Re: [Astlinux-users] after a hard power-down by unplugging the power cord and the pbx won't boot

[nedi]

Hello Michael,
Thanks
I tried using the same USB stick on an old Alix board, and my serial connection 
works with a baud rate of 19200. However, when I use the Alix2e13 board, I'm 
only getting strange characters. After changing the baud rate to 38400, it 
works correctly. I've noticed that the Alix2e13 board's serial connection only 
works with a baud rate of 38400.I can see the boot process and error message 
when I try to boot from my USB install drive on the Alix2e13 board, I encounter 
the error message "01f0 no drive found! No booting device available." 
Interestingly, the same USB drive successfully boots on the Alix1 board. Any 
ideas on how to resolve this issue?

if I use a CF Card wit installer ( serial iso ) , I can boot must change  to 
19200 baud  and the installer on the Alix2e13 board works  but I cant install 
on this boar without USB boot stick that work.  Tomorow i will try install it 
on alix1  board to the cf card and after instal try it in Alix2e13.

I would prefer to use a USB stick.  

Is there a way to run the Alix1 image on CF Card to  the Alix2e13 board and 
make the necessary changes to enable booting and troubleshoot through the 
terminal?

nedi

> Am 26.09.2023 um 21:53 schrieb Michael Keuter :
> 
> 
> Screen command set baud rate for terminal communication
> cyberciti.biz
>  
> <https://www.cyberciti.biz/faq/unix-linux-apple-osx-bsd-screen-set-baud-rate/>Screen
>  command set baud rate for terminal communication 
> <https://www.cyberciti.biz/faq/unix-linux-apple-osx-bsd-screen-set-baud-rate/>
> cyberciti.biz 
> <https://www.cyberciti.biz/faq/unix-linux-apple-osx-bsd-screen-set-baud-rate/>
> 19200 8N1 is the right baudrate for Alix on the 1.2.10 ISO.
> 
> userdoc:legacy-installer-iso [AstLinux Documentation]
> doc.astlinux-project.org
> 
>  
> <http://doc.astlinux-project.org/userdoc:legacy-installer-iso>userdoc:legacy-installer-iso
>  [AstLinux Documentation] 
> <http://doc.astlinux-project.org/userdoc:legacy-installer-iso>
> doc.astlinux-project.org 
> <http://doc.astlinux-project.org/userdoc:legacy-installer-iso>   
>  
> <http://doc.astlinux-project.org/userdoc:legacy-installer-iso>
> 
> Sent from a mobile device.
> 
> Michael Keuter
> 
>> Am 26.09.2023 um 21:33 schrieb nedi :
>> 
>> Hello Michael,
>> 
>> I've tried everything, but I can't seem to read the USB stick.
>> 
>> I have two Alix devices with 3 LAN ports and a serial connection (Alix2e13). 
>> I'm interested in migrating all the settings to the same AstLinux version. 
>> Is this possible?
>> 
>> I attempted to use the AstLinux 1.2.10 serial ISO with a serial terminal, 
>> but I'm only getting strange characters when trying to access AstLinux via a 
>> serial connection on my Mac using both the "serial" app and the Mac Terminal.
>> 
>> I have a USB-to-serial adapter and I've tried the following commands:
>> 
>>• screen /dev/cu.usbserial 19200
>>• screen /dev/cu.usbserial 19200,cs8
>>• screen /dev/cu.usbserial 19200n8
>>• screen /dev/cu.usbserial 115200
>> Is this the correct ISO for the Alix2e13? Also, is it possible to restore 
>> the last backup on this new installation?
>> 
>> Best regards, Nedi
>>> Am 26.09.2023 um 11:23 schrieb Michael Keuter :
>>> 
>>> 
>>>> Am 26.09.2023 um 10:53 schrieb nedi :
>>>> 
>>>> Hello,
>>>> 
>>>> I have an Astlinux installation using the old astlinux astlinux-1.2.8 i586 
>>>> - Asterisk 1.8.32.3 and alix board. 
>>>> Astlinus is on Kingston USB stick. The customer do a hard power-down by 
>>>> unplugging the power cord and the pbx won't boot. has anyone the soulution 
>>>> to fix that. 
>>>> on boot I  get errors
>>>> squashfs error  unable to read dpage ,block 
>>>> and read on /mnt/root/bzimage failed :input output error
>>>> 
>>>> I hope someone can help me to fix that without new install
>>>> regards nedi 
>>> 
>>> 
>>> In the Syslinux bootloader you can select "shell" (instead of the default 
>>> "runnix") to skip booting into AstLinux and try to repair the filesystem 
>>> with "fsck":
>>> 
>>> https://doc.astlinux-project.org/userdoc:system-boot-process
>>> 
>>> Michael
>>> 
>>> http://www.mksolutions.info
>>> 
>>> 
>>> 
>>> 
>>> 
>>> ___
>>> Astlinux-users mailing list
>>>

Re: [Astlinux-users] after a hard power-down by unplugging the power cord and the pbx won't boot

[nedi]

Hello Michael,

I've tried everything, but I can't seem to read the USB stick.

I have two Alix devices with 3 LAN ports and a serial connection (Alix2e13). 
I'm interested in migrating all the settings to the same AstLinux version. Is 
this possible?

I attempted to use the AstLinux 1.2.10 serial ISO with a serial terminal, but 
I'm only getting strange characters when trying to access AstLinux via a serial 
connection on my Mac using both the "serial" app and the Mac Terminal.

I have a USB-to-serial adapter and I've tried the following commands:

• screen /dev/cu.usbserial 19200
• screen /dev/cu.usbserial 19200,cs8
• screen /dev/cu.usbserial 19200n8
• screen /dev/cu.usbserial 115200
Is this the correct ISO for the Alix2e13? Also, is it possible to restore the 
last backup on this new installation?

Best regards, Nedi
> Am 26.09.2023 um 11:23 schrieb Michael Keuter :
> 
> 
>> Am 26.09.2023 um 10:53 schrieb nedi :
>> 
>> Hello,
>> 
>> I have an Astlinux installation using the old astlinux astlinux-1.2.8 i586 - 
>> Asterisk 1.8.32.3 and alix board. 
>> Astlinus is on Kingston USB stick. The customer do a hard power-down by 
>> unplugging the power cord and the pbx won't boot. has anyone the soulution 
>> to fix that. 
>> on boot I  get errors
>> squashfs error  unable to read dpage ,block 
>> and read on /mnt/root/bzimage failed :input output error
>> 
>> I hope someone can help me to fix that without new install
>> regards nedi 
> 
> 
> In the Syslinux bootloader you can select "shell" (instead of the default 
> "runnix") to skip booting into AstLinux and try to repair the filesystem with 
> "fsck":
> 
> https://doc.astlinux-project.org/userdoc:system-boot-process
> 
> Michael
> 
> http://www.mksolutions.info
> 
> 
> 
> 
> 
> ___
> Astlinux-users mailing list
> Astlinux-users@lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to 
> pay...@krisk.org.



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[Astlinux-users] after a hard power-down by unplugging the power cord and the pbx won't boot

[nedi]

Hello,

I have an Astlinux installation using the old astlinux astlinux-1.2.8 i586 - 
Asterisk 1.8.32.3 and alix board. 
Astlinus is on Kingston USB stick. The customer do a hard power-down by 
unplugging the power cord and the pbx won't boot. has anyone the soulution to 
fix that. 
on boot I  get errors
squashfs error  unable to read dpage ,block 
and read on /mnt/root/bzimage failed :input output error

I hope someone can help me to fix that without new install
regards nedi 



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[Astlinux-users] Windows SIP-Client with BLF and Speeddial buttons

[nedi]

Hi Michael, can be this  CTI with SIP Phone, I use it for one Customer with Astlinux and snom, but this CTI have  SIP Client to.https://www.phonesuite.de/hlp/de/client/topics/sip_softphone.htmRegards Nedi

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Re: [Astlinux-users] Windows SIP-Client with BLF and Speeddial buttons

[nedi]

Hi Michael, 
do you know how to make working BFL trough OpenVPN, and I can’t Provisioning 
trough OpenVPN.
can be I have some Routing issue?

Regards Nedi 

> Am 25.04.2021 um 11:31 schrieb Michael Keuter :
> 
> 
> 
>> Am 23.04.2021 um 08:53 schrieb nedi :
>> 
>> 
>> Hi Michael, 
>> can be this  CTI with SIP Phone, I use it for one Customer with Astlinux and 
>> snom, but this CTI have  SIP Client to.
>> 
>> https://www.phonesuite.de/hlp/de/client/topics/sip_softphone.htm
>> 
>> Regards Nedi
> 
> Yes, the "Phonesuite CTI Client" can be used for CTI (TAPI) via the Asterisk 
> Manager Interface (AMI) and it has an integrated softphone which works fine 
> with Asterisk. There are 2 versions of the client:
> 
> https://www.phonesuite.de/de/produkte/client/functions.htm
> 
> I'm not sure, if the BLF keys in the client work without the (optional) 
> Phonesuite CTI server. There is a separate license for "Präsenz-Management". 
> To be sure I would call the programmer (he speaks German).
> 
> https://www.phonesuite.de/de/kontakt.htm
> 
> Michael
> 
> http://www.mksolutions.info
> 
> 
> 
> 
> 
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> 
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> pay...@krisk.org.



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[Astlinux-users] Windows SIP-Client with BLF and Speeddial buttons

[nedi]

Hi Michael, 
can be this  CTI with SIP Phone, I use it for one Customer with Astlinux and 
snom, but this CTI have  SIP Client to.

https://www.phonesuite.de/hlp/de/client/topics/sip_softphone.htm 


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Re: [Astlinux-users] Astlinux Webgui Update

[nedi]

Thanks Michael,
O. K. I don't know that.

I updated trough CLI  both astlinux and RUNIX and get new release after 
rebooted. The Webgui was still old,  and after that I copied ower alternate 
webgui tar and owerwrited files  in www.
After copy I have new gui.

regards Nedi

Am 20. Apr. 2021, 22:42, um 22:42, Michael Keuter  
schrieb:
>The WebGUI is updated automatically when you update the AstLinux
>distro.
>The tar.gz files are only needed if your want to use a custom WebGUI
>(or test something).
>
>Sent from a mobile device.
>
>Michael Keuter
>
>> Am 20.04.2021 um 18:50 schrieb nedi :
>>
>> Hi,
>> i updated one  PBX to the : astlinux-1.2.8 i586 - Asterisk
>1.8.32.3   Runnix Release: runnix-0.4-8057
>>
>> For the Webgui update Im not sure should I untar  both tar files or
>only the update file?
>> Can I untar and copy all file to my www folder as sample
>/stat/var/www  folder
>>
>> my  HTTPSDIR  is /stat/var/www
>>
>> AstLinux 1.0.0 through 1.3.6 GUI Version: 1.8.49
>> Untar's into altweb/ directory: altweb.tar.gz
>> Untar's into current (./) directory: altweb-update.tar.gz
>>
>> Regards Nedi
>> ___
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>> Astlinux-users@lists.sourceforge.net
>> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>>
>> Donations to support AstLinux are graciously accepted via PayPal to
>pay...@krisk.org.
>
>
>
>
>
>
>
>
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>
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>pay...@krisk.org.
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[Astlinux-users] Astlinux Webgui Update

[nedi]

Hi, 
i updated one  PBX to the : astlinux-1.2.8 i586 - Asterisk 1.8.32.3 Runnix 
Release: runnix-0.4-8057

For the Webgui update Im not sure should I untar  both tar files or only the 
update file?  
Can I untar and copy all file to my www folder as sample  /stat/var/www  folder

my  HTTPSDIR  is/stat/var/www

AstLinux 1.0.0 through 1.3.6 GUI Version: 1.8.49 
Untar's into altweb/ directory: altweb.tar.gz 
 
Untar's into current (./) directory: altweb-update.tar.gz 
 

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Re: [Astlinux-users] how to confogure OpenVPN on Astlinux for Snom Phone

[nedi]

Hi, 
can anyone tell me how easy to update astlinux from: AstLinux Release:  
astlinux-1.2.4.1 - Asterisk 1.8.32.3Runnix Release: runnix-0.4-6956  GUI 
Version:   1.8.21
to: AstLinux Release:astlinux-1.2.6.1 i586 - Asterisk 1.8.32.3 Runnix 
Release:runnix-0.4-7671  GUI Version:1.8.40
by one PBX I have openvpn config downloaded and there is no openvpn config file 
only key and cert.
Regards nedi 


> Am 15.04.2021 um 23:19 schrieb nedi :
> 
> Hi Michael,
> Thanks,
>  I get it working with putting  this to my vpn.cnf on snom phone
> dhcp-option DNS 10.0.0.1
> route 10.0.0.0 255.255.255.0
> 
> 
> 
> remote xx.xx.xx.xx 1194 udp
> comp-lzo yes
> cipher AES-256-CBC
> auth SHA1
> key-direction 1
> client
> ns-cert-type server
> nobind
> persist-key
> persist-tun
> dev tun
> verb 3
> dhcp-option DNS 10.0.0.1
> route 10.0.0.0 255.255.255.0
> redirect-gateway def1
> ca /openvpn/ca.crt
> cert /openvpn/client.crt
> key /openvpn/client.key
> 
> Regards Nedi 
> 
>> Am 13.04.2021 um 07:12 schrieb Michael Keuter :
>> 
>> 
>> 
>>> Am 12.04.2021 um 21:32 schrieb nedi :
>>> 
>>> Hi Michael, 
>>> i don't understand you exactly
>>> 
>>> I have NTP Server ch.pool.ntp.org
>>> 
>>> I have in my sip.conf
>>> 
>>> deny = 0.0.0.0/0.0.0.0
>>> permit = 10.0.0.0/255.255.255.0
>>> permit = 10.8.0.0/255.255.255.0
>>> permit = 10.10.11.0/255.255.255.0
>>> 
>>> you mean i must put into my sip.conf  under [general] localnet for all 
>>> network’s  to? Or only localnet  and remove this with deny and permit?
>>> after nat=yes?
>>> localnet = 10.0.0.0/255.255.255.0
>>> localnet = 10.8.0.0/255.255.255.0
>>> localnet = 10.10.11.0/255.255.255.0
>> 
>> Looks good.
>> 
>>> I don’t understand must configure phone to register to IP adress 10.10.11.? 
>>>  if my pbx is 10.0.0.132?
>> 
>> This is the IP address of your OpenVPN server (possibly 10.10.11.1). The 
>> phone does not need to know anything else about your network (e.g. other 
>> routes).
>> 
>>> all others clinets on mac and android working only snom not.
>>> 
>>> regards Nedi
>>> 
>>> 
>>>> Am 12.04.2021 um 13:54 schrieb Michael Keuter :
>>>> 
>>>> 
>>>> 
>>>>> Am 12.04.2021 um 13:48 schrieb Michael Keuter :
>>>>> 
>>>>> 
>>>>> 
>>>>>> Am 12.04.2021 um 13:01 schrieb nedi :
>>>>>> 
>>>>>> Hi,
>>>>>> I have my snom phone connected to the PBX trough OpenVPN, (on the 
>>>>>> display I see VPN  Active, on PBX VPN Status is User1 connected but I 
>>>>>> can’t make provisioning and can't  register, what can bee the issues?
>>>>>> My Macbook or Android phone with SIP Client work trough this OpenVPN 
>>>>>> with the same VPN 
>>>>>> settings.
>>>>>> 
>>>>>> My lan PBX is 10.0.0.132
>>>>>> My virtual Network IP for VPN Client is 10.10.11.2
>>>>>> My LTE Router for testing VPN is 192.168.1.1
>>>>>> 
>>>>>> what must be in PUSH section  of my PBX VPN Config?
>>>>>> 
>>>>>> I have This
>>>>>> dhcp-option DNS 10.0.0.1
>>>>>> route 10.0.0.0 255.255.255.0
>>>>>> redirect-gateway def1
>>>>>> 
>>>>>> 
>>>>>> OpenVPN Status on PBX 
>>>>>> 
>>>>>> 
>>>>>> User1194.230.148.217:618410.10.11.2  41824520
>>>>>> Mon Apr 12 10:47:57 20211618217277
>>>>>> 
>>>>>> in sip.conf   general I have this
>>>>>> 
>>>>>> alwaysauthreject=yes
>>>>>> deny = 0.0.0.0/0.0.0.0
>>>>>> permit = 10.0.0.0/255.255.255.0
>>>>>> permit = 10.8.0.0/255.255.255.0
>>>>>> permit = 10.10.11.0/255.255.255.0
>>>>>> 
>>>>>> regards Nedi
>>>>> 
>>>>> Hi Nedi,
>>>>> 
>>>>> important is that the phone registers to Asterisk on the virtual IP 
>>>>> "10.10.11.x" and not on 10.0.0.132!
>>>>> 
>>>>> You also need to add "localnet" in sip.conf for this virtual IP range in 
>>>>> the NAT section.
>

Re: [Astlinux-users] how to confogure OpenVPN on Astlinux for Snom Phone

[nedi]

Hi Michael,
Thanks,
 I get it working with putting  this to my vpn.cnf on snom phone
dhcp-option DNS 10.0.0.1
route 10.0.0.0 255.255.255.0



remote xx.xx.xx.xx 1194 udp
comp-lzo yes
cipher AES-256-CBC
auth SHA1
key-direction 1
client
ns-cert-type server
nobind
persist-key
persist-tun
dev tun
verb 3
dhcp-option DNS 10.0.0.1
route 10.0.0.0 255.255.255.0
redirect-gateway def1
ca /openvpn/ca.crt
cert /openvpn/client.crt
key /openvpn/client.key

