[Asterisk-Users] ASTERISK PROGRAMER
Matt Klein
mklein at nmedia.net
Sat Apr 23 11:13:54 MST 2005
$4,172.38 USD and I'll programin anything you want for asterisk server.
On Sat, 23 Apr 2005, Franz wrote:
> 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
>
> Send Asterisk-Users mailing list submissions to
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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
>> To UNSUBSCRIBE or update options visit:
>> 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
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam? Yahoo! Mail has the best spam protection around
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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 $.50/min regardless of destination, a flat fee for all long
>>> distance and international.
>>> This is so cool.
>>
>> Hi Chris
>>
>> Grouping is a good idea, will not be in the first release, but later.
>>
>> There will only be a charge rate in the first release. You
>> can charge depending on the destination.
>>
>> Thorben
>>
>>
>>
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>
> _______________________________________________
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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>
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> ------------------------------
>
> Message: 6
> Date: Sat, 23 Apr 2005 16:48:25 +0100
> From: "Steve Rawlings" <steve at rawlings.demon.co.uk>
> Subject: Re: [Asterisk-Users] Hotel billing in IPSwitchBoard
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID: <000601c5481b$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>
> 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: 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.
>>
>> _______________________________________________
>> 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
>>
>>
>
>
>
> ------------------------------
>
> 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
> _______________________________________________
> 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
>
>
>
>
>
>
>
> ------------------------------
>
> 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
>>
>>
>>
>> This message is intended for the addressee only. If you are not the
> intended
>> recipient of this message, you are notified that any distribution, use
> of or
>> copying of this communication is strictly prohibited. If you have
> received
>> the communication in error, please notify the sender immediately. The
> views
>> and opinions expressed in this message are those of the individual
> sender of
>> this message and do not necessarily represent the views and opinions
> of
>> ActiCom. Consequently, ActiCom does not accept responsibility for such
> views
>> and opinions and this message should not be read as representing the
> views
>> and opinions of ActiCom without subsequent written confirmation. Each
> page
>> attached hereto must also be read in conjunction with this disclaimer.
>
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
>
> --
> 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
>>
>>
>> _______________________________________________
>> 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
>>
>
>
> ------------------------------
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> End of Asterisk-Users Digest, Vol 9, Issue 209
> **********************************************
>
>
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