[Asterisk-Users] ASTERISK PROGRAMER
Bob Goddard
asterisk at bgcomp.co.uk
Sat Apr 23 11:23:05 MST 2005
On Saturday 23 April 2005 19:13, Matt Klein wrote:
> $4,172.38 USD and I'll programin anything you want for asterisk server.
You are too stupid for the job.
> 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
> >
> > -------------- next part --------------
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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
> >
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam? Yahoo! Mail has the best spam protection around
> > http://mail.yahoo.com
> >
> >
> > ------------------------------
> >
> > 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
> > http://mail.yahoo.com
> >
> >
> > ------------------------------
> >
> > 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 $.50/min regardless of destination, a flat fee for all long
> >>> distance and international.
> >>> This is so cool.
> >>
> >> Hi Chris
> >>
> >> Grouping is a good idea, will not be in the first release, but later.
> >>
> >> There will only be a charge rate in the first release. You
> >> can charge depending on the destination.
> >>
> >> Thorben
> >>
> >>
> >>
> >> _______________________________________________
> >> Asterisk-Users mailing list
> >> Asterisk-Users at lists.digium.com
> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >> To UNSUBSCRIBE or update options visit:
> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
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> > ------------------------------
> >
> > Message: 6
> > Date: Sat, 23 Apr 2005 16:48:25 +0100
> > From: "Steve Rawlings" <steve at rawlings.demon.co.uk>
> > Subject: Re: [Asterisk-Users] Hotel billing in IPSwitchBoard
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > <asterisk-users at lists.digium.com>
> > Message-ID: <000601c5481b$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.
> >
> > -------------- next part --------------
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> > Type: application/x-pkcs7-signature
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> > Url :
> > http://lists.digium.com/pipermail/asterisk-users/attachments/20050423/f1
> > 952746/smime-0001.bin
> >
> > ------------------------------
> >
> > 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
> > **********************************************
> >
> >
> > _______________________________________________
> > 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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