Regards Nedi 

> Am 13.04.2021 um 07:12 schrieb Michael Keuter :
> 
> 
> 
>> Am 12.04.2021 um 21:32 schrieb nedi mailto:n...@gmx.ch>>:
>> 
>> Hi Michael, 
>> i don't understand you exactly
>> 
>> I have NTP Server ch.pool.ntp.org <http://ch.pool.ntp.org/>
>> 
>> I have in my sip.conf
>> 
>> deny = 0.0.0.0/0.0.0.0
>> permit = 10.0.0.0/255.255.255.0
>> permit = 10.8.0.0/255.255.255.0
>> permit = 10.10.11.0/255.255.255.0
>> 
>> you mean i must put into my sip.conf  under [general] localnet for all 
>> network’s  to? Or only localnet  and remove this with deny and permit?
>> after nat=yes?
>> localnet = 10.0.0.0/255.255.255.0
>> localnet = 10.8.0.0/255.255.255.0
>> localnet = 10.10.11.0/255.255.255.0
> 
> Looks good.
> 
>> I don’t understand must configure phone to register to IP adress 10.10.11.?  
>> if my pbx is 10.0.0.132?
> 
> This is the IP address of your OpenVPN server (possibly 10.10.11.1). The 
> phone does not need to know anything else about your network (e.g. other 
> routes).
> 
>> all others clinets on mac and android working only snom not.
>> 
>> regards Nedi
>> 
>> 
>>> Am 12.04.2021 um 13:54 schrieb Michael Keuter >> <mailto:li...@mksolutions.info>>:
>>> 
>>> 
>>> 
>>>> Am 12.04.2021 um 13:48 schrieb Michael Keuter >>> <mailto:li...@mksolutions.info>>:
>>>> 
>>>> 
>>>> 
>>>>> Am 12.04.2021 um 13:01 schrieb nedi mailto:n...@gmx.ch>>:
>>>>> 
>>>>> Hi,
>>>>> I have my snom phone connected to the PBX trough OpenVPN, (on the display 
>>>>> I see VPN  Active, on PBX VPN Status is User1 connected but I can’t make 
>>>>> provisioning and can't  register, what can bee the issues?
>>>>> My Macbook or Android phone with SIP Client work trough this OpenVPN with 
>>>>> the same VPN 
>>>>> settings.
>>>>> 
>>>>> My lan PBX is 10.0.0.132
>>>>> My virtual Network IP for VPN Client is 10.10.11.2
>>>>> My LTE Router for testing VPN is 192.168.1.1
>>>>> 
>>>>> what must be in PUSH section  of my PBX VPN Config?
>>>>> 
>>>>> I have This
>>>>> dhcp-option DNS 10.0.0.1
>>>>> route 10.0.0.0 255.255.255.0
>>>>> redirect-gateway def1
>>>>> 
>>>>> 
>>>>> OpenVPN Status on PBX 
>>>>> 
>>>>> 
>>>>> User1 194.230.148.217:618410.10.11.2  41824520
>>>>> Mon Apr 12 10:47:57 20211618217277
>>>>> 
>>>>> in sip.conf   general I have this
>>>>> 
>>>>> alwaysauthreject=yes
>>>>> deny = 0.0.0.0/0.0.0.0
>>>>> permit = 10.0.0.0/255.255.255.0
>>>>> permit = 10.8.0.0/255.255.255.0
>>>>> permit = 10.10.11.0/255.255.255.0
>>>>> 
>>>>> regards Nedi
>>>> 
>>>> Hi Nedi,
>>>> 
>>>> important is that the phone registers to Asterisk on the virtual IP 
>>>> "10.10.11.x" and not on 10.0.0.132!
>>>> 
>>>> You also need to add "localnet" in sip.conf for this virtual IP range in 
>>>> the NAT section.
>>>> For provisioning to work you need to add the virtual IP range to "HTTP & 
>>>> HTTPS /phoneprov/ Allowed IP's:" (if not all (*) is allowed) and restart 
>>>> AstLinux.
>>> 
>>> Update: and you need an external time server on the IP-phone and not the 
>>> internal one from AstLinux (e.g. "europe.pool.ntp.org 
>>> <http://europe.pool.ntp.org/>")
>>> 
>>>>>> Am 10.04.2021 um 18:04 schrieb nedi mailto:n...@gmx.ch>>:
>>>>>> 
>>>>>> Hi , 
>>>>>> has anyone working config for the snom phones and astlinux openVPN i 
>>>>>> tried and tried , it 

Re: [Astlinux-users] how to confogure OpenVPN on Astlinux for Snom Phone

[nedi]

Hi Michael, 
i don't understand you exactly

I have NTP Server ch.pool.ntp.org

I have in my sip.conf

deny = 0.0.0.0/0.0.0.0
permit = 10.0.0.0/255.255.255.0
permit = 10.8.0.0/255.255.255.0
permit = 10.10.11.0/255.255.255.0

you mean i must put into my sip.conf  under [general] localnet for all 
network’s  to? Or only localnet  and remove this with deny and permit?
after nat=yes?
localnet = 10.0.0.0/255.255.255.0
localnet = 10.8.0.0/255.255.255.0
localnet = 10.10.11.0/255.255.255.0

I don’t understand must configure phone to register to IP adress 10.10.11.?  if 
my pbx is 10.0.0.132?
all others clinets on mac and android working only snom not.

regards Nedi


> Am 12.04.2021 um 13:54 schrieb Michael Keuter :
> 
> 
> 
>> Am 12.04.2021 um 13:48 schrieb Michael Keuter :
>> 
>> 
>> 
>>> Am 12.04.2021 um 13:01 schrieb nedi :
>>> 
>>> Hi,
>>> I have my snom phone connected to the PBX trough OpenVPN, (on the display I 
>>> see VPN  Active, on PBX VPN Status is User1 connected but I can’t make 
>>> provisioning and can't  register, what can bee the issues?
>>> My Macbook or Android phone with SIP Client work trough this OpenVPN with 
>>> the same VPN 
>>> settings.
>>> 
>>> My lan PBX is 10.0.0.132
>>> My virtual Network IP for VPN Client is 10.10.11.2
>>> My LTE Router for testing VPN is 192.168.1.1
>>> 
>>> what must be in PUSH section  of my PBX VPN Config?
>>> 
>>> I have This
>>> dhcp-option DNS 10.0.0.1
>>> route 10.0.0.0 255.255.255.0
>>> redirect-gateway def1
>>> 
>>> 
>>> OpenVPN Status on PBX 
>>> 
>>> 
>>> User1   194.230.148.217:618410.10.11.2  41824520    
>>> Mon Apr 12 10:47:57 20211618217277
>>> 
>>> in sip.conf   general I have this
>>> 
>>> alwaysauthreject=yes
>>> deny = 0.0.0.0/0.0.0.0
>>> permit = 10.0.0.0/255.255.255.0
>>> permit = 10.8.0.0/255.255.255.0
>>> permit = 10.10.11.0/255.255.255.0
>>> 
>>> regards Nedi
>> 
>> Hi Nedi,
>> 
>> important is that the phone registers to Asterisk on the virtual IP 
>> "10.10.11.x" and not on 10.0.0.132!
>> 
>> You also need to add "localnet" in sip.conf for this virtual IP range in the 
>> NAT section.
>> For provisioning to work you need to add the virtual IP range to "HTTP & 
>> HTTPS /phoneprov/ Allowed IP's:" (if not all (*) is allowed) and restart 
>> AstLinux.
> 
> Update: and you need an external time server on the IP-phone and not the 
> internal one from AstLinux (e.g. "europe.pool.ntp.org")
> 
>>>> Am 10.04.2021 um 18:04 schrieb nedi :
>>>> 
>>>> Hi , 
>>>> has anyone working config for the snom phones and astlinux openVPN i tried 
>>>> and tried , it works with MacBook and Asttlinux OpenVPN, 
>>>> snom won’t connecting , what can I do to get it working?
>>>> 
>>>> I putting ip adress of vpn server into vpn.cnf
>>>> This is my snom vpn config:
>>>> remote IP Adress of my PBX OpenVPN Server  1194 udp
>>>> comp-lzo yes
>>>> cipher AES-256-CBC
>>>> key-direction 1
>>>> client
>>>> ns-cert-type server
>>>> nobind
>>>> persist-key
>>>> persist-tun
>>>> dev tun
>>>> verb 3
>>>> 
>>>> -BEGIN CERTIFICATE-
>>>> …..my cert , key, and tl….
>>>> 
>>>> After not working i tried this vpn.cnf  to put cert key  separately as 
>>>> described by snom wiki  putting all files into openvpn folder and  made 
>>>> tarbal of them 
>>>> 
>>>> remote IP Adress of my PBX OpenVPN Server  1194 udp
>>>> comp-lzo yes
>>>> cipher AES-256-CBC
>>>> key-direction 1
>>>> client
>>>> ns-cert-type server
>>>> nobind
>>>> persist-key
>>>> persist-tun
>>>> dev tun
>>>> verb 3
>>>> ca /openvpn/ca.crt
>>>> cert /openvpn/client.crt
>>>> key /openvpn/client.key 
>>>> 
>>>> Thanks
>>>> 
>>>> Regards nedi
>>>> 
>>>> 
>>>>> Am 08.04.2021 um 23:22 schrieb Michael Keuter :
>>>>> 
>>>>> 
>>>>> 
>>>>>> Am 08.04.2021 um 22:59 schrieb nedi :
>>>>>> 
>>>>>> Hi MIchael, 
>>>&

Re: [Astlinux-users] how to confogure OpenVPN on Astlinux for Snom Phone

[nedi]

Hi,
I have my snom phone connected to the PBX trough OpenVPN, (on the display I see 
VPN  Active, on PBX VPN Status is User1 connected but I can’t make provisioning 
and can't  register, what can bee the issues?
My Macbook or Android phone with SIP Client work trough this OpenVPN with the 
same VPN 
settings.

My lan PBX is 10.0.0.132
My virtual Network IP for VPN Client is 10.10.11.2
My LTE Router for testing VPN is 192.168.1.1

what must be in PUSH section  of my PBX VPN Config?

I have This
dhcp-option DNS 10.0.0.1
route 10.0.0.0 255.255.255.0
redirect-gateway def1


OpenVPN Status on PBX 


User1   194.230.148.217:618410.10.11.2  41824520Mon Apr 
12 10:47:57 20211618217277

in sip.conf   general I have this

alwaysauthreject=yes
deny = 0.0.0.0/0.0.0.0
permit = 10.0.0.0/255.255.255.0
permit = 10.8.0.0/255.255.255.0
permit = 10.10.11.0/255.255.255.0

regards Nedi

> Am 10.04.2021 um 18:04 schrieb nedi :
> 
> Hi , 
> has anyone working config for the snom phones and astlinux openVPN i tried 
> and tried , it works with MacBook and Asttlinux OpenVPN, 
> snom won’t connecting , what can I do to get it working?
> 
> I putting ip adress of vpn server into vpn.cnf
> This is my snom vpn config:
> remote IP Adress of my PBX OpenVPN Server  1194 udp
> comp-lzo yes
> cipher AES-256-CBC
> key-direction 1
> client
> ns-cert-type server
> nobind
> persist-key
> persist-tun
> dev tun
> verb 3
> 
> -BEGIN CERTIFICATE-
> …..my cert , key, and tl….
> 
> After not working i tried this vpn.cnf  to put cert key  separately as 
> described by snom wiki  putting all files into openvpn folder and  made 
> tarbal of them 
> 
> remote IP Adress of my PBX OpenVPN Server  1194 udp
> comp-lzo yes
> cipher AES-256-CBC
> key-direction 1
> client
> ns-cert-type server
> nobind
> persist-key
> persist-tun
> dev tun
> verb 3
> ca /openvpn/ca.crt
> cert /openvpn/client.crt
> key /openvpn/client.key 
> 
> Thanks
> 
> Regards nedi
> 
> 
>> Am 08.04.2021 um 23:22 schrieb Michael Keuter :
>> 
>> 
>> 
>>> Am 08.04.2021 um 22:59 schrieb nedi :
>>> 
>>> Hi MIchael, 
>>> I need 2 Phoen connect to pbx from outside 
>>> I have this snom Firmware and Patch for VPN flashed as Update,  but this 
>>> not working with my Synology. From Synology I can Export  openvpn config 
>>> file and use on Macbook OpenVpn app  but there is not user.key included and 
>>> user.crt ther are only ca.crt and openvpn.conf  files. 
>> 
>> The Synology OpenVPN server is very limited from the WebGUI.
>> 
>>> I think is not wrong with snom, I can make those tar file .. and flash the 
>>> snome phone.  After that I tried with Astlinux openvpn and forwarded port 
>>> to Astlinux ip  but with Astlinux i can’t Connect from snom, can’t connect 
>>> from my smartphone or macbook. There is no connecting to VPN server. I 
>>> thinK on astlinux side is something wrong.
>>> regards 
>>> nedi
>> 
>> You should definitely get it working first with your Mac, before trying the 
>> snom.
>> 
>> https://doc.astlinux.org/userdoc:tt_openvpn_server
>> 
>> For the snom use "Auth Method" => "Certificate"
>> 
>> When I download the credentials and import the "openvpn-cert-key" *.ovpn 
>> file into Viscosity or Tunnelblick it works fine on a Mac.
>> 
>>> 
>>>> Am 08.04.2021 um 22:36 schrieb Michael Keuter :
>>>> 
>>>> 
>>>> 
>>>>> Am 08.04.2021 um 22:24 schrieb nedi :
>>>>> 
>>>>> Hi, 
>>>>> I tried to configure OpenVpn for the Snom phone without  success on 
>>>>> Astlinux box and on Synology. 
>>>>> On Synology I have VPN working but I think  is not compatible to the Snom 
>>>>> phone I need a key file.
>>>>> 
>>>>> 
>>>>> astlinux-1.2.6.1 i586 - Asterisk 1.8.32.3 Runnix Release: runnix-0.4-7671 
>>>>> GUI Version:1.8.40
>>>>> 
>>>>> Can Anyone help me to configure OpenVPN on Astlinux box. I  Have Alix 
>>>>> with only one Lan Port can be this is the issue because VPN won’t work or 
>>>>> iptables firewall make some issues? 
>>>>> I tried with firewall enabled or disabled, I  rebooted, tried import 
>>>>> openvpn.conf on Macbook. I made port-forwarding .  
>>>>> 
>>>>> In Firewal options is all disabled and I put into firewall  Pass 
>>>>> EXT->Local  UDP 0/0 1194
>> 
>&

Re: [Astlinux-users] how to confogure OpenVPN on Astlinux for Snom Phone

[nedi]

Hi , 
has anyone working config for the snom phones and astlinux openVPN i tried and 
tried , it works with MacBook and Asttlinux OpenVPN, 
snom won’t connecting , what can I do to get it working?

I putting ip adress of vpn server into vpn.cnf
This is my snom vpn config:
remote IP Adress of my PBX OpenVPN Server  1194 udp
comp-lzo yes
cipher AES-256-CBC
key-direction 1
client
ns-cert-type server
nobind
persist-key
persist-tun
dev tun
verb 3

-BEGIN CERTIFICATE-
…..my cert , key, and tl….

After not working i tried this vpn.cnf  to put cert key  separately as 
described by snom wiki  putting all files into openvpn folder and  made tarbal 
of them 

remote IP Adress of my PBX OpenVPN Server  1194 udp
comp-lzo yes
cipher AES-256-CBC
key-direction 1
client
ns-cert-type server
nobind
persist-key
persist-tun
dev tun
verb 3
ca /openvpn/ca.crt
cert /openvpn/client.crt
key /openvpn/client.key 

Thanks

Regards nedi


> Am 08.04.2021 um 23:22 schrieb Michael Keuter :
> 
> 
> 
>> Am 08.04.2021 um 22:59 schrieb nedi :
>> 
>> Hi MIchael, 
>> I need 2 Phoen connect to pbx from outside 
>> I have this snom Firmware and Patch for VPN flashed as Update,  but this not 
>> working with my Synology. From Synology I can Export  openvpn config file 
>> and use on Macbook OpenVpn app  but there is not user.key included and 
>> user.crt ther are only ca.crt and openvpn.conf  files. 
> 
> The Synology OpenVPN server is very limited from the WebGUI.
> 
>> I think is not wrong with snom, I can make those tar file .. and flash the 
>> snome phone.  After that I tried with Astlinux openvpn and forwarded port to 
>> Astlinux ip  but with Astlinux i can’t Connect from snom, can’t connect from 
>> my smartphone or macbook. There is no connecting to VPN server. I thinK on 
>> astlinux side is something wrong.
>> regards 
>> nedi
> 
> You should definitely get it working first with your Mac, before trying the 
> snom.
> 
> https://doc.astlinux.org/userdoc:tt_openvpn_server
> 
> For the snom use "Auth Method" => "Certificate"
> 
> When I download the credentials and import the "openvpn-cert-key" *.ovpn file 
> into Viscosity or Tunnelblick it works fine on a Mac.
> 
>> 
>>> Am 08.04.2021 um 22:36 schrieb Michael Keuter :
>>> 
>>> 
>>> 
>>>> Am 08.04.2021 um 22:24 schrieb nedi :
>>>> 
>>>> Hi, 
>>>> I tried to configure OpenVpn for the Snom phone without  success on 
>>>> Astlinux box and on Synology. 
>>>> On Synology I have VPN working but I think  is not compatible to the Snom 
>>>> phone I need a key file.
>>>> 
>>>> 
>>>> astlinux-1.2.6.1 i586 - Asterisk 1.8.32.3  Runnix Release: runnix-0.4-7671 
>>>> GUI Version:1.8.40
>>>> 
>>>> Can Anyone help me to configure OpenVPN on Astlinux box. I  Have Alix with 
>>>> only one Lan Port can be this is the issue because VPN won’t work or 
>>>> iptables firewall make some issues? 
>>>> I tried with firewall enabled or disabled, I  rebooted, tried import 
>>>> openvpn.conf on Macbook. I made port-forwarding .  
>>>> 
>>>> In Firewal options is all disabled and I put into firewall  Pass 
>>>> EXT->Local  UDP 0/0 1194
> 
> You don't need that, it is done by the openvpn firewall plugin automatically.
> 
>>>> 
>>>> My network  is: 10.0.0.1   DNS:  10.0.0.1  NM:  255.255.255.0ipv4 
>>>> Gateway: 10.0.0.1
>>>> 
>>>> Tunnel Options:
>>>> Protocol: UDPv4Port:   1194
>>>> Log Verbosity: medium  Compression:yes
>>>> QoS Passthrough:YesLegacy Cipher:  
>>>> AES-256-CBC
>>>> Device:tun0Auth 
>>>> HMAC:  Use default
>>>> Raw Commands:
>>>> 
>>>> Authentication:
>>>> Auth Method:   Certificate 
>>>> Extra TLS-Auth:Yes
>>>> 
>>>> Firewall Options:
>>>> External Hosts:0/0
>>>> 
>>>> Server Mode:
>>>> Server Hostname(s):my dyndns
> 
> Is this domain reachable?
> Depending on your network/DNS configuration you might fail to test the VPN 
> connection from your internal network.
> 
> Try testing with your MacBook via a smartphone with Wifi Hotspot and mobile 
> data instead.
> 
>>>> Network IPv4 NM:   10.10.11.0 255.255.255.0
>>>>

Re: [Astlinux-users] how to confogure OpenVPN on Astlinux for Snom Phone

[nedi]

Hi MIchael, 
I need 2 Phoen connect to pbx from outside 
I have this snom Firmware and Patch for VPN flashed as Update,  but this not 
working with my Synology. From Synology I can Export  openvpn config file and 
use on Macbook OpenVpn app  but there is not user.key included and user.crt 
ther are only ca.crt and openvpn.conf  files. 

