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 > > [EMAIL PROTECTED] > > Tel. (504) 221-4062 (Honduras > > Tel. (507) 322-2259 (PanamÃ) > > Tel. (866) 978-0976 (U.S.A.) > > > > ******************************************** > > > > CONFIDENCIALIDAD. El contenido de esta comunicaciÃn, asà como el de toda > > la documentaciÃn anexa, es confidencial y va dirigido Ãnicamente al > > destinatario del mismo. En el supuesto de que usted no fuera el > > destinatario, le solicitamos que nos lo indique y no comunique su > > contenido a terceros, procediendo a su destrucciÃn. > > > > CONFIDENCIALITY. The content of this communication and any attached > > information is confidential and exclusively for the use of the > > addressee. If you are not the addressee, we ask you to notify to the > > sender and do not pass its content to another person, and please be sure > > you destroy it. > > > > > > -----Mensaje original----- > > De: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] En nombre de > > [EMAIL PROTECTED] > > Enviado el: SÃbado, 23 de Abril de 2005 11:00 a.m. > > Para: asterisk-users@lists.digium.com > > Asunto: Asterisk-Users Digest, Vol 9, Issue 209 > > > > Send Asterisk-Users mailing list submissions to > > asterisk-users@lists.digium.com > > > > To subscribe or unsubscribe via the World Wide Web, visit > > http://lists.digium.com/mailman/listinfo/asterisk-users > > or, via email, send a message with subject or body 'help' to > > [EMAIL PROTECTED] > > > > You can reach the person managing the list at > > [EMAIL PROTECTED] > > > > When replying, please edit your Subject line so it is more specific > > than "Re: Contents of Asterisk-Users digest..." > > > > > > 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" > > <asterisk-users@lists.digium.com> > > 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? > >> > >> _______________________________________________ > >> Asterisk-Users mailing list > >> Asterisk-Users@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 -------------- > > A non-text attachment was scrubbed... > > Name: smime.p7s > > Type: application/x-pkcs7-signature > > Size: 3032 bytes > > Desc: not available > > Url : > > http://lists.digium.com/pipermail/asterisk-users/attachments/20050423/3b > > f4397b/smime-0001.bin > > > > ------------------------------ > > > > 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 > > <asterisk-users@lists.digium.com> > > 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. > >> > >> > >> _______________________________________________ > >> Asterisk-Users mailing list > >> Asterisk-Users@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 <[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 > > <asterisk-users@lists.digium.com> > > 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 > > > > __________________________________________________ > > 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" <[EMAIL PROTECTED]> > > Subject: RE: [Asterisk-Users] Cisco 7960 won't register as SIP device > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > <asterisk-users@lists.digium.com>, "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" > > <asterisk-users@lists.digium.com> > > 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: asterisk-users@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@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@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 -------------- > > A non-text attachment was scrubbed... > > Name: not available > > Type: application/ms-tnef > > Size: 6688 bytes > > Desc: not available > > Url : > > http://lists.digium.com/pipermail/asterisk-users/attachments/20050424/68 > > c7f765/attachment-0001.bin > > > > ------------------------------ > > > > 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" > > <asterisk-users@lists.digium.com> > > 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'" > > <asterisk-users@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" <[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" > > <asterisk-users@lists.digium.com>, > > "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" > > <asterisk-users@lists.digium.com> > > 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. > > > > -------------- next part -------------- > > A non-text attachment was scrubbed... > > Name: smime.p7s > > Type: application/x-pkcs7-signature > > Size: 3032 bytes > > Desc: not available > > 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 <[EMAIL PROTECTED]> > > Subject: Re: [Asterisk-Users] Quadbri & bristuff: can * respond only > > to 1 MSN and leave 1 number to other ISDN phones ? > > To: asterisk-users@lists.digium.com > > 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: asterisk-users@lists.digium.com > > 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'" > > <asterisk-users@lists.digium.com> > > 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'" > >> <asterisk-users@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@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 <[EMAIL PROTECTED]> > > Subject: [Fwd: FW: [Asterisk-Users] IAX help] > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > <asterisk-users@lists.digium.com> > > 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] > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@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" <[EMAIL PROTECTED]> > > Subject: [Asterisk-Users] Re: Re: Hotel billing in IPSwitchBoard > > To: asterisk-users@lists.digium.com > > 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 > > <asterisk-users@lists.digium.com> > > 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 > >> > >> > >> > >> 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@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 <[EMAIL PROTECTED]> > > Subject: Re: [Fwd: FW: [Asterisk-Users] IAX help] > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > <asterisk-users@lists.digium.com> > > 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 > > <asterisk-users@lists.digium.com> > > 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 > >> > >> > >> _______________________________________________ > >> Asterisk-Users mailing list > >> Asterisk-Users@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@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@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@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