11:00 a.m.
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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" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Cisco 7960 won't register as SIP device
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[email protected]>
Message-ID:
<[EMAIL PROTECTED]>
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:[EMAIL PROTECTED];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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 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?
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------------------------------
Message: 2
Date: Sat, 23 Apr 2005 08:25:29 -0700 (PDT)
From: Thomas Miller <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] if outgoing call fails with provider 1
then auto try provider 2
To: Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
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.
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------------------------------
Message: 3
Date: Sat, 23 Apr 2005 08:26:25 -0700 (PDT)
From: Thomas Miller <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] if outgoing call fails with provider 1
then auto try provider 2
To: Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=us-ascii
Thanks Andrew for the great example! Anybody else have
any input?
Tom
--- Andrew Kohlsmith <[EMAIL PROTECTED]>
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" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Cisco 7960 won't register as SIP device
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[email protected]>, "List Receiver"
<[EMAIL PROTECTED]>
Message-ID: <[EMAIL PROTECTED]>
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:[EMAIL PROTECTED];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" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Re: Hotel billing in IPSwitchBoard
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[email protected]>
Message-ID:
<[EMAIL PROTECTED]>
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: [EMAIL PROTECTED] 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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of tgj
Sent: Saturday, April 23, 2005 7:55 AM
To: [email protected]
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
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------------------------------
Message: 6
Date: Sat, 23 Apr 2005 16:48:25 +0100
From: "Steve Rawlings" <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Hotel billing in IPSwitchBoard
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
reply-type=original
----- Original Message -----
From: "Thorben Jensen" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<[email protected]>
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" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Cisco 7960 won't register as SIP device
To: "[EMAIL PROTECTED]" <[EMAIL PROTECTED]>, "Asterisk Users Mailing
List - Non-Commercial Discussion"
<[email protected]>,
"List Receiver" <[EMAIL PROTECTED]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 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:[EMAIL PROTECTED];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" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Cisco 7960 won't register as SIP device
To: <[EMAIL PROTECTED]>, "Asterisk Users Mailing List -
Non-Commercial Discussion"
<[email protected]>
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"
Aye...that was it...
Thanks a billion!
-----Original Message-----
From: Robert Webb [mailto:[EMAIL PROTECTED] On Behalf Of Robert Webb
Sent: Saturday, April 23, 2005 8:54 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion; List Receiver
Subject: RE: [Asterisk-Users] Cisco 7960 won't register as SIP device
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 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:[EMAIL PROTECTED];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 <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Quadbri & bristuff: can * respond only
to 1 MSN and leave 1 number to other ISDN phones ?
To: [email protected]
Message-ID: <[EMAIL PROTECTED]>
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
[EMAIL PROTECTED]
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" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Re: Hotel billing in IPSwitchBoard
To: [email protected]
Message-ID: <[EMAIL PROTECTED]>
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)" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Hotel billing in IPSwitchBoard
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"
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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 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" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<[email protected]>
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.
_______________________________________________
Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
------------------------------
Message: 12
Date: Sat, 23 Apr 2005 12:31:35 -0400
From: Michael DiMartino <[EMAIL PROTECTED]>
Subject: [Fwd: FW: [Asterisk-Users] IAX help]
To: Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
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:[EMAIL PROTECTED]/${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:[EMAIL PROTECTED]
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 <[EMAIL PROTECTED]> wrote:
3. Extensions.conf (telx-NY17S)
;Extentions at telx-nyc
exten => _7XXX,1,Dial(IAX2/telx-nyc/${EXTEN})
exten => _7XXX,1,Dial(IAX2/username:[EMAIL PROTECTED]/${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: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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Message: 13
Date: Sat, 23 Apr 2005 18:26:28 +0200
From: "tgj" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Re: Re: Hotel billing in IPSwitchBoard
To: [email protected]
Message-ID: <[EMAIL PROTECTED]>
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 <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] OctoBRI and 2.6kernel
To: Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
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 <[EMAIL PROTECTED]> 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
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------------------------------
Message: 15
Date: Sat, 23 Apr 2005 17:39:01 +0100
From: Peter Bowyer <[EMAIL PROTECTED]>
Subject: Re: [Fwd: FW: [Asterisk-Users] IAX help]
To: Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1
On 23/04/05, Michael DiMartino <[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]/${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: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
------------------------------
Message: 16
Date: Sat, 23 Apr 2005 17:48:54 +0100
From: David John Walsh <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Re: Hotel billing in IPSwitchBoard
To: Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
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 <[EMAIL PROTECTED]> 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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End of Asterisk-Users Digest, Vol 9, Issue 209
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