I think is not wrong with snom, I can make those tar file .. and flash the 
snome phone.  After that I tried with Astlinux openvpn and forwarded port to 
Astlinux ip  but with Astlinux i can’t Connect from snom, can’t connect from my 
smartphone or macbook. There is no connecting to VPN server. I thinK on 
astlinux side is something wrong.
regards 
nedi

> Am 08.04.2021 um 22:36 schrieb Michael Keuter :
> 
> 
> 
>> Am 08.04.2021 um 22:24 schrieb nedi :
>> 
>> Hi, 
>> I tried to configure OpenVpn for the Snom phone without  success on Astlinux 
>> box and on Synology. 
>> On Synology I have VPN working but I think  is not compatible to the Snom 
>> phone I need a key file.
>> 
>> 
>> astlinux-1.2.6.1 i586 - Asterisk 1.8.32.3Runnix Release: runnix-0.4-7671 
>> GUI Version:1.8.40
>> 
>> Can Anyone help me to configure OpenVPN on Astlinux box. I  Have Alix with 
>> only one Lan Port can be this is the issue because VPN won’t work or 
>> iptables firewall make some issues? 
>> I tried with firewall enabled or disabled, I  rebooted, tried import 
>> openvpn.conf on Macbook. I made port-forwarding .  
>> 
>> In Firewal options is all disabled and I put into firewall  Pass EXT->Local  
>> UDP 0/0 1194
>> 
>> My network  is: 10.0.0.1   DNS:  10.0.0.1  NM:  255.255.255.0ipv4 
>> Gateway: 10.0.0.1
>> 
>> Tunnel Options:
>> Protocol: UDPv4  Port:   1194
>> Log Verbosity: mediumCompression:yes
>> QoS Passthrough:Yes  Legacy Cipher:  
>> AES-256-CBC
>> Device:tun0  Auth HMAC:  
>> Use default
>> Raw Commands:
>> 
>> Authentication:
>> Auth Method: Certificate 
>> Extra TLS-Auth:  Yes
>> 
>> Firewall Options:
>> External Hosts:  0/0
>> 
>> Server Mode:
>> Server Hostname(s):  my dyndns
>> Network IPv4 NM: 10.10.11.0 255.255.255.0
>> Network IPv6/nn: 
>> 
>> Topology:subnet latest , requires openvpn 2.1+ clients
>> "push":  dhcp-option DOMAIN priv.mydomain.ch   <==   can be 
>> it is wrong what i have here , do I need this?
>>  dhcp-option DNS 10.10.10.1 
>>  route10.10.10.0 255.255.255.0 
>>  redirect-gateway def1
>> 
>> Server Certificate and Key:
>> 
>> Private Key Size:2048
>> Signature Algorithm: SHA-256
>> 
>> I made 2 Usr and downloaded zip files 
>> 
>> after import into openvpn app won’r connecting.
>> 
>> regards Nedi
> 
> I think some years ago snom removed OpenVPN from their default firmware 
> images. You need a special firmware that enables OpenVPN.
> 
> https://service.snom.com/display/wiki/Configuring+VPN+on+Snom+Deskphones#ConfiguringVPNonSnomDeskphones-InstallandconfigureOpenVPNontheSnomphones.1
> 
> Michael
> 
> http://www.mksolutions.info
> 
> 
> 
> 
> 
> ___
> Astlinux-users mailing list
> Astlinux-users@lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to 
> pay...@krisk.org.



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[Astlinux-users] how to confogure OpenVPN on Astlinux for Snom Phone

[nedi]

Hi, 
I tried to configure OpenVpn for the Snom phone without  success on Astlinux 
box and on Synology. 
On Synology I have VPN working but I think  is not compatible to the Snom phone 
I need a key file.


astlinux-1.2.6.1 i586 - Asterisk 1.8.32.3   Runnix Release: runnix-0.4-7671 
GUI Version:1.8.40

Can Anyone help me to configure OpenVPN on Astlinux box. I  Have Alix with only 
one Lan Port can be this is the issue because VPN won’t work or iptables 
firewall make some issues? 
I tried with firewall enabled or disabled, I  rebooted, tried import 
openvpn.conf on Macbook. I made port-forwarding .  

In Firewal options is all disabled and I put into firewall  Pass EXT->Local  
UDP 0/0 1194

My network  is: 10.0.0.1   DNS:  10.0.0.1  NM:  255.255.255.0ipv4 Gateway: 
10.0.0.1

Tunnel Options:
Protocol: UDPv4 Port:   1194
Log Verbosity: medium   Compression:yes
QoS Passthrough:Yes Legacy Cipher:  
AES-256-CBC
Device:tun0 Auth HMAC:  
Use default
Raw Commands:

Authentication:
Auth Method:Certificate 
Extra TLS-Auth: Yes

Firewall Options:
External Hosts: 0/0

Server Mode:
Server Hostname(s): my dyndns
Network IPv4 NM:10.10.11.0 255.255.255.0
Network IPv6/nn:

Topology:   subnet latest , requires openvpn 2.1+ clients
"push": dhcp-option DOMAIN priv.mydomain.ch   <==   can be it is 
wrong what i have here , do I need this?
dhcp-option DNS 10.10.10.1 
route10.10.10.0 255.255.255.0 
redirect-gateway def1

Server Certificate and Key:

Private Key Size:   2048
Signature Algorithm:SHA-256

I made 2 Usr and downloaded zip files 

after import into openvpn app won’r connecting.

regards Nedi




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[Astlinux-users] old astlinux can't acces trough ssh putty and web pbx wokrs and I can ping it.

[nedi]

Hi, 
can anyone help me to get access on customer astlinux one old version   1.1.2  
and Astrisk 1.8.22.0  the pbx is to far away from me about 300 km. 

i can ping the PBX, The PBX  works but sometime after 5 second the customer 
told me that can't hear the caller. 
I tried with teamviewer to connect to pbx and reboot the pbx and can’t access 
it trough web and trough ssh and putty , I can ping it and get the Answear. 
can be te CF card damaged?
regards
Nedi



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Re: [Astlinux-users] HP t5730 Thin Client

[nedi]

Thanks


> Am 09.10.2020 um 16:47 schrieb Lonnie Abelbeck :
> 
>> Not clear ( to me ) which current build will work.
> 
> I'm guessing the latest AstLinux 1.4.0 may work on this, though the NIC is 
> unknown.  If not, give 1.3.10 a try.
> 
> The limited flash storage of 1 GB will only yield a 512 MB /mnt/kd/ 
> partition, resulting in limited voicemail storage, but may be fine for 
> certain installs.
> 
> For something inexpensive and new, I would also look at PC Engines apu2 (2 
> LAN), currently they have a sale on 2 GB RAM boards ($ 86.00 USD board-only). 
>  With 2x the RAM, 2x NICs and 4x the CPU cores of the AMD Sempron 2100+ (HP 
> t5730).
> 
> USD prices
> https://www.pcengines.ch/newshop.php?c=4
> 
> Lonnie
> 
> 
> 
> 
>> On Oct 9, 2020, at 8:30 AM, John Novack  wrote:
>> 
>> I have several users who have the T5730. one even has an expansion frame 
>> with a T1 card and a custom Asterisk built to work with his WE ANI
>> 
>> It has been in use for a couple years now with no issues. Used with our CNET 
>> collectors network.
>> 
>> Asterisk version 13.??
>> 
>> Not clear ( to me ) which current build will work.
>> 
>> 
>> 
>> John Novack
>> 
>> 
>> nedi wrote:
>>> Hi , 
>>> Has anyone tried Astlinux on HP t5730 Thin Client?
>>> AMD Sempron 2100+ Single Core 1.00GHz / 1GB RAM / 1GB IDE Flash  Microsoft 
>>> Windows XP Embedded 64bit
>>> 
>>> Regards Nedi
>>> 
>>> 
>>> 
>>> 
>>> ___
>>> Astlinux-users mailing list
>>> 
>>> Astlinux-users@lists.sourceforge.net
>>> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>>> 
>>> 
>>> Donations to support AstLinux are graciously accepted via PayPal to 
>>> pay...@krisk.org
>>> .
>>> 
>>> 
>> 
>> -- 
>> Dog is my Co-Pilot
>> 
>> ___
>> Astlinux-users mailing list
>> Astlinux-users@lists.sourceforge.net
>> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>> 
>> Donations to support AstLinux are graciously accepted via PayPal to 
>> pay...@krisk.org.
> 
> 
> 
> ___
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> Astlinux-users@lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to 
> pay...@krisk.org.



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[Astlinux-users] HP t5730 Thin Client

[nedi]

Hi , 
Has anyone tried Astlinux on HP t5730 Thin Client?
AMD Sempron 2100+ Single Core 1.00GHz / 1GB RAM / 1GB IDE Flash  Microsoft 
Windows XP Embedded 64bit

Regards Nedi




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Re: [Astlinux-users] When a call goes to a group of snom phones and is answered by one phone, all the other phones display a missed call notification

[nedi]

Michael do you mean c in dialpilan everywere  or only where I dial all 
extension together?

> Am 16.06.2020 um 00:11 schrieb The Cadillac Kid via Astlinux-users 
> :
> 
> I didnt think about dial options..  
> 
> this is my Dial command
> 
> exten = 
> s,n,Dial(SIP/${receiver}${ARG5},${fbnatimer},rIM(answerme,${receiver},${origuniqueid},${cidoriginate}))
> 
> im not using 'c' in it the 'r' in it pretty much means that 180 will be sent 
> back to the originating phone no matter what. I had to use that as we had 
> some older analog FXS gateways that refused to send ringback
> 
> as for the config I sent its meant for a D700 series..  all of our snoms in 
> the field are 700s. 
> 
> 
> On Monday, June 15, 2020, 4:26:41 PM EDT, Michael Keuter 
>  wrote:
> 
> 
> Have you tried in Asterisk the option „c“ of the „dial“ command?
> 
> Sent from a mobile device.
> 
> Michael Keuter
> 
>> Am 15.06.2020 um 22:21 schrieb nedi :
>> 
> 
>> 
> 
> Thanks , I will try tomorow.
> Regards nedi
> 
>> Am 15.06.2020 um 22:16 schrieb The Cadillac Kid via Astlinux-users 
>> > <mailto:astlinux-users@lists.sourceforge.net>>:
>> 
>> hit send too quick..
>> 
>> this is the SIP sent back to it..   
>> 
>> CANCEL sip:5051@172.16.9.2:54836;line=up7wg15k <> SIP/2.0
>> Via: SIP/2.0/UDP 172.16.8.2:5060;branch=z9hG4bK21641ecc
>> Max-Forwards: 70
>> From: "3200" >;tag=as73a14520
>> To: >
>> Call-ID: 328a1eaf23fc726f695258ef65aa2e51@172.16.8.2 
>> <mailto:328a1eaf23fc726f695258ef65aa2e51@172.16.8.2>:5060
>> CSeq: 102 CANCEL
>> User-Agent: Asterisk PBX 11.20.0
>> Reason: SIP;cause=200;text="Call completed elsewhere"
>> Content-Length: 0
>> 
>> 
>> my call was 3200 dials a ring group with 5002 and 5051(snom),  5002 answers 
>> the phone and 5051 shows answered elsewhere with no missed call  
>> 
>> On Monday, June 15, 2020, 3:54:19 PM EDT, nedi > <mailto:n...@gmx.ch>> wrote:
>> 
>> 
>> The phone showing message  "call completed elsewhere" on the screen  coming 
>> but  missed call is still there on all phones.
>> Nedi
>> 
>>> Am 15.06.2020 um 21:45 schrieb The Cadillac Kid via Astlinux-users 
>>> >> <mailto:astlinux-users@lists.sourceforge.net>>:
>>> 
>>> this was a PJ issue wasnt it? im still using Chan_sip and my snoms dont do 
>>> it if I answer at another phone, they actually show call completed 
>>> elsewhere on the screen...  im using 10.1.49.X firmware on my snoms..  
>>> I dont remember having to set anything different in the config to make it 
>>> work.
>>> 
>>> 
>>> On Monday, June 15, 2020, 3:31:16 PM EDT, nedi >> <mailto:n...@gmx.ch>> wrote:
>>> 
>>> 
>>> Hi,
>>> 
>>> When a call goes to a group of snom phones and is answered by one phone, 
>>> all the other phones display a missed call notification.
>>> 
>>> According to the SNOM FAQ this can be stopped by way of the following 
>>> changes to SIP protocols:
>>> 
>>> Has anyone idea how to fix that
>>> 
>>> Regards 
>>> Nedi
>>> ___
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>>> 
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>> 
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>> Donations to support AstLinux are graci

Re: [Astlinux-users] When a call goes to a group of snom phones and is answered by one phone, all the other phones display a missed call notification

[nedi]

Thanks , I will try tomorow.
Regards nedi

> Am 15.06.2020 um 22:16 schrieb The Cadillac Kid via Astlinux-users 
> :
> 
> hit send too quick..
> 
> this is the SIP sent back to it..   
> 
> CANCEL sip:5051@172.16.9.2:54836;line=up7wg15k SIP/2.0
> Via: SIP/2.0/UDP 172.16.8.2:5060;branch=z9hG4bK21641ecc
> Max-Forwards: 70
> From: "3200" ;tag=as73a14520
> To: 
> Call-ID: 328a1eaf23fc726f695258ef65aa2e51@172.16.8.2:5060
> CSeq: 102 CANCEL
> User-Agent: Asterisk PBX 11.20.0
> Reason: SIP;cause=200;text="Call completed elsewhere"
> Content-Length: 0
> 
> 
> my call was 3200 dials a ring group with 5002 and 5051(snom),  5002 answers 
> the phone and 5051 shows answered elsewhere with no missed call  
> 
> On Monday, June 15, 2020, 3:54:19 PM EDT, nedi  wrote:
> 
> 
> The phone showing message  "call completed elsewhere" on the screen  coming 
> but  missed call is still there on all phones.
> Nedi
> 
>> Am 15.06.2020 um 21:45 schrieb The Cadillac Kid via Astlinux-users 
>> > <mailto:astlinux-users@lists.sourceforge.net>>:
>> 
>> this was a PJ issue wasnt it? im still using Chan_sip and my snoms dont do 
>> it if I answer at another phone, they actually show call completed elsewhere 
>> on the screen...  im using 10.1.49.X firmware on my snoms..  
>> I dont remember having to set anything different in the config to make it 
>> work.
>> 
>> 
>> On Monday, June 15, 2020, 3:31:16 PM EDT, nedi > <mailto:n...@gmx.ch>> wrote:
>> 
>> 
>> Hi,
>> 
>> When a call goes to a group of snom phones and is answered by one phone, all 
>> the other phones display a missed call notification.
>> 
>> According to the SNOM FAQ this can be stopped by way of the following 
>> changes to SIP protocols:
>> 
>> Has anyone idea how to fix that
>> 
>> Regards 
>> Nedi
>> ___
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>> <mailto:Astlinux-users@lists.sourceforge.net>
>> https://lists.sourceforge.net/lists/listinfo/astlinux-users 
>> <https://lists.sourceforge.net/lists/listinfo/astlinux-users>
>> 
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>> 
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>> pay...@krisk.org.
> 
> ___
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> <https://lists.sourceforge.net/lists/listinfo/astlinux-users>
> 
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> 
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Re: [Astlinux-users] When a call goes to a group of snom phones and is answered by one phone, all the other phones display a missed call notification

[nedi]

I have another snom phone d385, can I import this?

> Am 15.06.2020 um 22:07 schrieb The Cadillac Kid via Astlinux-users 
> :
> 
> yeah I dont get the missed indicator unless I hangup or the call forwards..  
> the missed call doesnt pop on the screen when I see answered elsewhere..  
> 
> here is the template im using for my phone..  ignore all the $$ entries those 
> are where my config generator puts in variables..  maybe theres a setting in 
> it you can leverage?  or perhaos its a firmware thing?  
> 
> snom's online manuals are in flex..  Vtech acquired snom  and we have worked 
> with vtech for many years but I think our firmware is standard GA..  
> 
> 
> 
>   
> 
>   http://$serverip/snom/snom-dialplan.xml"; />
>  
> 
> F_DIALMODE(not:have_incoming_call) F_BACK 
> F_DEFLECT(not:edit_for_transfer) F_ACCEPT_CALL(not:edit_for_transfer) 
> F_DENY(have_incoming_call) F_REDIAL(not:edit_for_transfer) 
> F_LEFT F_RIGHT F_CONF_ON 
> F_HOLD F_CONFERENCE F_DUAL_AUDIO(not:Conference) F_DELETE_MSG 
> 
> snomD785
> keyevent F_TRANSFER
> blind
> on
> attended
> on
> blind
> English
> http://$serverip/snom/{mac}.cfg
> $serverip
> innAcloud
> 8.8.8.8
> -18000
> admin
> admin
> 3600 03.02.07 02:00:00 11.01.07 02:00:00
> USA-5
> perm="">http://$serverip/snom/getlights.php?user=$extension
> 123456
> 
> USA
> 4
> off
> Ringer2
> auto_update
> normal
>  perm="">http://$serverip/snom/snomD785firmware.xml
> 
> PhoneHasCallInStateRinging 
> PhoneHasCallInStateHolding DateOngoing DateReminding 
> 
> off
> speed $voicemail_direct
> keyevent F_DND
> keyevent F_NONE
> keyevent F_NONE
> keyevent F_NONE
> keyevent F_NONE
> off
> off
> 
> 
> off
> 2010.12-1-gd311851f1
> on
>  perm="RW">CallForPickupAvailable:/1
> 
> 
> off
> $extension
> $extension
> $serverip
> $extension
> $extension
> $extension
> off
> 3116c69a-c1b9-46da-8250-00041392D649
> Ringer7
> $serverip
> true
> 
> off
> false
>  perm="">off
>  perm="">pcmu,telephone-event
> off
> off
> off
> off
> off
> off
> off
> off
> 
> 
> off
> 
> 
>  default_text="$name" perm="">$snomD785-SIP_button_1_type 
> $snomD785-SIP_button_1_value
>  default_text="$name" perm="">$snomD785-SIP_button_2_type 
> $snomD785-SIP_button_2_value
>  default_text="$name" perm="">$snomD785-SIP_button_3_type 
> $snomD785-SIP_button_3_value
>  default_text="$name" perm="">$snomD785-SIP_button_4_type 
> $snomD785-SIP_button_4_value
>  default_text="$name" perm="">$snomD785-SIP_button_5_type 
> $snomD785-SIP_button_5_value
>  default_text="$name" perm="">$snomD785-SIP_button_6_type 
> $snomD785-SIP_button_6_value
>  default_text="$name" perm="">$snomD785-SIP_button_7_type 
> $snomD785-SIP_button_7_value
>  default_text="$name" perm="">$snomD785-SIP_button_8_type 
> $snomD785-SIP_button_8_value
>  default_text="$name" perm="">$snomD785-SIP_button_9_type 
> $snomD785-SIP_button_9_value
>  default_text="$name" perm="">$snomD785-SIP_button_10_type 
> $snomD785-SIP_button_10_value
>  default_text="$name" perm="">$snomD785-SIP_button_11_type 
> $snomD785-SIP_button_11_value
>  default_text="$name" perm="">$snomD785-SIP_button_12_type 
> $snomD785-SIP_button_12_value
>  default_text="$name" perm="">$snomD785-SIP_button_13_type 
> $snomD785-SIP_button_13_value
>  default_text="$name" perm="">$snomD785-SIP_button_14_type 
> $snomD785-SIP_button_14_value
>  default_text="$name" perm="">$snomD785-SIP_button_15_type 
> $snomD785-SIP_button_15_value
>  default_text="$name" perm="">$snomD785-SIP_button_16_type 
> $snomD785-SIP_button_16_value
>  default_text="$name" perm="">$snomD785-SIP_button_17_type 
> $snomD785-SIP_button_17_value
>  default_text="$name" perm="">$snomD785-SIP_button_18_type 
> $snomD785-SIP_button_18_value
>  default_text="$name" perm="">$snomD785-SIP_button_19_type 
> $snomD785-SIP_button_19_value
>  default_text="$name" perm="">$snomD785-SIP_button_20_type 
> $snomD785-SIP_but

Re: [Astlinux-users] When a call goes to a group of snom phones and is answered by one phone, all the other phones display a missed call notification

[nedi]

The phone showing message  "call completed elsewhere" on the screen  coming but 
 missed call is still there on all phones.
Nedi

> Am 15.06.2020 um 21:45 schrieb The Cadillac Kid via Astlinux-users 
> :
> 
> this was a PJ issue wasnt it? im still using Chan_sip and my snoms dont do it 
> if I answer at another phone, they actually show call completed elsewhere on 
> the screen...  im using 10.1.49.X firmware on my snoms..  
> I dont remember having to set anything different in the config to make it 
> work.
> 
> 
> On Monday, June 15, 2020, 3:31:16 PM EDT, nedi  wrote:
> 
> 
> Hi,
> 
> When a call goes to a group of snom phones and is answered by one phone, all 
> the other phones display a missed call notification.
> 
> According to the SNOM FAQ this can be stopped by way of the following changes 
> to SIP protocols:
> 
> Has anyone idea how to fix that
> 
> Regards 
> Nedi
> ___
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> <mailto:Astlinux-users@lists.sourceforge.net>
> https://lists.sourceforge.net/lists/listinfo/astlinux-users 
> <https://lists.sourceforge.net/lists/listinfo/astlinux-users>
> 
> Donations to support AstLinux are graciously accepted via PayPal to 
> pay...@krisk.org. 
> <mailto:pay...@krisk.org.>___
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> Astlinux-users@lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to 
> pay...@krisk.org.

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[Astlinux-users] When a call goes to a group of snom phones and is answered by one phone, all the other phones display a missed call notification

[nedi]

Hi,

When a call goes to a group of snom phones and is answered by one phone, all 
the other phones display a missed call notification.

According to the SNOM FAQ this can be stopped by way of the following changes 
to SIP protocols:

Has anyone idea how to fix that

Regards 
Nedi___
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Re: [Astlinux-users] APU2 Install hang

[nedi]

Thanks

> Am 08.05.2020 um 00:48 schrieb Michael Keuter :
> 
> 
> 
>> Am 08.05.2020 um 00:40 schrieb nedi :
>> 
>> Michael, 
>> Is there a working download  link for German Asterisk Sounds Packages  for 
>> Astlinux or I must copy my old files to the new installed PBX.
>> Nedi
> 
> Yes, setup this URL in the AstLinux Prefs as "Sounds Pkg URL":
> http://mirror.astlinux-project.org/asterisk-sounds
> 
> Then choose "core" "german" and "alaw" (all other codecs might not work).
> 
>> 
>>> Am 07.05.2020 um 23:34 schrieb Michael Keuter :
>>> 
>>> 
>>> 
>>>> Am 07.05.2020 um 23:17 schrieb Michael Knill 
>>>> :
>>>> 
>>>> I gave up on screen and ZTerm. Just use Serial on Mac. It just works and 
>>>> supports many USB-Serial convertors.
>>> 
>>> Hi Michael,
>>> I use this as well for years, it is great, but it is not cheap (and now the 
>>> 2.0 upgrade, I dont need those new features).
>>> That's why I suggested it not at this time :-).
>>> 
>>>> Regards
>>>> Michael Knill
>>>> 
>>>> On 8/5/20, 5:42 am, "nedi"  wrote:
>>>> 
>>>> Thanks, 
>>>> I thing that was Mac issue.   With serial app works
>>>> 
>>>> Which version do you recomend me to load 
>>>> genx86_64-serial-1.3.8-asterisk-13.23.1se  [1]| |
>>>>   | |genx86_64-serial-1.3.8-asterisk-13.31.0[2]| |
>>>>   | |genx86_64-serial-1.3.8-asterisk-16.8.0 [3]
>>>> Nedi
>>>> 
>>>>> Am 07.05.2020 um 21:30 schrieb Michael Keuter :
>>>>> 
>>>>> 
>>>>> 
>>>>>> Am 07.05.2020 um 21:13 schrieb Lonnie Abelbeck 
>>>>>> :
>>>>>> 
>>>>>> Nedi,
>>>>>> 
>>>>>> Followup, bottom line, there is something goofy with the macOS 'screen'.
>>>>>> 
>>>>>> I just tried the astlinux-1.3.8-genx86_64-serial.iso installer on the 
>>>>>> APU2.
>>>>>> 
>>>>>> If I use the "Serial" macOS application, it works just fine.
>>>>>> https://www.decisivetactics.com/products/serial/
>>>>>> 
>>>>>> If I use "minicom" on an old MacBook, it works fine.
>>>>>> 
>>>>>> I used another AstLinux box (I have lots of them), same USB/serial 
>>>>>> converter connected to AstLinux, ssh into AstLinux and issue:
>>>>>> --
>>>>>> screen --version
>>>>>> Screen version 4.08.00 (GNU) 05-Feb-20
>>>>>> 
>>>>>> screen /dev/ttyUSB0 115200
>>>>>> --
>>>>>> it works fine.
>>>>>> 
>>>>>> But, if I use macOS's screen:
>>>>>> --
>>>>>> screen --version
>>>>>> Screen version 4.00.03 (FAU) 23-Oct-06
>>>>>> 
>>>>>> screen /dev/tty.usbserial-A102MMV1 115200
>>>>>> ---
>>>>>> as soon as the menus appear, the screen flashes wildly as if I was 
>>>>>> typing characters.
>>>>>> 
>>>>>> Bottom line, don't use the macOS 'screen' with a USB/serial adapter and 
>>>>>> expect it to always work.
>>>>>> 
>>>>>> Lonnie
>>>>> 
>>>>> As a free alternative you can try ZTerm:
>>>>> https://dalverson.com/zterm/
>>>>> 
>>>>> It is very, very old, but still work for me on High Sierra (I guess it 
>>>>> will not work on Catalina anymore).
>>>>> 
>>>>>>> On May 7, 2020, at 1:22 PM, Lonnie Abelbeck  
>>>>>>> wrote:
>>>>>>> 
>>>>>>> Nedi,
>>>>>>> 
>>>>>>> Very weird you can receive data but not send.
>>>>>>> 
>>>>>>> On my MacBook Air, USB to serial and a NULL modem cable to APU2.
>>>>>>> 
>>>>>>> $ ls /dev/tty.*
>>>>>>> /dev/tty.Bluetooth-Incoming-Port/dev/tty.usbserial-A102MMV1
>>>>>>> 
>>>>>>> $ screen /dev/tty.usbserial-A102MMV1 115200
>>>>>>> 
>>>&

Re: [Astlinux-users] APU2 Install hang

[nedi]

Michael, 
Is there a working download  link for German Asterisk Sounds Packages  for 
Astlinux or I must copy my old files to the new installed PBX.
Nedi

> Am 07.05.2020 um 23:34 schrieb Michael Keuter :
> 
> 
> 
>> Am 07.05.2020 um 23:17 schrieb Michael Knill 
>> :
>> 
>> I gave up on screen and ZTerm. Just use Serial on Mac. It just works and 
>> supports many USB-Serial convertors.
> 
> Hi Michael,
> I use this as well for years, it is great, but it is not cheap (and now the 
> 2.0 upgrade, I dont need those new features).
> That's why I suggested it not at this time :-).
> 
>> Regards
>> Michael Knill
>> 
>> On 8/5/20, 5:42 am, "nedi"  wrote:
>> 
>>   Thanks, 
>>   I thing that was Mac issue.   With serial app works
>> 
>>   Which version do you recomend me to load 
>>   genx86_64-serial-1.3.8-asterisk-13.23.1se  [1]| |
>>     | |genx86_64-serial-1.3.8-asterisk-13.31.0[2]| |
>> | |genx86_64-serial-1.3.8-asterisk-16.8.0 [3]
>>   Nedi
>> 
>>> Am 07.05.2020 um 21:30 schrieb Michael Keuter :
>>> 
>>> 
>>> 
>>>> Am 07.05.2020 um 21:13 schrieb Lonnie Abelbeck :
>>>> 
>>>> Nedi,
>>>> 
>>>> Followup, bottom line, there is something goofy with the macOS 'screen'.
>>>> 
>>>> I just tried the astlinux-1.3.8-genx86_64-serial.iso installer on the APU2.
>>>> 
>>>> If I use the "Serial" macOS application, it works just fine.
>>>> https://www.decisivetactics.com/products/serial/
>>>> 
>>>> If I use "minicom" on an old MacBook, it works fine.
>>>> 
>>>> I used another AstLinux box (I have lots of them), same USB/serial 
>>>> converter connected to AstLinux, ssh into AstLinux and issue:
>>>> --
>>>> screen --version
>>>> Screen version 4.08.00 (GNU) 05-Feb-20
>>>> 
>>>> screen /dev/ttyUSB0 115200
>>>> --
>>>> it works fine.
>>>> 
>>>> But, if I use macOS's screen:
>>>> --
>>>> screen --version
>>>> Screen version 4.00.03 (FAU) 23-Oct-06
>>>> 
>>>> screen /dev/tty.usbserial-A102MMV1 115200
>>>> ---
>>>> as soon as the menus appear, the screen flashes wildly as if I was typing 
>>>> characters.
>>>> 
>>>> Bottom line, don't use the macOS 'screen' with a USB/serial adapter and 
>>>> expect it to always work.
>>>> 
>>>> Lonnie
>>> 
>>> As a free alternative you can try ZTerm:
>>> https://dalverson.com/zterm/
>>> 
>>> It is very, very old, but still work for me on High Sierra (I guess it will 
>>> not work on Catalina anymore).
>>> 
>>>>> On May 7, 2020, at 1:22 PM, Lonnie Abelbeck  
>>>>> wrote:
>>>>> 
>>>>> Nedi,
>>>>> 
>>>>> Very weird you can receive data but not send.
>>>>> 
>>>>> On my MacBook Air, USB to serial and a NULL modem cable to APU2.
>>>>> 
>>>>> $ ls /dev/tty.*
>>>>> /dev/tty.Bluetooth-Incoming-Port  /dev/tty.usbserial-A102MMV1
>>>>> 
>>>>> $ screen /dev/tty.usbserial-A102MMV1 115200
>>>>> 
>>>>> ## interact over serial
>>>>> ## exit with Control-a Control-\ (confirm with y)
>>>>> 
>>>>> [screen is terminating]
>>>>> 
>>>>> 
>>>>> Have you used your USB/serial terminal setup successfully before ?
>>>>> 
>>>>> Lonnie
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>>> On May 7, 2020, at 11:43 AM, nedi  wrote:
>>>>>> 
>>>>>> Hi lonnie, 
>>>>>> Yes I have SSD  on slot 3 mSATA 
>>>>>> I Used the Image astlinux-1.3.8-genx86_64-serial.iso 
>>>>>> I made USB boot stick with windows 10  Rufus in DD image mode
>>>>>> 
>>>>>> I see booting I can chose F10  and make memory test  after that or 
>>>>>> directtly  if I don’t press F10 I can  booting directly to the  
>>>>>> astlinux, the installer menu appears  and if the menu appears I can’t 
>>>>>> chose anything.
>>>>>> 
>>>>>> if I press enter or arows nothing h

Re: [Astlinux-users] APU2 Install hang

[nedi]

Thanks, 
I thing that was Mac issue.   With serial app works

Which version do you recomend me to load 
genx86_64-serial-1.3.8-asterisk-13.23.1se  [1]| |
  | |genx86_64-serial-1.3.8-asterisk-13.31.0[2]| |
  | |genx86_64-serial-1.3.8-asterisk-16.8.0 [3]
Nedi

> Am 07.05.2020 um 21:30 schrieb Michael Keuter :
> 
> 
> 
>> Am 07.05.2020 um 21:13 schrieb Lonnie Abelbeck :
>> 
>> Nedi,
>> 
>> Followup, bottom line, there is something goofy with the macOS 'screen'.
>> 
>> I just tried the astlinux-1.3.8-genx86_64-serial.iso installer on the APU2.
>> 
>> If I use the "Serial" macOS application, it works just fine.
>> https://www.decisivetactics.com/products/serial/
>> 
>> If I use "minicom" on an old MacBook, it works fine.
>> 
>> I used another AstLinux box (I have lots of them), same USB/serial converter 
>> connected to AstLinux, ssh into AstLinux and issue:
>> --
>> screen --version
>> Screen version 4.08.00 (GNU) 05-Feb-20
>> 
>> screen /dev/ttyUSB0 115200
>> --
>> it works fine.
>> 
>> But, if I use macOS's screen:
>> --
>> screen --version
>> Screen version 4.00.03 (FAU) 23-Oct-06
>> 
>> screen /dev/tty.usbserial-A102MMV1 115200
>> ---
>> as soon as the menus appear, the screen flashes wildly as if I was typing 
>> characters.
>> 
>> Bottom line, don't use the macOS 'screen' with a USB/serial adapter and 
>> expect it to always work.
>> 
>> Lonnie
> 
> As a free alternative you can try ZTerm:
> https://dalverson.com/zterm/
> 
> It is very, very old, but still work for me on High Sierra (I guess it will 
> not work on Catalina anymore).
> 
>>> On May 7, 2020, at 1:22 PM, Lonnie Abelbeck  
>>> wrote:
>>> 
>>> Nedi,
>>> 
>>> Very weird you can receive data but not send.
>>> 
>>> On my MacBook Air, USB to serial and a NULL modem cable to APU2.
>>> 
>>> $ ls /dev/tty.*
>>> /dev/tty.Bluetooth-Incoming-Port/dev/tty.usbserial-A102MMV1
>>> 
>>> $ screen /dev/tty.usbserial-A102MMV1 115200
>>> 
>>> ## interact over serial
>>> ## exit with Control-a Control-\ (confirm with y)
>>> 
>>> [screen is terminating]
>>> 
>>> 
>>> Have you used your USB/serial terminal setup successfully before ?
>>> 
>>> Lonnie
>>> 
>>> 
>>> 
>>> 
>>>> On May 7, 2020, at 11:43 AM, nedi  wrote:
>>>> 
>>>> Hi lonnie, 
>>>> Yes I have SSD  on slot 3 mSATA 
>>>> I Used the Image astlinux-1.3.8-genx86_64-serial.iso 
>>>> I made USB boot stick with windows 10  Rufus in DD image mode
>>>> 
>>>> I see booting I can chose F10  and make memory test  after that or 
>>>> directtly  if I don’t press F10 I can  booting directly to the  astlinux, 
>>>> the installer menu appears  and if the menu appears I can’t chose anything.
>>>> 
>>>> if I press enter or arows nothing happend
>>>> 
>>>> Nedi
>>>> 
>>>>> Am 07.05.2020 um 18:11 schrieb Lonnie Abelbeck 
>>>>> :
>>>>> 
>>>>> 
>>>>> 
>>>>>> On May 7, 2020, at 10:02 AM, nedi  wrote:
>>>>>> 
>>>>>> Hi, 
>>>>>> I have new apu4d2  Mainboard. Installer hang on Astlinux installer menu, 
>>>>>> I can’t install if I press enter in the Terminal or on physical usb 
>>>>>> keyboard.
>>>>>> I use USB Serial Adapter on my Mac OSX and Baudrate is 115200 
>>>>>> Bios Version is 4.11.0.2
>>>>>> screen /dev/cu.usbserial 115200
>>>>>> 
>>>>>> Regards 
>>>>>> Nedi
>>>>> 
>>>>> Questions ...
>>>>> 
>>>>> Do you have a mSATA SSD installed in the APU2 in the proper slot (labeled 
>>>>> mSATA) ?
>>>>> 
>>>>> Did you use this ISO ...
>>>>> Generic x86-64bit (Serial Console):
>>>>> https://s3.amazonaws.com/mirror.astlinux-project/downloads/iso/astlinux-1.3.8-genx86_64-serial.iso
>>>>> 
>>>>> Did you use something like this to copy the ISO to a USB drive: (macOS)
>>>>> https://abelbeck.com/lonnie/astlinux/info/os-x-cf-write.php
>>>>> 
>>>>> 
>>>>> If "yes" to all the above, what do you mean by "Installer hang" ?
>>>>> 
>>>>> Lonnie
> 
> Michael
> 
> http://www.mksolutions.info
> 
> 
> 
> 
> 
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Re: [Astlinux-users] APU2 Install hang

[nedi]

Hi lonnie, 
Yes I have SSD  on slot 3 mSATA 
I Used the Image astlinux-1.3.8-genx86_64-serial.iso 
I made USB boot stick with windows 10  Rufus in DD image mode

I see booting I can chose F10  and make memory test  after that or directtly  
if I don’t press F10 I can  booting directly to the  astlinux, the installer 
menu appears  and if the menu appears I can’t chose anything.

if I press enter or arows nothing happend

Nedi

> Am 07.05.2020 um 18:11 schrieb Lonnie Abelbeck :
> 
> 
> 
>> On May 7, 2020, at 10:02 AM, nedi  wrote:
>> 
>> Hi, 
>> I have new apu4d2  Mainboard. Installer hang on Astlinux installer menu, I 
>> can’t install if I press enter in the Terminal or on physical usb keyboard.
>> I use USB Serial Adapter on my Mac OSX and Baudrate is 115200 
>> Bios Version is 4.11.0.2
>> screen /dev/cu.usbserial 115200
>> 
>> Regards 
>> Nedi
> 
> Questions ...
> 
> Do you have a mSATA SSD installed in the APU2 in the proper slot (labeled 
> mSATA) ?
> 
> Did you use this ISO ...
> Generic x86-64bit (Serial Console):
> https://s3.amazonaws.com/mirror.astlinux-project/downloads/iso/astlinux-1.3.8-genx86_64-serial.iso
> 
> Did you use something like this to copy the ISO to a USB drive: (macOS)
> https://abelbeck.com/lonnie/astlinux/info/os-x-cf-write.php
> 
> 
> If "yes" to all the above, what do you mean by "Installer hang" ?
> 
> Lonnie
> 
> 
> 
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[Astlinux-users] APU2 Install hang

[nedi]

Hi, 
I have new apu4d2 <https://www.pcengines.ch/apu4d2.htm>  Mainboard. Installer 
hang on Astlinux installer menu, I can’t install if I press enter in the 
Terminal or on physical usb keyboard.
I use USB Serial Adapter on my Mac OSX and Baudrate is 115200 
Bios Version is 4.11.0.2
screen /dev/cu.usbserial 115200


Regards 
Nedi


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Re: [Astlinux-users] Which Alix Mainboard to use with new Astlinux

[nedi]

Hi Michael,
You mean I should buy this one?  
apu4d4  APU.4D4 system board (GX-412TC quad core / 4GB / 4 Intel GigE)
Regards
Nedi
 
> Am 05.05.2020 um 02:24 schrieb Michael Knill 
> :
> 
> You would be surprised the number of times I have needed a 4th port. After 
> EXT, INT and Failover you have none left.
> My standard is 4 port Astlinux box now. I really like APU2-4.
>  
> Regards
> Michael Knill
>  
> From: Michael Keuter mailto:li...@mksolutions.info>>
> Reply to: AstLinux List  <mailto:astlinux-users@lists.sourceforge.net>>
> Date: Tuesday, 5 May 2020 at 7:12 am
> To: AstLinux List  <mailto:astlinux-users@lists.sourceforge.net>>
> Subject: Re: [Astlinux-users] Which Alix Mainboard to use with new Astlinux
>  
> For 10 € more I‘ve rather have a spare one. 
>  
> BTW: Did you look at the commit log the last 2 days :-) ?
>  
> Sent from a mobile device. 
>  
> Michael Keuter
> 
> 
>> Am 04.05.2020 um 23:08 schrieb David Kerr > <mailto:da...@kerr.net>>:
>> 
>> Michael,
>>   What are the good uses for a 4th NIC?  I suppose if you wanted two 
>> completely isolated internal LAN networks without using VLAN it could be 
>> useful, would require separate h/w switch and wiring of course.  Any other 
>> good uses?
>>  
>> David
>>  
>> On Mon, May 4, 2020 at 4:21 PM Michael Keuter > <mailto:li...@mksolutions.info>> wrote:
>>> 
>>> 
>>> > Am 04.05.2020 um 22:18 schrieb Michael Keuter >> > <mailto:li...@mksolutions.info>>:
>>> > 
>>> > 
>>> > 
>>> >> Am 04.05.2020 um 22:10 schrieb Nedeljko Grgic >> >> <mailto:n...@gmx.ch>>:
>>> >> 
>>> >> Sorry i made mistake ,  I thought all pcengines boards are alix 
>>> >> mainboards. 
>>> >> apu2e4 will be good for some time? 
>>> > 
>>> > Yes.
>>> 
>>> BTW: The 4 LAN version APU4E4 is only 10 € more.
>>> 
>>> >> Beste Grüsse
>>> >> Nedeljko Grgić
>>> > 
>>> > Ebenfalls Grüße aus Hamburg :-).
>>> > 
>>> >> Am 4. Mai 2020, um 21:59, Michael Keuter >> >> <mailto:li...@mksolutions.info>> schrieb:
>>> >> 
>>> >> 
>>> >> Am 04.05.2020 um 16:43 schrieb David Kerr >> >> <mailto:da...@kerr.net>>:
>>> >> 
>>> >> Nedi,
>>> >>   The ALIX boards are kind of old.  PC Engines recommends their 
>>> >> APU2/APU4 line and I would too unless you have a large installed base of 
>>> >> Alix and really must keep the same for new deployments (though looking 
>>> >> at PC Engines web site you might not even be able to buy ALIX any more). 
>>> >>  Any of the APU2/4's with a minimum of 3 NIC ports would be fine.  You 
>>> >> need three to allow for WAN, LAN and WAN failover.  4th is not necessary 
>>> >> though I suppose there are some niche applications.
>>> >> 
>>> >> David
>>> >> 
>>> >> On Mon, May 4, 2020 at 9:04 AM nedi mailto:n...@gmx.ch>> 
>>> >> wrote:
>>> >> HI, which  Alix mainboard do you recommend me to use with newest 
>>> >> Astlinux, I would like to upgrade my pix to newest Astlinux. And Is 
>>> >> there a possibility to use Hylfax on Astlinux to end fax from windows 
>>> >> through Astlinux
>>> >> Regards
>>> >> Nedi
>>> >> 
>>> >> And BTW with the next AstLinux version 1.3.10 32-bit boards are not 
>>> >> supported anymore.
>>> >> So please don't buy a new ALIX board. Take a look a our Wiki:
>>> >> 
>>> >> https://doc.astlinux.org/userdoc:documentation#generic_64-bit_x86_64_boards_and_appliances
>>> >>  
>>> >> <https://doc.astlinux.org/userdoc:documentation#generic_64-bit_x86_64_boards_and_appliances>
>>> >> 
>>> >> Michael
>>> > 
>>> > Michael
>>> 
>>> Michael
>>> 
>>> http://www.mksolutions.info <http://www.mksolutions.info/>
>>> 
>>> 
>>> 
>>> 
>>> 
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Re: [Astlinux-users] Which Alix Mainboard to use with new Astlinux

[nedi]

What is with those LTE optimized mainboard, is this usable as wan failover?


> Am 04.05.2020 um 23:07 schrieb David Kerr :
> 
> Michael,
>   What are the good uses for a 4th NIC?  I suppose if you wanted two 
> completely isolated internal LAN networks without using VLAN it could be 
> useful, would require separate h/w switch and wiring of course.  Any other 
> good uses?
> 
> David
> 
> On Mon, May 4, 2020 at 4:21 PM Michael Keuter  <mailto:li...@mksolutions.info>> wrote:
> 
> 
> > Am 04.05.2020 um 22:18 schrieb Michael Keuter  > <mailto:li...@mksolutions.info>>:
> > 
> > 
> > 
> >> Am 04.05.2020 um 22:10 schrieb Nedeljko Grgic  >> <mailto:n...@gmx.ch>>:
> >> 
> >> Sorry i made mistake ,  I thought all pcengines boards are alix 
> >> mainboards. 
> >> apu2e4 will be good for some time? 
> > 
> > Yes.
> 
> BTW: The 4 LAN version APU4E4 is only 10 € more.
> 
> >> Beste Grüsse
> >> Nedeljko Grgić
> > 
> > Ebenfalls Grüße aus Hamburg :-).
> > 
> >> Am 4. Mai 2020, um 21:59, Michael Keuter  >> <mailto:li...@mksolutions.info>> schrieb:
> >> 
> >> 
> >> Am 04.05.2020 um 16:43 schrieb David Kerr  >> <mailto:da...@kerr.net>>:
> >> 
> >> Nedi,
> >>   The ALIX boards are kind of old.  PC Engines recommends their APU2/APU4 
> >> line and I would too unless you have a large installed base of Alix and 
> >> really must keep the same for new deployments (though looking at PC 
> >> Engines web site you might not even be able to buy ALIX any more).  Any of 
> >> the APU2/4's with a minimum of 3 NIC ports would be fine.  You need three 
> >> to allow for WAN, LAN and WAN failover.  4th is not necessary though I 
> >> suppose there are some niche applications.
> >> 
> >> David
> >> 
> >> On Mon, May 4, 2020 at 9:04 AM nedi mailto:n...@gmx.ch>> 
> >> wrote:
> >> HI, which  Alix mainboard do you recommend me to use with newest Astlinux, 
> >> I would like to upgrade my pix to newest Astlinux. And Is there a 
> >> possibility to use Hylfax on Astlinux to end fax from windows through 
> >> Astlinux
> >> Regards
> >> Nedi
> >> 
> >> And BTW with the next AstLinux version 1.3.10 32-bit boards are not 
> >> supported anymore.
> >> So please don't buy a new ALIX board. Take a look a our Wiki:
> >> 
> >> https://doc.astlinux.org/userdoc:documentation#generic_64-bit_x86_64_boards_and_appliances
> >>  
> >> <https://doc.astlinux.org/userdoc:documentation#generic_64-bit_x86_64_boards_and_appliances>
> >> 
> >> Michael
> > 
> > Michael
> 
> Michael
> 
> http://www.mksolutions.info <http://www.mksolutions.info/>
> 
> 
> 
> 
> 
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[Astlinux-users] Which Alix Mainboard to use with new Astlinux

[nedi]

HI, which  Alix mainboard do you recommend me to use with newest Astlinux, I 
would like to upgrade my pix to newest Astlinux. And Is there a possibility to 
use Hylfax on Astlinux to end fax from windows through Astlinux
Regards
Nedi

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[Astlinux-users] astlinux No channel type registered for 'PJSIP'

[nedi]

Hi
I loged in as agent and in the CLI I can see loged in Agent

But if I try to dialing a number I get this error

astlinux No channel type registered for ‚PJSIP'
astlinux No channel type registered for ‚PJSIP'

Can anyone help ?

regards
 Nedi

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[Astlinux-users] Reverse Caller ID lockup with tel.search.ch and API

[nedi]

HI Can somebody help me with this tel.search script?
Regard
Nedi


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[Astlinux-users] Reverse Caller ID lockup with tel.search.ch and API

[nedi]

Hi 
long time ago , I used  one script to resolve the caller number trough 
tel.search.ch <http://tel.search.ch/>

After update to ssl  this script won’t work and after some time the 
tel.search.ch <http://tel.search.ch/> use api to check the caller id.

And now the php changed from 5.6 to new one 7.3 

Can anyone help me to fix this script again.

I use one code in Astlinux  ( I have a old one AstLinux Release:
astlinux-1.2.6.1 i586 - Asterisk 1.8.32.3   Runnix Release: runnix-0.4-7671
GUI Version:1.8.40)

Last working php code and snoopy on my website to check the caller  was :

https://tel.search.ch/?was=".$number;
include "Snoopy.class.php";
$snoopy = new Snoopy;
$snoopy->fetch("$url");
$GrabStart = '';
$GrabEnd = '';
$GrabData = preg_match("/$grabStart(.*?)$grabEnd/i", $snoopy->results, 
$output1[1]);

echo $name[0];
?>


And now I get the API Key and must rewrite those script to be compatible with 
php 7 and  ssl  on my website I use https


The instruction is to use api key with this link to get the number 
https://tel.search.ch/api/?was=john+meier&key=c1e6a4c666c0a2ce9e38a69be7c6a

I tried to change only this url $url = 
"https://tel.search.ch/api/?was=".$number&key=c1e6a4c666c0a2ce9e38a69be7c6a;

But that dosn’t work.

Regards 
Nedi

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[Astlinux-users] WARNING[27685]: pbx_dundi.c:4652 in set_config: Unable to look up host 'mypbx'

[nedi]

Hi
I get this message on my log, 
local0.err asterisk[22023]: ERROR[22023]: netsock2.c:269 in 
ast_sockaddr_resolve: getaddrinfo(„mypbx", "(null)", ...): Name or service not 
known
WARNING[27685]: pbx_dundi.c:4652 in set_config: Unable to look up host ‚mypbx'
Can I ignore this or should I change this in my network settings, as I wrote 
last Time I have issue with "rejected „   after internet provider make firmware 
update and my modem make reboot.

I have in my network settings under domain and host „mypbx“ and local domain 
ist activated

To avoid "rejected" I testing now this script,  can anyone with more experience 
look at this script .Can I load it every one minute for long time without 
afraid to damage CF Card or overload log of my pbx.
Thanks.

Script:

#!/bin/bash -x

DIR=/tmp/watchdog

# Create dir if it doesn't exist
if [ ! -d $DIR ]; then
  mkdir $DIR
fi

cd $DIR

# Save current registration times
/usr/sbin/asterisk -rx "sip show registry" | grep "Registered" | cut -b 92- > 
current

# If last exists, compare current to last
if [ -f last ]; then
  cmp current last

# If they match, restart Asterisk
  if [ $? == 0 ]; then
/etc/init.d/asterisk restart
  fi
fi

rm -f last
mv current last


Regards Nedi





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[Astlinux-users] Astlinux network check and restart if internet down.

[nedi]

Hi, i have trouble with one bx and message rejected after internet going down 
for some time.

I found this script, but I’m not sure if I can use this  on Astlinux.

Can anyone help me to adapt this to Astlinux I would like to check internet and 
if internet connection broke and coming again I would like to restart asterisk 
if no registered.

I think I need  an script in combination with a cron job that check if 
registration fail. If registration fail then make sip reload. 
And if Internet going down for some time and up after some time, I would like 
check that and if internet UP an the registration fail or rejected. I would 
like reboot PBX or only Asterisk… 

could be something like thiis what I found:

asterisk -rx " sip show registry" | grep -w Unregistered && asterisk -rx " sip 
reload"
I found this to on the net. Can someone help me to combine those two cron.


asterisk restart if the net is down and once a 
day just before 7am:
crontab -e:
55 6 * * * /usr/sbin/asterisk -r -x "restart gracefully" >/dev/null 2>&1
5,10,15,20,25,30,35,40,45,50,55 * * * * /root/scripts/check_net

/root/check_net
#!/usr/bin/perl
$net=`/bin/ping -c 02 google.com 2>&1 | /bin/grep -c 'unknown host'`;
if ($net==1) {
print `/bin/date`;
print `/usr/sbin/asterisk -r -x "restart gracefully"`;
}


Regards 
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Re: [Astlinux-users] Issue periodically my Astlinux can't register and show me rejected

[nedi]

Hi Michael, thanks for the Answer, 
I will trying this register_retry_403=yes in sip.conf I hope that will fix my 
issue to.

Best regards
Nedi

> Am 10.05.2018 um 05:41 schrieb Michael Knill 
> :
> 
> Hi Nedi
> 
> This is what I have found:
> 1) If you are registering to a DNS name, if the address changes Asterisk does 
> not know about it unless you are using srvlookup=yes in sip.conf. PS I don't 
> use it as it does a lookup for every call and I don't like this. I instead 
> use dnsmgr.conf to check periodically. Nate that this only affects SIP 
> registration and not SIP OPTIONS (qualify). For this I actually use a Monit 
> script but the cause of this problem is not what you are experiencing.
> 2) Yes using the IP Address solves 1) but if you have a provider with 
> multiple IP Addresses then this is not the best approach.
> 3) I have found this problem only recently and I think it may be what you are 
> experiencing. Providers with clustered SBC's (which yours sounds like), when 
> they fail over for some reason, any registration attempts to the standby SBC 
> responds with a 403 Forbidden. This was proved by my provider with a packet 
> capture. By default Asterisk views a 403 Forbidden as an authentication 
> failure and stops trying to register. If you set register_retry_403=yes in 
> sip.conf it fixes this problem.
> 
> Good luck. 
> 
> Regards
> Michael Knill
> 
> On 10/5/18, 1:24 pm, "Lonnie Abelbeck"  wrote:
> 
>Hi Nedi,
> 
>I appreciate your detailed description and for the record I feel your 
> frustration.
> 
>I seem to recall Michael Knill described a similar situation in the recent 
> past.
> 
>My educated guess is this is a firewall statefull-inspection "stuck-state" 
> somewhere in the network path.  AstLinux's firewall is not the issue 
> otherwise a reboot would solve the problem.
> 
>Long story short, as a test, I would determine a time of day where 
> shutting down asterisk is acceptable (say 2:00 am) and keep asterisk down for 
> 10 minutes or so, something like:
> 
>-- cron at 2:00 am or so --
>service asterisk stop
>sleep 600
>service asterisk init
>--
>(make sure you don't have any other background scripts looking to restart 
> asterisk other than the cron script)
> 
>Possibly 5 minutes is all that is needed, but I would start at 10 minutes. 
>  Most any firewall UDP state should expire in 10 minutes.
> 
>Sadly, this test may take weeks to confirm if it helps.
> 
>If you are already using cron to rewrite the sip.conf this test will be a 
> simple addition by delaying starting up asterisk.
> 
>Lonnie
> 
> 
> 
> 
>> On May 9, 2018, at 6:58 PM, nedi  wrote:
>> 
>> Hi,
>> 
>> I have some Astlinux boxes as local pbx after Modem with standard firewall 
>> inside modems , Astlinux have different Releases,  and since years I have 
>> this issue and can’t fix that:  
>> After a week or month and sometimes longer  the voip number can’t be 
>> registered and Status show me „rejected“ 
>> 
>> Reboot can’t help - Aterisk Reload can’t help - I can’t reproduce that for 
>> testing. 
>> If I have this problem only change the SIP Server to another Sipcall sip 
>> server can help to fix that issues. 
>> 
>> As sample if I was registered to business1.voipgateway.org I must change 
>> registrar  to business2.voipgateway.org and I can save and reload to  
>> register without reboot.
>> 
>> I have tried  instead of registrar name to  use the ip adress but this not 
>> fixed my problem. 
>> I use Sipcall DNS IP’s to in the Astlinux Network setting that don’t help, 
>> I don’t use IPV6.  IPV6 is disabled in the Network Settings.
>> Nat and Firewall in Astlinux are enabled and.
>> 
>> I found  that I have this issue only with the provider Sipcall.ch ( if 
>> registered on different subdomains of  voipgateay.org ) and only if I use 
>> Swisscom as Internet Provider. On others provider on the same astlinux and 
>> at the same time I don’ t have this issues.
>> On the same Astlinux box by some customer I have different provider and 
>> sipcall on the same astlinux box and only the Sipcall have this all other 
>> provider worked onlythe the Sipcall account get rejected. 
>> 
>> At now I use Alix Board and AstLinux Release astlinux-1.2.6.1 i586 - 
>> Asterisk 1.8.32.3, but i had this issues with oter astlinux release to.
>> 
>> Some of my customer changed to internet with fix ip adress and  there are 
>> this issue not so often.before with Dynamic IP Adress i had this issues 
>>

Re: [Astlinux-users] Issue periodically my Astlinux can't register and show me rejected

[nedi]

Hi Lonnie,

Thanks for the answer,
Do you mean that will fix this issue wit the registering? Or is this for 
testing only why мy astlinux not register?

The Michael has the same issue with clustered registrar.. 
My Provider told me that is not provider issue, and I’m confused, I have one 
Gigaset SIP Phone as backup line  registered inside the same network to this 
same number that I used on astlinux ( SIP-Forking) and if  I have this 
registering issue  the gigaset sip phone is still  registered astlinux not. At 
the first I will trying this with register_retry_403=yes and waiting.

I have this issue in my case only with one internet provider swisscom.ch and 
one VOIP Provider Sipcall.ch  If I have  more different  provider on the same 
pbx usually only the one VOIP Provider in my case sipcall(voipgateway.org) have 
the status  „rejected" or *send request" 

Can you please check my script for Ip change.
I have this script running on the pbx but if I checking  my sip.conf I can't 
notice any change with this script.
The script is executable and running but my sip.conf not change as sample if I 
put manually in my sip.conf any ip adress  under general.
I put wrong external IP adress in sip.conf general and run the script  but 
nothing changing. I think there is something wrong or I not understood this.
As sample I put wrong external  IP Adress under general  and runing the script.
externip=94.128.12.12

I have this script under /mnt/kd/checksetexternip.sh


This is the script I’m try ti using to overcome this problem. It should check 
the
external IP and update the sip.conf with the new IP . I Run this script as
cronjob every 5 minutes. But I can’t notice any change I my sip.conf.

#!/bin/bash
# checksetexternip.sh
# Author: John Cahill email at johncahill.net
# Licence: GPL v3
# Description: script that queries checkip.dyndns.com to find the server's
external IP address. Updates asterisk's externip value and does a sip
reload if necessary.
# Last modified 06/02/2012

is_ip(){

input=$1
octet1=$(echo $input | cut -d "." -f1)
octet2=$(echo $input | cut -d "." -f2)
octet3=$(echo $input | cut -d "." -f3)
octet4=$(echo $input | cut -d "." -f4)
stat=1

if [[ $input =~ ^[0-9]{1,3}\.[0-9]{1,3}\.[0-9]{1,3}\.[0-9]{1,3}$ ]] && [
$octet1 -le 255 ] && [ $octet2 -le 255 ] && [ $octet3 -le 255 ] && [
$octet4 -le 255 ];
  then
stat=0
fi

return  $stat

}

EXTERNIP=`wget -qO- http://checkip.dyndns.com | awk '{print $6}'| cut -d"<"
-f1`
is_ip $EXTERNIP
if [ $? -ne 0 ]
then
logger -s "checksetexternip.sh: External IP address invalid
or unavailable, exiting."
exit 1
fi

OLDEXTERNIP=`grep externip /etc/asterisk/sip.conf | cut -d"=" -f2`
if [ "$EXTERNIP" = "$OLDEXTERNIP" ]
then
logger -s "checksetexternip.sh: External IP address is the
same, nothing to do exiting."
exit 0
else
logger -s "checksetexternip.sh: External IP address has
changed, changing /etc/asterisk/sip.conf"
#grep -v "externip" /etc/asterisk/sip.conf >
/etc/asterisk/sip.conf.tmp
#echo "externip=$EXTERNIP" >> /etc/asterisk/sip.conf.tmp
#cp /etc/asterisk/sip.conf.tmp /etc/asterisk/sip.conf
#rm /etc/asterisk/sip.conf.tmp
sed -i -e "s/^externip *=.*/externip=$EXTERNIP/" /etc/asterisk/sip.conf
logger -s "Doing asterisk -rx "sip reload""
asterisk -rx "sip reload"

Regards nedi




> Am 10.05.2018 um 05:23 schrieb Lonnie Abelbeck :
> 
> Hi Nedi,
> 
> I appreciate your detailed description and for the record I feel your 
> frustration.
> 
> I seem to recall Michael Knill described a similar situation in the recent 
> past.
> 
> My educated guess is this is a firewall statefull-inspection "stuck-state" 
> somewhere in the network path.  AstLinux's firewall is not the issue 
> otherwise a reboot would solve the problem.
> 
> Long story short, as a test, I would determine a time of day where shutting 
> down asterisk is acceptable (say 2:00 am) and keep asterisk down for 10 
> minutes or so, something like:
> 
> -- cron at 2:00 am or so --
> service asterisk stop
> sleep 600
> service asterisk init
> --
> (make sure you don't have any other background scripts looking to restart 
> asterisk other than the cron script)
> 
> Possibly 5 minutes is all that is needed, but I would start at 10 minutes.  
> Most any firewall UDP state should expire in 10 minutes.
> 
> Sadly, this test may take weeks to confirm if it helps.
> 
> If you are already using cron to rewrite the sip.conf this test wi

[Astlinux-users] Issue periodically my Astlinux can't register and show me rejected

[nedi]

Hi,

I have some Astlinux boxes as local pbx after Modem with standard firewall 
inside modems , Astlinux have different Releases,  and since years I have this 
issue and can’t fix that:  
After a week or month and sometimes longer  the voip number can’t be registered 
and Status show me „rejected“ 

Reboot can’t help - Aterisk Reload can’t help - I can’t reproduce that for 
testing. 
If I have this problem only change the SIP Server to another Sipcall sip server 
can help to fix that issues. 

As sample if I was registered to business1.voipgateway.org 
<http://business1.voipgateway.org/> I must change registrar  to 
business2.voipgateway.org <http://business2.voipgateway.org/> and I can save 
and reload to  register without reboot.

I have tried  instead of registrar name to  use the ip adress but this not 
fixed my problem. 
I use Sipcall DNS IP’s to in the Astlinux Network setting that don’t help, 
I don’t use IPV6.  IPV6 is disabled in the Network Settings.
Nat and Firewall in Astlinux are enabled and.

I found  that I have this issue only with the provider Sipcall.ch 
<http://sipcall.ch/> ( if registered on different subdomains of  voipgateay.org 
<http://voipgateay.org/> ) and only if I use Swisscom as Internet Provider. On 
others provider on the same astlinux and at the same time I don’ t have this 
issues.
On the same Astlinux box by some customer I have different provider and sipcall 
on the same astlinux box and only the Sipcall have this all other provider 
worked onlythe the Sipcall account get rejected. 

At now I use Alix Board and AstLinux Release astlinux-1.2.6.1 i586 - Asterisk 
1.8.32.3, but i had this issues with oter astlinux release to.

Some of my customer changed to internet with fix ip adress and  there are this 
issue not so often.before with Dynamic IP Adress i had this issues oftener.

The Sipcall support told me that is not the problem on the Sipcall site.  Some 
times the Sipcall told me Astlinux trying to register with old IP Adress and 
sometimes they told me Astlinux don’t try to register for some hours or days 
and they don’t  getting inquiry from pbx to register)

I tried to change some settings in my sip.conf and  there is nothing what 
helped me to resolve the issue.

egisterattempts=0
registertimeout=20
maxexpiry=3600
defaultexpiry=600
qualify=2500
srvlookup=no
nat=yes

I use a cron script if external IP changed and update the sip.conf with the new 
IP for dynamic IP and  again I have this problem, that not helped me great.

The PBX can work for weeks and months without problem but after some time I get 
that again.  

I don’t know have this issue something with Astlinux firewall or with swisscom 
modem or with Sipcall.

I think when, for some reason the registry fail and the state of the registry 
becomes REJECTED, asterisk stops trying to register. Can someone help me 
figuring out why this occur and a way to prevent asterisk from stop sending the 
registration requests. 

Does anybody have an idea about that problem ? How can we configure Asterisk to 
retry after some time in case of rejection ?


Im my sip.conf I have this:


I use a cron script to change the IP Adress in Astlinux  if external ip adress 
was changed and  again I have this problem that  some customer get rejected.

The PBX can work for weeks and months without problem but after some time I get 
that again.  

I don’t know have this issue something with Astlinux firewall or with swisscom 
modem or with Sipcall.

I think when, for some reason the registry fail and the state of the registry 
becomes REJECTED, asterisk stops trying to register. Can someone help me 
figuring out why this occur and a way to prevent asterisk from stop sending the 
registration requests. 

Does anybody have an idea about that problem ? How can we configure Asterisk to 
retry after some time in case of rejection ?


Im my sip.conf I have this:

[general]
useragent=mypbxname
port=5060   
context = from-sip-external; send unknown sip callers to this context
alwaysauthreject=yes
deny = 0.0.0.0/0.0.0.0
permit = 10.0.0.0/255.255.255.0
permit = 10.8.0.6/255.255.255.0
allowguest=no
disallow=all 
allow=alaw
allow=ulaw
language=de
registerattempts=0
registertimeout=20
maxexpiry=3600
defaultexpiry=600
qualify=2500
notifycid=yes
srvlookup=no
nat=yes
allow subscribe = yes
subscribecontext = hints
trustrpid=yes
sendrpid=yes
;t38pt_udptl = yes
faxdetect=yes

register => 4171511:2345678@212.117.203.35 
<mailto:2345678@212.117.203.35>/4171511

[071511]
type=peer
username=4171511
secret=2345678
host=212.117.203.35
;fromuser=4171511
fromdomain=212.117.203.35
directmedia=no
insecure=port,invite
disallow=all 
allow=alaw
allow=ulaw
context=incoming212.117.203.35
dtmfmode=info
trustrpid=yes
sendrpid=pai



Best regards
Nedi

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Re: [Astlinux-users] Is there a way durring Incomming call to dial internal number and play audio file to the caller at the same time.

[Nedi]

David i have resolved this with m(Willkommen)  in dialplan ,  yesterday this 
not working because i had in musiconhold.conf setting random=yes  

exten =>10,1,Answer()
exten =>10,n,Set(CHANNEL(language)=de)
exten =>10,n,Set(destination=${CUT(EXTEN,"*",3)})
exten =>10,n,Set(source=${CUT(EXTEN,"*",2)})
exten =>10,n,Dial(SIP/17&SIP/10,35,m(Willkommen)tTwWxXr)

I changed in musiconhold.conf  to radnom =no and works
[default]
mode=files
directory=/var/lib/asterisk/moh/default
random=no   ; Play the files in a random order


Thanks
regards nedi
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[Astlinux-users] Is there a way durring Incomming call to dial internal number and play audio file to the caller at the same time.

[Nedi]

Hi David, 
thanks

I need Dial and Playback at the same time.  
I dian two internal number 10  is my deskphone and 17 is my voip gsm gateway ..
The problem is the gsm gateway, trough  GSM Gateway i get   the call but with 
delay and I would like to get incomin calls trough  gsm gateway at the same 
time in the time the caller hearing the message Playback(Willkommen).

 At the moment I have this in my extension,conf 
exten =>10,1,Answer()
exten =>10,n,Set(CHANNEL(language)=de)
exten =>10,n,Set(destination=${CUT(EXTEN,"*",3)})
exten =>10,n,Set(source=${CUT(EXTEN,"*",2)})
exten =>10,n,Playback(Willkommen)
exten =>10,n,Dial(SIP/17&SIP/10,35,mtTwWxXr)

Regards
nedi
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[Astlinux-users] Is there a way durring Incomming call to dial internal number and play audio file to the caller at the same time.

[Nedi]

Hi  List
Is there a way durring  Incomming call to  dial internal  number and play audio 
file to the caller  at the same time. 
best regards 
nedi
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[Astlinux-users] Fax recive don't work anymore

[Nedi]

Hi 
I use newest Astlinux-1.2.6.1 - Asterisk 1.8.32.3  and Runix  on my alix board .
Since some time i noticed Fax recive don’t work anymore. I nothing changend 
since 2 years, I only made some astlinux update.
what can I doo to fix this.

in my extensions.conf  i have this:

exten =>14,1,System(/mnt/kd/email_in_FAX14.sh "${CALLERID(name)}" 
"${CALLERID(num)}")
exten =>14,n,Gosub(fax14-rx,s,1)
exten =>14,n,Hangup()

[fax14-rx]
exten = s,1,NoOp(Receive FAX)
exten = 
s,n,Set(FAXFILE=/tmp/usb/fax14/fax~${CALLERID(NUM)}~n...@gmx.ch~${FILTER(0123456789,${UNIQUEID})})
exten = s,n,Set(LOCALHEADERINFO=My Name)
exten = s,n,Set(LOCALSTATIONID=My Name)
exten = s,n,ReceiveFAX(${FAXFILE}.tiff)  ;Asterisk 1.8
exten = s,n,Log(NOTICE,New FAX: ${FAXPAGES} page(s) from 
${CALLERID(NUM)}(${REMOTESTATIONID}) to ${CALLERID(dnid)})
exten = s,n,Hangup()
exten = s,n,Return()

to convert fax to pdf and send I use this script:
#!/bin/bash
#
background () {
  while true;
do
for f in /tmp/usb/fax14/*.tiff
do
  if [ -f $f ]
  then
fuser -s $f
if [ $? -ne 0 ]
then
  tiff2pdf -o "${f%\.*}".pdf "$f"
  mv -f "$f" /tmp/usb/fax14/lastFaxBackup14
 fi
  fi
done
for f in /tmp/usb/fax14/*.pdf
do
  if [ -f $f ]
  then
cidemail="${f%~*}"
cidemail="${cidemail#*~}"
email="${cidemail#*~}"
cid="${cidemail%~*}"
shortf="${f##*~}"
mv -f "$f" "/tmp/usb/fax14/fax-$shortf"
echo "Subject: New FAX from $cid
From: n...@gmx.ch
To: $email" | \
mime-pack "New FAX received from $cid" "/tmp/usb/fax14/fax-$shortf" 
"application/pdf" | \
sendmail -t
rm -f "/tmp/usb/fax14/fax-$shortf"
  fi
done
sleep 30
  done
}
background&
echo $! > /tmp/usb/fax14/check_fax14.pid




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[Astlinux-users] make Call from Outlook ans Phonebook Sync with Exchange

[nedi]

Hi, I seek a solution with astlinux and Snom Phone be able  to make phone calls 
from Outlook and sync Snom Phonebook with Exchange or Google account. 

Has anyone Idea or if someone know how to do that. 

I found some client like PhoneSuite CTI Client but I'm not sure if that works 
in the praxis with astlinux.

Regards Nedi

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Re: [Astlinux-users] how to block all inbound calls and pass only calls from switzerland..

[nedi]

Thanks all,
James, how do you think to make this with session progress play a rejection 
Message so I don't get billed on the blacklisted numbers? 

at the moment  if someone calling this Number from wrong area code, the Message 
"Number not Exist" is played.

I included all Swiss area codes and all others caller  from wrong area code get 
the Message "Number not Exist" is played. That works. But I need a idea how to 
make better solution for  rejection message for caller that don't have Swiss 
area code .
And I don't tested Private callers and callers  where no Caller ID information 
is provided.

exten => 90434279234/_021XXX,1,Answer()
exten => 90434279234/_022XXX,1,Answer()
exten => 90434279234/_024XXX,1,Answer()
exten => 90434279234/_026XXX,1,Answer()
exten => 90434279234/_027XXX,1,Answer()
exten => 90434279234/_031XXX,1,Answer()
exten => 90434279234/_032XXX,1,Answer()
exten => 90434279234/_033XXX,1,Answer()
exten => 90434279234/_034XXX,1,Answer()
exten => 90434279234/_041XXX,1,Answer()
exten => 90434279234/_043XXX,1,Answer()
exten => 90434279234/_044XXX,1,Answer()
exten => 90434279234/_051XXX,1,Answer()
exten => 90434279234/_052XXX,1,Answer()
exten => 90434279234/_055XXX,1,Answer()
exten => 90434279234/_056XXX,1,Answer()
exten => 90434279234/_058XXX,1,Answer()
exten => 90434279234/_061XXX,1,Answer()
exten => 90434279234/_071XXX,1,Answer()
exten => 90434279234/_076XXX,1,Answer()
exten => 90434279234/_077XXX,1,Answer()
exten => 90434279234/_078XXX,1,Answer()
exten => 90434279234/_079XXX,1,Answer()
exten => 90434279234/_081XXX,1,Answer()
exten => 90434279234,n,Goto(interno,13,1)

Regards  Nedi

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Re: [Astlinux-users] CallerID after call Transfer

[nedi]

Hi all
I found after some test with the Snom D-375 phone Caller ID after call Trasfer 
works with this phone. 
I put only in my  sip.conf under default and I in every extensions  those two 
lines.

trustrpid=yes
sendrpid=yes

Regards Nedi


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Re: [Astlinux-users] every time wan ip changed ... astlinux accounts can't register to provider.

[nedi]

Thanks to all 
I will try dyndns and the script.

Nedi

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[Astlinux-users] how to block all inbound calls and pass only calls from switzerland..

[nedi]

Hi All,

I have a customer with 0800 Number  and  I would like to block all inbound 
calls to this number. Only Number from switzerland should pass the filter.

I can make blacklist by provider but i must put every area code  number manual  
to blacklist 

regards
Nedi

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[Astlinux-users] every time wan ip changed ... astlinux accounts can't register to provider.

[nedi]

Hi all,
i have some astlinux installations and one big problem every time wan ip 
changed ... astlinux accounts can't register to provider.

Reboot of Astlinux (alix board) can't solve this problem
sip reload or asterisk reload can't fix that problem.
shutdown and wait for some time  and after that boot again can't solve this 
problem.
I tryd to put the DNS Server form Provider and I tryed to put the IP adress 
instead of sip server name.


I found  my provider sipcall.ch  have some other sip server if I change the 
server and connect to other server all number are registered after that I can 
change again to the old server and all works.


The Provider Sipcall told me astlinux trying to register and use old IP adress 
not the new one. 


in 2010 wrote dominko  how to fix i think the same problem but i can't find 
that script for  /etc/ppp/ip-up and /etc/ppp/ip-down.

Has anyone the same problem and is there a way to fix that ?

best regards


Nedi

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[Astlinux-users] transfer and callerid problem

[nedi]

Hi, has someone fixed this problem with asterisk and caller id by transfering 
call to internal number?
I can make the transfer only with asterisks *2 an the caller id trasfer to but 
with snom phone and hold that dos'nt work.

regards Nedi

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[Astlinux-users] CallerID after call Transfer

[nedi]

Hi 
I use snom phone,and new astlinux. is there a way to fix CallerID after call 
transfer.

Phone B receives a call from Phone A
Phone B answers
Phone B makes an attended transfer to Phone C using one of the Snom attended 
transfer methods 
Phone C still sees Phone B as Caller ID (instead of Phone A)

folowed snom wiki and put sendrpid=pai in sip.conf on every peer and sip 
account and I put in general trustrpid=yes
sendrpid=yes

regards
Nedi

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Re: [Astlinux-users] Astlinux and Masquerade bug ?

[nedi]

Dear Lonnie, 
the provider Sipcall.ch do not use or recommend stun server, the provider 
dosn't have stun server.

I don’t understand how is possible that  the provider get old IP address after 
IP address changed.

at first I would check the firewall and check SIP ALG..

Thanks

regards Nedi

 
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[Astlinux-users] Astlinux and Masquerade bug ?

[nedi]

HI 
sometime astlinux can’t connect to provider sipcall.ch,  every time if internet 
loss connection and the router get a new IP address.
I found  with the provider the asterisk send to the provider request with the 
old ip adrees instead with the new ip adress.

Ist there a way to fix that? Is this a Astlinux Firewall bug or Firewall bug 
from Router ?
Or has this something with Masquerade?

Should I disable firewall on astlinux  if I have firewall in my router before 
astlinux ? 

I forwarded only some ports on Astlinux firewall.

Is to difficult to reproduce it  i don't get  new IP if I reset my Router.

This issue I have only with Sipcall.ch  to all other provider I can connect 
without problems.

regards
nedi
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[Astlinux-users] During power failure my PBX reboot and one Provider is offline all other Provider are online

[nedi]

Hi
I notice sometime During power failure my PBX reboot and one provider is 
offline un -registered all other providers are online one (provider Sipcall.ch 
) i use newest AstLinux .

has anyone problems like this and have someone idea how to solve this.

I think everytime if the Line not logged off Properly (by power loss ) the line 
is blocked . If I make reboot through web interface the number going after 
reboot online.

Can I put some code in asterisk to awake if all provider line are online and 
registered and if some provider- going offline for 30 second make a reboot .

best regards
Nedi
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[Astlinux-users] Is there a way to use wildcard or put the range in Astlinux blacklist

[nedi]

Hi 
has someone try to  use wildcard or put a range  in Astlinux blacklist.. or 
only a part of Number as sample041 588
I would like to block full range of Numbers  which include as sample 041 588,   
052 588,   061 588…

best regards



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[Astlinux-users] Astlinux as IPSEC Client I would like connect to Fritzbox VPN

[nedi]

Hello,

has anyone tried to configure Astlinux to be a IPSEC VPN client I would Like to 
connect to Fritzbox?

I have working fritzbox VPN with:
Fritzbox serveradress as IP 
Accountname:
Password:
and shared secret key
and Group Name is the Accountname (Username)

Best Regards,

Nedi

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Re: [Astlinux-users] DTMF from GSM VOIP Gateway

[nedi]

Thanks All for the Answer, but DTMF won't work anymore without fail, 
I use only 711 Codec and I tried all DTMF mode and all Settings in Gateway to 
change without success.
I think the Provider is the Problem the provider changins something  or the 
Gateway is damaged 
sometime working and most of time not working.

Best regards 
Nedi

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[Astlinux-users] DTMF from GSM VOIP Gateway

[nedi]

Thanks Benjamin, 
I tried all DTMF mode without success.
I have this problem  since one or two months ago.
Can anyone help hof to restore old firmware younger than 1.1.6  without new 
install?
I there  a way to make restore to old firmware trough web interface?

Best Regards

Nedi

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[Astlinux-users] DTMF from GSM VOIP Gateway

[nedi]

Hi,
I have problem since one months ago, I can't remember exactly the datum I use 
this way to make phone not so frequently. 
After retour from Holiday I can't make phone through GSM Gateway SC-385. I 
don't change anything 
only I made one or two update to newest Astlinux. 1.1.7

If I connect trough VPN and Use Sip client there  is no problem DTMF are 
recognized.
I tried another Gateway and Another provider SIM Card. So I think the problem 
coming from Asterisk???
I use since about 2 years dtmfmode=info in my Astlinux, yesterday i changed for 
testing on GSM Gateway and in my sip.conf to Inbound and rfc2833 but without 
success.

I have now Astlinux. 1.1.7

I can see this message if I debugging this extension  

on-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), 
combined - 0x0 (nothing)

has anyone idea what can be the problem 

is there was new in asterisk and DTMF on last 2 Updates?

best regards
nedi





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[Astlinux-users] Is there a way to Block Number range with Blacklist

[nedi]

Thanks Michael and James,

I try to put this code in my extension.com under incoming to block a range to 
blocking call centers but all calls are blocked through Blacklist-Zap

I have all incoming calls in international format and would like to check first 
11 digits and use loonies sample code.

;exten => nedi_ch,1,GotoIf($[${DB_EXISTS(blacklist/${CALLERID(num)})} = 0]?200) 
; blacklist
exten => nedi_ch,1,GotoIf(${DB_EXISTS(blacklist/${CALLERID(num):0:11})} = 
0]?200) ; blacklist
exten => nedi_ch,n,GotoIf($["${DB_RESULT}" = "0"]?110)
exten => nedi_ch,n,GotoIf($["${DB_RESULT}" = "2"]?120)
exten => nedi_ch,n,Goto(Blacklist-Zap,s,1) ; ì1″ TN in blacklist 
database, answer and Zapateller
exten => nedi_ch,110,Goto(Blacklist-Hangup,s,1) ; ì0″ TN in blacklist, 
Hangup
exten => nedi_ch,120,Goto(Blacklist-VM,s,1)  ; ì2″ TN in blacklist, 
direct to voicemail
;exten => nedi_ch,200,Dial(local/10@10)
exten => nedi_ch,200,Goto(interno,10,1)

I tried first this code as static entry  
exten => nedi_ch,n,GotoIf(${DB_EXISTS(blacklist/${CALLERID(num):0:11})}?200) ; 
blacklist
and second time I tried edit this to exten => 
nedi_ch,1,GotoIf(${DB_EXISTS(blacklist/${CALLERID(num):0:11})} = 0]?200) ; 
blacklist
both of them doesn't work there is something wrong ${CALLERID(num):0:11})} = 
0]?200) ; blacklist
has anyone a idea how to edit loonnies script and check only first 11 digits of 
CID


Best Regards 
Nedi

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[Astlinux-users] Is there a way to Block Number range with Blacklist

[nedi]

Hi 
Is there a way to block a number Range with Astlinux Blacklist? 
Best Regards
Nedi

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Re: [Astlinux-users] one way voice problem if the transfer message enabled and parallel calls to internal ext.

[nedi]

Thanks, I have the newest Astlinux Version I would try "directmedia=no" and 
disable Nat

Best Regards Nedi

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[Astlinux-users] one way voice problem if the transfer message enabled and parallel calls to internal ext.

[nedi]

Hi, 
I have problem with  my pbx and  Gigaset C610 IP Phone on internal number 71.

If the incoming call coming to the phone Number 41712345670 and the call ist 
forwarded to internal numbers as parallel call included internal number 71  
after 16 second there is the problem with only one way voice, the caller can’t 
hear the person on the extensions 71. But If the call coming direct to this 
extension 41712345671 there is no audio problem both person can hear each 
other. 

I have testet this with a snom 300 Phone and there is no one way audio problem. 
The Gigaset phone don't have gsm codes and on pbx i have disabled gsm codec
I have tested if I disable transfer message, after disable transfer message 
there is no problem with one way voice 
;exten =>70,n,Playback(transfer)  

I have tried transfer message with ulaw, alaw and gsm codec all the same 
problem i can hear the message but after transfer there is only one way audio.

has anyone ideaa what can I try to fix that?

In my SIP.Conf i have

[general]
useragent=MYPBX2SG
port=5060   
context = from-sip-external; send unknown sip callers to this context
alwaysauthreject=yes
deny = 0.0.0.0/0.0.0.0
permit = 192.168.1.0/255.255.255.0
allowguest=no
language=de
disallow=all 
allow=ulaw
allow=alaw
maxexpiry=120
defaultexpiry=50
qualify=2000
srvlookup=yes
nat=yes
allowsubscribe = yes
subscribecontext = hints

[41712345670]
type=peer
username=41712345670
secret=xx
fromuser=41712345670
host=siphost.ch
fromdomain=sipdomain.ch
canreinvite=no
insecure=port,invite
disallow=all
allow=alaw
allow=ulaw
allow=gsm
context=incominmygsipprovider
dtmfmode=info

[41712345671]
type=peer
username=41712345671
secret=xx
fromuser=41712345671
host=siphost.ch
fromdomain=sipdomain.ch
canreinvite=no
insecure=port,invite
disallow=all
allow=alaw
allow=ulaw
allow=gsm
context=incominmygsipprovider
dtmfmode=info

[70]
type=friend
username=70
secret=x
callerid="70" <70>
host=dynamic
mailbox=70@default
dtmfmode=info
canreinvite=no
insecure=port,invite
context=70
disallow=all 
allow=ulaw
allow=alaw
callgroup=2
pickupgroup=2
notifyringing=yes
callcounter=yes
limitonpeers = yes

[71]
type=friend
username=71
secret=x
callerid="71" <71>
host=dynamic
mailbox=70@default
dtmfmode=info
canreinvite=no
insecure=port,invite
context=71
disallow=all 
allow=ulaw
allow=alaw
callgroup=2
pickupgroup=2
notifyringing=yes
callcounter=yes
limitonpeers = yes

in my Extensions.conf I have 

[incomingmysipprovider]
exten => 41712345670,1,Dial(local/70@70)
exten => 41712345671,1,Dial(local/71@71)

exten =>70,1,Dial(SIP/70,16,r)
exten =>70,n,Playback(transfer)
exten 
=>70,n,Dial(SIP/71&SIP/72&SIP/73&SIP/74&SIP/75&SIP/76&SIP/77&SIP/78&SIP/79,13,r)
exten =>70,n,Playback(vm-nobodyavail) 
exten =>70,n,Voicemail(70)
exten =>70,n,Hangup

exten =>71,1,Dial(SIP/71,18,r)
exten =>71,n,Playback(transfer)
exten =>71,n,Dial(SIP/70,13,r)
exten =>71,n,Answer
exten =>71,n,Playback(vm-nobodyavail) 
exten =>71,n,Voicemail(71)
exten =>71,n,Hangup


best regards nedi 

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Re: [Astlinux-users] Astlinux all account unregistered and in my Log I have DriveReady SeekComplete Error

[nedi]

Hi Michael,
thanks, I use there a CF Card I don't know why is it shown as hda ant not as 
sda. I have there astlinux 0.79 

best regards 
Nedi

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[Astlinux-users] Astlinux all account unregistered and in my Log I have DriveReady SeekComplete Error

[nedi]

Hi 
I have DriveReady SeekComplete Error in my log file and all accounts are 
unregistered, is my SD Card damaged?


Latest System Logs:

Jul  3 13:00:47 allversal user.warn kernel: hda: task_in_intr: status=0x51 { 
DriveReady SeekComplete Error }
Jul  3 13:00:47 allversal user.warn kernel: hda: task_in_intr: error=0x04 { 
DriveStatusError }
Jul  3 13:00:47 allversal user.warn kernel: ide: failed opcode was: unknown
Jul  3 13:00:47 allversal user.warn kernel: hda: task_in_intr: status=0x51 { 
DriveReady SeekComplete Error }
Jul  3 13:00:47 allversal user.warn kernel: hda: task_in_intr: error=0x04 { 
DriveStatusError }
Jul  3 13:00:47 allversal user.warn kernel: ide: failed opcode was: unknown
Jul  3 13:00:47 allversal user.warn kernel: hda: task_in_intr: status=0x51 { 
DriveReady SeekComplete Error }
Jul  3 13:00:47 allversal user.warn kernel: hda: task_in_intr: error=0x04 { 
DriveStatusError }
Jul  3 13:00:47 allversal user.warn kernel: ide: failed opcode was: unknown
Jul  3 13:00:47 allversal user.warn kernel: hda: task_in_intr: status=0x51 { 
DriveReady SeekComplete Error }
Jul  3 13:00:47 allversal user.warn kernel: hda: task_in_intr: error=0x04 { 
DriveStatusError }
Jul  3 13:00:47 allversal user.warn kernel: ide: failed opcode was: unknown
Jul  3 13:00:47 allversal user.warn kernel: ide0: reset: success
Jul  3 13:00:47 allversal user.warn kernel: hda: task_in_intr: status=0x51 { 
DriveReady SeekComplete Error }
Jul  3 13:00:48 allversal user.warn kernel: hda: task_in_intr: error=0x04 { 
DriveStatusError }
Jul  3 13:00:48 allversal user.warn kernel: ide: failed opcode was: unknown
Jul  3 13:00:48 allversal user.warn kernel: hda: task_in_intr: status=0x51 { 
DriveReady SeekComplete Error }
Jul  3 13:00:48 allversal user.warn kernel: hda: task_in_intr: error=0x04 { 
DriveStatusError }
Jul  3 13:00:48 allversal user.warn kernel: ide: failed opcode was: unknown
Jul  3 13:00:48 allversal user.warn kernel: hda: task_in_intr: status=0x51 { 
DriveReady SeekComplete Error }
Jul  3 13:00:48 allversal user.warn kernel: hda: task_in_intr: error=0x04 { 
DriveStatusError }
Jul  3 13:00:48 allversal user.warn kernel: ide: failed opcode was: unknown
Jul  3 13:00:48 allversal user.warn kernel: hda: task_in_intr: status=0x51 { 
DriveReady SeekComplete Error }
Jul  3 13:00:48 allversal user.warn kernel: hda: task_in_intr: error=0x04 { 
DriveStatusError }
Jul  3 13:00:48 allversal user.warn kernel: ide: failed opcode was: unknown
Jul  3 13:00:48 allversal user.warn kernel: ide0: reset: success
Jul  3 13:00:48 allversal user.warn kernel: hda: task_in_intr: status=0x51 { 
DriveReady SeekComplete Error }
Jul  3 13:00:48 allversal user.warn kernel: hda: task_in_intr: error=0x04 { 
DriveStatusError }
Jul  3 13:00:48 allversal user.warn kernel: ide: failed opcode was: unknown
Jul  3 13:00:48 allversal user.err kernel: end_request: I/O error, dev hda, 
sector 611392


Nedi

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[Astlinux-users] Can anyone help me with the script from Lonnie how to use follow me, blacklist and White list

[nedi]

Hi,
Can anyone help me with the script from Lonnie how to use those macro's in my 
case 
 
I have my incomming calls going to internal 10. 
I would like to Check all calls if those are on the blacklist
I would like to forward incomming calls sometime to antoher internal extensions 
or external my mobile number.
On Extensions 12 I would like to check if on White list if not on White list 
the call musst be forwardet to extensions 13
 
In the prefs I have
Number Format:  ^[0-9]{2,16}$
CID Name Max Length: 16
 
I have this in my extensions.conf , lockup.agi script make reverese lockup for 
the caller ID. 
 
exten =>10,hint,SIP/10
exten =>10,1,Set(CHANNEL(language)=de)
exten =>10,n,AGI(lookup.agi, ${CALLERID(num)})
exten =>10,n,Set(CALLERID(name)=${LONGNAME})
exten =>10,n,Macro(local-followme,${EXTEN})
exten =>10,n,System(/mnt/kd/email_in.sh "${CALLERID(name)}" "${CALLERID(num)}")
exten =>10,n,Dial(SIP/10&SIP/12&SIP/17&SIP/18,35,tTwWxXr)
;exten =>10,n,Dial(SIP/10&SIP/12&SIP/032510@032520,35,tTwWxXr)
exten =>10,n,system(/mnt/kd/sms_077.sh "${CALLERID(name)}" "${CALLERID(num)}")
exten =>10,n,Playback(vm-nobodyavail) 
exten =>10,n,Voicemail(10)
exten =>10,n,Hangup 

exten =>12,hint,SIP/12
exten =>12,1,Dial(SIP/12,35,tTwWxXr)
exten =>12,n,Answer
exten =>12,n,Playback(vm-nobodyavail)
exten =>12,n,Voicemail(12) 
exten =>12,n,Hangup

[macro-local-followme]
exten => s,1,GotoIf($[${DB_EXISTS(followme/${ARG1})}=0]?nofollow)
exten => s,n,GotoIf($[${DB_RESULT:0:1}=0]?nofollow:follow)
exten => s,n(follow),Dial(SIP/${ARG1},20)
exten => s,n,Followme(${ARG1},san)
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s,n(nofollow),Dial(SIP/${ARG1},20)
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to voicemail
exten => s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/ busy ann
exten => _s-.,1,Goto(s-NOANSWER,1)
 
[Blacklist-Zap]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Zapateller
exten => s,4,Zapateller
exten => s,5,Playback(ss-noservice)
exten => s,6,Hangup
 
[Blacklist-Hangup]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Hangup
 
[Blacklist-VM]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Voicemail(11,u)
exten => s,4,Hangup

Best Regards 
Nedi

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Re: [Astlinux-users] Asterisk won't play gsm files if the call coming from outside

[nedi]

Thanks Michael, 
I have copied all files from /stat/var/lib/asterisk/sounds/de folder  to 
/stat/var/lib/asterisk/sounds/ and it works.

I think sometimes  in sip.conf  language=de is ignored.

Best Regards Nedi

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[Astlinux-users] Asterisk won't play gsm files if the call coming from outside

[nedi]

Hi, 
my Astlinux astlinux-1.1.1 - Asterisk 1.8.21.0  won't play gsm files if the 
call coming from outside

if the call coming from inside all is o.k.  

I have in the sip.conf  language=de  and german files are in folder 
/stat/var/lib/asterisk/sounds/de

what can I try?

instead from german prompts I get english prompts

regards Nedi

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[Astlinux-users] how to use followme, blacklist and White list in astlinux with macro from lonnie

[nedi]

Hi,
how to use those macro's in my case 

 

I have my incomming calls going to internal 10. 
I would like to Check all calls if those are on the blacklist
I would like to forward incomming calls sometime to antoher internal extensions or external my mobile number.
On Extensions 12 I would like to check if on White list if not on White list the call musst be forwardet to extensions 13    

 

In the prefs I have

Number Format:  ^[0-9]{2,16}$
CID Name Max Length: 16

 

I have this in my extensions.conf , lockup.agi script make reverese lockup for the caller ID. 

 

exten =>10,hint,SIP/10
exten =>10,1,Set(CHANNEL(language)=de)
exten =>10,n,AGI(lookup.agi, ${CALLERID(num)})
exten =>10,n,Set(CALLERID(name)=${LONGNAME})
exten =>10,n,Macro(local-followme,${EXTEN})
exten =>10,n,System(/mnt/kd/email_in.sh "${CALLERID(name)}" "${CALLERID(num)}")
exten =>10,n,Dial(SIP/10&SIP/12&SIP/17&SIP/18,35,tTwWxXr)
;exten =>10,n,Dial(SIP/10&SIP/12&SIP/032510@032520,35,tTwWxXr)
exten =>10,n,system(/mnt/kd/sms_077.sh "${CALLERID(name)}" "${CALLERID(num)}")
exten =>10,n,Playback(vm-nobodyavail) 
exten =>10,n,Voicemail(10)
exten =>10,n,Hangup 


exten =>12,hint,SIP/12
exten =>12,1,Dial(SIP/12,35,tTwWxXr)
exten =>12,n,Answer
exten =>12,n,Playback(vm-nobodyavail)
exten =>12,n,Voicemail(12) 
exten =>12,n,Hangup


[macro-local-followme]
exten => s,1,GotoIf($[${DB_EXISTS(followme/${ARG1})}=0]?nofollow)
exten => s,n,GotoIf($[${DB_RESULT:0:1}=0]?nofollow:follow)
exten => s,n(follow),Dial(SIP/${ARG1},20)
exten => s,n,Followme(${ARG1},san)
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s,n(nofollow),Dial(SIP/${ARG1},20)
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to voicemail
exten => s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/ busy ann
exten => _s-.,1,Goto(s-NOANSWER,1)

 

[Blacklist-Zap]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Zapateller
exten => s,4,Zapateller
exten => s,5,Playback(ss-noservice)
exten => s,6,Hangup

 

[Blacklist-Hangup]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Hangup

 

[Blacklist-VM]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Voicemail(11,u)
exten => s,4,Hangup


Regards 
Nedi


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Re: [Astlinux-users] is there a way to use ? symbol in Password, I have newest Astlinux and all passwords with ? are not accepted.

[nedi]

Thanks

register => sip-inbound?1234:password@..   works
 
Regards Nedi

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Re: [Astlinux-users] is there a way to use ? symbol in Password, I have newest Astlinux and all passwords with ? are not accepted.

[nedi]

I mean SIP Provider Password 

 

Nedi

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[Astlinux-users] is there a way to use ? symbol in Password, I have newest Astlinux and all passwords with ? are not accepted.

[nedi]

Hi, 
is there a way to use  "?" symbol in Password, I get those password from 
provider an I have newest Astlinux but all passwords with ? in Password are not 
connected I think ? is not accepted.
Regards Nedi

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[Astlinux-users] What Hardware would you recommend with which I can connect AstLinux to ISDN NT.

[nedi]

Michael, 
do the fritzbox work well as 2 Channer ISDN VOIP Gateway without echo?

Nedi

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[Astlinux-users] What Hardware would you recommend with which I can connect AstLinux to ISDN NT.

[nedi]

Michael, do the fritzbox work well as 2 Channer ISDN VOIP Gateway without echo?Nedi

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[Astlinux-users] What Hardware would you recommend with which I can connect AstLinux to ISDN NT.

[nedi]

Hi I would like for first time connect my Astlinux with ISDN NT (2 ISDN Line) What Hardware would you recommend with which I can connect AstLinux to ISDN NT. I would like to use this hardware as  ISDN VoIP gateway with In and out call routing.I've seen some people use AVM Fritzboxand maybe I can use Gigaset VOIP ISDN telephone as ISDN2VOIP Gateway. Does anyone Gigaset ISDN Phone DE PRO series tries AstLinux as a gateway.Auserwald PBX can be used with Askozia PBX, Is Auserwald PBX Hardware with ISDN compatible with AstLinux.regards nedi

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[Astlinux-users] is there a way to make staff access only read only inkl. network settings to

[nedi]

Hi
is there a way to make staff access only read only inkl. network settings to
regards
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[Astlinux-users] Copy Astlinux from CF card to USB thumb

[nedi]

Hi
i tried to copy Astlinux SD card to USB thumb and I the Astlinux boot for about 
50% and stop there should I make some changes somewhere to boot from USB thumb 
instead from CF card.
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[Astlinux-users] (no subject)

[nedi]

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[Astlinux-users] BLF and Snom 320 sometimes works sometime not

[nedi]

Michae,
Thanks

If I have 2 Hints 79@hints  and DND_79 Custom hint how can I subscribe to both 
of them  with only one Snom key?
I have this on Snom 320  BLF |*8
can I left this settings?

I can monitor 79 if tis Extensios is BUSY or if there ringing I can call 79 or 
Transfer to 79 an I can use the Key  for pickup  instead of *8


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[Astlinux-users] BLF and Snom 320 sometimes works sometime not

[nedi]

Michael,

have I understand you?
- I can't use the internal DND function key  wit LED (Snom DND key) for this 
function.

- I should make a extension and call this extension to put  DND on and off 

- Than should be one snom key mapped to call this extension to put the DND on 
and off.

if I have a normal hint for the Phone state mapped on one snom key for BLF on 
extension 79 can I make a Custom DND hint to for this extension 79?
and can I subscribe with one snom key to extension 79 and show both 
subscription on only one snom key with lamp?

Should I make 2 hints one normal for BLF to show the phone state if the phone 
ringing and  if the phone is busy  and one custom dnd hint to show if the phone 
on DND?

now I have so the sonm key mapped  to show BLF

BLF |*8

with *8 for pickup can I left this settings to monitor both hints  

Thanks
Regards
Nedi 
 


 
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Re: [Astlinux-users] BLF and Snom 320 sometimes works sometime not

[nedi]

Hi Michael,
Thanks 
I have make those changes and  now I have only one hint 79@hints. 
I can' test on my Customers PBX if this Works on mine Test PBX works.

If i make a custom state for DND and I would like subscribe those snom phone 
state on other snom320 

I have now in DB this  /CustomDevstate/DND_79: INUSE

How can I make a subscription on Snom320 for Custom DND dev state, Ringing 
State and On Phone state
. 
I have now this on Snom 320 for the extension 79 

 |*8 

the subscriptions work I can: 
Pickup all ringing phone 
Dial internal 79 if press this Snom Key 
Transfer to 79 If I press this Snom Key

that all works, but i need DND state in the same Snom 320 key. 

Can I have subscription DND79 und 79 Normal BLF if on Phone, Ringing on 79 and 
make transfer to 79, dial 79 on only one Snom BLF key.

Regards Nedi
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[Astlinux-users] how can i delete ustomdevstate from Aterisk Db

[nedi]

Thanks James,
That Work with database del CustomDevstate 79
Regards Nedi
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[Astlinux-users] how can i delete ustomdevstate from Aterisk Db

[nedi]

Hi James,
I have tried this and get Database entry does not exist.

mypbx*CLI> database show customdevstate 
 
/CustomDevstate/79: BUSY 
/CustomDevstate/DND_79: INUSE   
 
/CustomDevstate/SIP/79: inuse   
 
3 results found.
 
mypbx*CLI> database del customdevstate /79  
 
Database entry does not exist.
mypbx*CLI> database del customdevstate 79   
 
Database entry does not exist.
regards Nedi
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[Astlinux-users] how can i delete ustomdevstate from Aterisk Db

[nedi]

Hi 
I tried to make work DND an BLF and I have put some Customdevstate in the 
Asterisk Db how can I delete them.

mypbx*CLI> database show customdevstate
/CustomDevstate/79: BUSY 
/CustomDevstate/DND_79: INUSE   
 
/CustomDevstate/SIP/79: BUSY

database del CustomDevState 79 don't work for me
regards Nedi


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[Astlinux-users] BLF and Snom 320 sometimes works sometime not

[nedi]

Hi,
I found in the CLI in one Astlinux PBX which I give to my Customer, if I put in 
the CLI "core show hints" I can see the Phone Status but If I make a Call and 
if I put in the CLI "core show hints" again the status of Phone not Change what 
can be the Problem?

should I change something on the Other Config files for a Working hint
I have in

sip.conf

[79]
type=friend
username=79
secret=xx
callerid="79" <79>
host=dynamic
mailbox=70@default
dtmfmode=info
canreinvite=no
insecure=port,invite
context=79
disallow=all 
allow=alaw
allow=ulaw
callgroup=1
pickupgroup=1
notifyringing=yes
callcounter=yes
limitonpeers = yes

in Extensions.conf

[default]
include => hints

exten =>79,hint,SIP/79
exten =>79,1,Set(CHANNEL(language)=de)
exten =>79,n,AGI(lookup.agi, ${CALLERID(num)})
exten =>79,n,Set(CALLERID(name)=${LONGNAME})
exten =>79,n,Dial(SIP/79,18,r)
exten =>79,n,Playback(vm-m-no-avail)
exten =>79,n,Dial(SIP/70,18,r)
exten =>79,n,Voicemail(70)
exten =>79,n,Hangup

[hints]
exten => 79,hint,SIP/79


I have another Astlinux PBX at home with the same Firmware and there works all 
with the same settings.

has anyone idea what I should check can be the Firewall block something? there 
is a Sonicwall.

Best Regards

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[Astlinux-users] can anyone help me configure Astlinux and Snom 320 to show DND through BLF

[nedi]

Hi 
I try to configure BLF o0 snom 320 phone to use BLF and Show if someone put DND 
without success. can anyone give me information how that work on Astlinux and 
what I must do to show subscribe snom 320 to show if someone put DND.
BLF works but I can't see DND through BLF.
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[Astlinux-users] BLF and Snom 320 sometimes works sometime not

[nedi]

Hi
i have newest Astlinux 5 SNom 320 newest Firmware 8.7.3.15
sometime BLF works and sometime won't work and is there a way to show DND with 
BLF

sip.conf

[71]
type=peer
port=5060  
username=71
secret=
host=dynamic
mailbox=71@default
dtmfmode=info
canreinvite=no
insecure=port,invite
context=71
disallow=all 
allow=alaw
allow=ulaw
callgroup=1
pickupgroup=1
notifyringing=yes
callcounter=yes
incominglimit=1
limitonpeers = yes

[79]
type=peer
port=5060  
username=79
secret=xxx
host=dynamic
mailbox=79@default
dtmfmode=info
canreinvite=no
insecure=port,invite
context=79
disallow=all 
allow=alaw
allow=ulaw
callgroup=1
pickupgroup=1
notifyringing=yes
callcounter=yes
incominglimit=1
limitonpeers = yes

I have include => hints  in extensions.conf  under Default
I have internal nubers under [intern]

I have in my extensions.conf
[hints]
exten => 71,hint,SIP/71
exten => 79,hint,SIP/79

every extensions begin with exten =>71,hint,SIP/ext...

exten =>71,hint,SIP/71
exten =>71,1,Dial(SIP/71,25,tTwWxXr)
exten =>71,n,Answer
exten =>71,n,Playback(vm-nobodyavail) 
exten =>71,n,Voicemail(71) 
exten =>71,n,Hangup

exten =>79,hint,SIP/79
exten =>79,1,Dial(SIP/79,25,tTwWxXr)
exten =>79,n,Answer
exten =>79,n,Playback(vm-nobodyavail) 
exten =>79,n,Voicemail(79) 
exten =>79,n,Hangup

CLI show me this and I thing Asterisk is not the problem:

71@hints : SIP/71  State:Idle   Watchers 0  
 
71@intern : SIP/71 State:Idle Watchers 1   
79@hints : SIP/79 State:Idle Watchers 0   
79@intern : SIP/79  State:Idle Watchers  1  


regards
Nedi
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[Astlinux-users] The Customer ask me for Asterisk Admin Password

[nedi]

Hi 
one of my customer ask me for Asterisk admin password, I would prefer to avoid 
handing over of Password.
what should I do?
Best Regards
Nedi

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[Astlinux-users] if firewall enabled with no rule's, I can't acces to Astlinux through web interface or ssh

[Nedi]

Hi 
I have to nevest Version  updated  to version 1.0.3 - Asterisk 1.8.11.1 Runnix 
Release: runnix-0.4-5339
If I enable Arno Firewall  I can’t access to Astlinux through web interface or 
ssh. 
I don’t put any rules in the Firewall settings and no plugin is active. what 
can be wrong 
regards Nedi 



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[Astlinux-users] Fwd: if firewall enabled with no rule's, I can't acces to Astlinux through web interface or ssh

[Nedi]

Hi , 
I have to nevest Version  updated  to version 1.0.3 - Asterisk 1.8.11.1 Runnix 
Release: runnix-0.4-5339, If I enable Arno Firewall  I can’t access to Astlinux 
through web interface or ssh. , I don’t put any rules in the Firewall settings 
and no plugin is active. what can be wrong?
regards Nedi 
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[Astlinux-users] Astlinux and Fraud Protection is there a way to block some Destination Countries and protect from Fraud?

[Nedi]

Hi,
has someone goot solution to protect astlinux Asterisk from fraud? 
Is there a secure  way to block some destination/Countries and protect asterisk 
from Fraud’
regards Nedi


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[Astlinux-users] alot of mail servers won't accept an Voicemail from Astlinux with Return-Path root@mymailserver

[Nedi]

Thanks to all 
this settings in user variable user.conf  works 
SMTP_FROM="myemail@..."

Best Regards
Nedi

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[Astlinux-users] alot of mail servers won't accept an Voicemail from Astlinux with Return-Path root@mymailserver

[Nedi]

hi, 
can anyone help my  to fix that problem
For alot of mail servers to accept an email it must have either a valid 
Return-Path: or Reply To:  If the Return Path is not valid as I have  My 
Astlinus server sending return Path r...@mymailserver.ch  the receving server 
looks for a Reply To: if this is not a valid email address or simply not added 
to the email headers the mail server will reject the email as it will fail a 
basic whitelist check.  

Atlinux sending Email from=root@my mailserver or  from from=root@myIpadress of 
mymailserver to my email adress recipients=nedi@xx

how can I change that  root@my mailserver to  my email address   most od mail 
server block those emails from root@ as return path.

in my Network settings I have 
SMTP Authentication: Login
MTP Encryption: SSL/SMTP   Ignore Cert 
SMTP Username: is my from  email address

on Status after sending email I have  
mail.info msmtp: host=mymailserver.ch tls=on auth=on user=v...@mymailserver.ch 
from=r...@mymailserver.ch recipients=n...@mymailserver.ch mailsize=345 
smtpstatus=250 smtpmsg='250 ok 1337998203 qp 24326' exitcode=EX_OK

regards
Nedi
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[Astlinux-users] how to and where change automon folder to /tmp/usb/record/

[Nedi]

Hi 
how to and where change automon folder“/var/spool/asterisk/monitor“  to " 
/tmp/usb/record/ „
regards
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[Astlinux-users] How to make with DTMF Call Hold/ Transfer with Astlinux ?

[Nedi]

Dear Michael 
thanks to your quickly answer
I know that new phone have this future for the call transfer and hold. but my 
Iphone not I can only transfer and make conference call and I can’t go out from 
conference

I would like to  use DTMF tone to  make Hold and Transfer if I forwarded a call 
to  mobile phone and after answering I will try to hold the call inside of 
astlinux and  make attended call transfer again to internal number inside 
Astlinux. 

can be there is a software client for Iphone to make this?

thanks 
nedi

 


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[Astlinux-users] How to make with DTMF Call Hold/ Transfer with Astlinux ?

[Nedi]

Hi 
since long time I tried to find a soulutions for the attended call transfer  
with DTMF and Astlinux

how can I activate the call transfer an call hold with DTMF in astlinux. Or is 
this standard activated and i must use  # or * to make call transfer?

DTMF works well in other case only I cant make hold and transfer?
best regards
nedi

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[Astlinux-users] Has anyone tried the vodacom VadaXchange Dialer App with astlinux

[Nedi]

Hi 
Has anyone tried the vodacom  VadaXchange Dialer App  with astlinux, this app 
need a python CGI script on pbx, 
where is the cgi folder on astlinux?

This app is great and free to use with astierisk

I need a solution to manage calls forwarded to my Iphone ( call Hold,Transfer )

I would like to be able make the call transfer through web with the app to 
transfer calls  wich are forwarded to my iphone 

the call transfer on Phone is not possible I cann make on Iphone only 
conference and i can’ go out from conference. 
Or Has anyone another way hot to manage calls with Iphone.
 
app link: vadacom.co.nz/iphoneos/calls-iphoneos
-

Regards
Nedi

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[Astlinux-users] Steal call / Pickup how can this work in astlinux?

[Nedi]

Hi, 
I have 2 Phones with extension  11 and 12   if someone call the extension  11 
and the call is connected to extension 11 how can I pickup/steal  the call from 
the extension 12 if I cal from 12?   I found this  snippet off extensions.conf 
,  the snippet make a pickup/ steal the phone if the call connected but this 
snippet have pipes and new Astlinux 1.8 don’t use pipes  I tried to change the 
pipes with ,  and the snippet not works. Has anyone  working script for pickup 
which i can use to pickup a connected call from another phone. if the call 
rings I can pickup with *811 but if the call is connected i can’t fount nöthing 
that works.

exten => 11,1,ChanIsAvail(SIP/11|js)
exten => 11,2,Dial(SIP/11)
exten => 11,3,Hangup
exten => 11,102,GotoIf($[${CALLERID(num)}=12]?steal:)
exten => 11,103,Busy()
exten => 11,104,Hangup()

Regards 
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[Astlinux-users] can anyone explain me how to use those extensions.conf snippet from Lonnie for folowme blacklist and whitelist

[Nedi]

Hi 
how can I use the snipet from Lonie in my extensions conf  for folowme 
blacklist and whitelist 

my extensions.conf  for extensions 11 look so:


exten =>11,1,Dial(SIP/11,25,r)
exten =>11,n,Answer
exten =>11,n,Playback(vm-nobodyavail) 
exten =>11,n,Voicemail(11) 
exten =>11,n,Hangup


what to change to use astlinux tabs for make folowme and black-white list  for 
that extension 11 

[macro-local-followme] 
exten => s,1,GotoIf($[${DB_EXISTS(followme/${ARG1})}=0]?nofollow)
exten => s,n,GotoIf($[${DB_RESULT:0:1}=0]?nofollow:follow)
exten => s,n(follow),Dial(SIP/${ARG1},20) 
exten => s,n,Followme(${ARG1},san) 
exten => s,n,Goto(s-${DIALSTATUS},1) 
exten => s,n(nofollow),Dial(SIP/${ARG1},20) 
exten => s,n,Goto(s-${DIALSTATUS},1) 
exten => s-NOANSWER,1,Voicemail(${ARG1},u)  ; If unavailable, send to voicemail
exten => s-BUSY,1,Voicemail(${ARG1},b)  ; If busy, send to voicemail w/ busy ann
exten => _s-.,1,Goto(s-NOANSWER,1)


;blacklist snippet 

exten => s,100,GotoIf($[${DB_EXISTS(blacklist/${CALLERID(num)})} = 0]?200) ; 
blacklist test
exten => s,n,GotoIf($["${DB_RESULT}" = "0"]?110)
exten => s,n,GotoIf($["${DB_RESULT}" = "2"]?120)
exten => s,n,Goto(blacklist,s,1) ; "1" TN in blacklist database, answer and 
Zapateller

exten => s,110,Goto(blacklist,no-answer,1) ; "0" TN in blacklist, don't answer
exten => s,120,Goto(voicemail-ivr,s,1) ; "2" TN in blacklist, direct to 
voicemail

exten => s,200,NoOp(Valid TN:${CALLERID(num)})

;whitelist snippet

exten => s,100,GotoIf($[${DB_EXISTS(whitelist/${CALLERID(num)})} = 0]?200) ; 
whitelist test
exten => s,n,GotoIf($["${DB_RESULT}" = "0"]?110)
exten => s,n,GotoIf($["${DB_RESULT}" = "2"]?120)
exten => s,n,GotoIf($["${DB_RESULT}" = "3"]?130)
exten => s,n,GotoIf($["${DB_RESULT}" = "4"]?140)
exten => s,n,Goto(whitelist,s,1) ; "1" TN in whitelist database, Priority

exten => s,110,Goto(voicemail-ivr,s,1) ; "0" TN in whitelist, direct to 
voicemail
exten => s,120,Goto(whitelist,standard,1) ; "2" TN in whitelist, Standard
exten => s,130,Goto(whitelist,followme,1) ; "3" TN in whitelist, Follow Me
exten => s,140,Goto(whitelist,ivr,1) ; "4" TN in whitelist, IVR

exten => s,200,NoOp(Valid TN:${CALLERID(num)})

regards nedi 




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[Astlinux-users] can anyone explain me how to use those extensions.conf snippet from Lonnie for folowme blacklist and whitelist

[Nedi]

Hi, can anyone explain me how to use those extensions.conf snippet from Lonnie. 
I tried some times to insert it in my dialplan without success.

in the Prefs I changed the Number Format  to  ^[0-9]{6,16}$and  CID Name Mx 
Lengt to 16 

I would like to have follow me  with Internal  and External Number.

should I only insert those snippets in my dial plan  at the begin or at the 
end? Or shoul I change the exten => s to my internal Extension?

has anyone the working sample of extensions.com to show me how that works.


http://lonnie.abelbeck.com/astlinux/info/webgui.php


Thanks  

nedi



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