similar to: Anyone have provisioning documentation for LeadTek devices?

I have written up an guide on how to do bulk provisioning of the Linksys phones and ATAs. http://voipspeak.net/index.php?option=com_content <http://voipspeak.net/index.php?option=com_content&task=view&id=73> &task=view&id=73 Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 -

hello, I would to define and unattended process to configure devices which are http-server-enabled, use DHCP but do not use TFTP-DCHP to configure themselves during boot. Has anyone worked on such subject ? I was thinking of something like : populating configuration file from device web pages (rendering this as generic and flexible as possible) writing a script which reads this file and set

Has anyone got any information on bulk provisioning of Linksys SPA-941/94s? There is an overview in the admin guide but it refers to a different provisioning guide that I haven't found anywhere. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com

I have a lot of experience, all of it pretty good, with various Sipura products, Grandstreams, Zultys, IAXy, and numerous SIP/IAX soft phones connecting into Asterisk as clients. Good sound quality, great reliability. I've tried two of the units named in the subject line, and frankly I'm frustrated. Calls usually start out OK, but within a brief period the sound goes totally to

Other SIP clients behind the firewall (not using STUN, work). We have a SIP client using STUN and ICE behind a pfSense firewall. The firewall is behaving oddly. REGISTER packets work fine. But when the client tries to make a call, the first INVITE packet from the client pass through the firewall and makes it to the Asterisk server. The Asterisk server sends back a 401 client sends ACK,

Hello, We need Asterisk to register with a variable and changing number (hundreds) of VoIP providers, is there a way to do this in a database and without reloading the entire sip config? Where Asterisk needs to register is determined by downstream users, so we need to do it real time and with minimal impact on the server. If Asterisk can't do this, is anyone using anything else to

Is there any documentation that explains res_config_curl? Specifically, the format of realtime calls made to the web server and what the return string for each call should look like? -- Eric Chamberlain

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I occasionally need to connect to a serial device and in the past have used minicom and the Keyspan USA-19HS adapter. It has worked flawlessly. Once I upgraded, I could never seem to make this work. The kernel recognizes the adapter, it gets properly assigned as /dev/ttyUSB0, and minicom seems to transmit data to it (as evidenced by the green LED

Clarify language. Clean up grammar. Hyphenate some words. Change "low-ops" to "low-level" since "low-ops" isn't defined or even mentioned anywhere else in the kernel source tree. Signed-off-by: Randy Dunlap <rdunlap at infradead.org> Cc: Juergen Gross <jgross at suse.com> Cc: Ajay Kaher <akaher at vmware.com> Cc: Alexey Makhalov <amakhalov

I'm looking for a way to delay the disconnection of a call to a SIP extension (or pad it with silence) for a few seconds, after an incoming call to that extension hangs up. Rationale: I have an Asterisk PBX (current 1.4.20 codebase), with a Leadtek BVP8051S ATA hooked to an analog phone which has a built-in answering machine. Incoming SIP connections to the appropriate extension are dialed

Hello, I found the cause. It was the default kerberos config on CentOS: /etc/krb5.conf Please add to the wiki page: https://wiki.samba.org/index.php/Setting_up_Samba_as_an_Active_Directory_Domain_Controller that before provisioning we should remove this file. It wasn't confusing only for me, as the resolution I found was in bugzilla: https://bugzilla.samba.org/show_bug.cgi?id=11573

I`m trying to provision my Polycom 501s using Pure-FTP. Everything has been working fine, until I put a new bootlog.ld file. Sniffing the packets between my phone and the FTP server during transfer, I noticed that the file is not 100% sent (it seems to send about a third of it, and then the phone wisely detects it as being corrupted and reboots). I have suceeded in using other FTP clients to

I decided to restart the provisioning process. I stopped the samba_server, deleted /usr/local/etc/smb4.conf and /var/db/samba4. I then decided to do an interactive provisioning and this is the result: [root at smb4-1 ~ (master)]# ll /var/samba4 ; ll /usr/local/etc/smb4.conf ls: /var/samba4: No such file or directory ls: /usr/local/etc/smb4.conf: No such file or directory [root at smb4-1 ~

On 27/12/15 23:14, Viktor Trojanovic wrote: > > > On 27.12.2015 20:48, Marc Muehlfeld wrote: >> Hello Viktor, >> >> Am 27.12.2015 um 19:41 schrieb Viktor Trojanovic: >>> When I provisioned a new AD domain, I used the wrong network interface >>> in the provisioning command. >>> >>> samba-tool domain provision

I know that lvm supports thin provisioning, and I think I have a pretty good grasp on how that works. Does libvirt support lvm thin provisioning and thin snapshots? I know that in order to set up lvm thin provisioning by hand, I have to create a thin-provisioning pool within the volume group, and then thin-provisioned logical volumes and thin-provisioned snapshots can be created within that

Hello, is there a mailinglist where I can post questions regarding Digium IP-phones ? I have the following question : I'm trying to provision a Digium D70 IP-phone from a https provisioning server. The Digium D70 contacts the provisioning server correctly but seems to log in with the wrong credentials : /var/log/ssl_access_log : XX.XX.XX.46 - - [22/Jan/2014:12:15:09 +0100] "GET

Greetings everyone, I've been trying to setup samba domain controller with an Fedora DS LDAP backend. Every time I attempt to provision samba I receive the error message. [root at Linux64test samba-4.0.0alpha17]# ./source4/setup/provision --realm=REDACTED.LOCAL --domain=REDACTED --server-role='domain controller' --ldapadminpass=Redacted --ldap-backend-type=fedora-ds

Hello. I've got a Cisco SPA303 that I'm trying to provision via http. I noticed that this device looks very similar to a PAP2T, so I used that as a template for my provisioning file. However, the result is less than stellar. Line 1 registers and works. However, lines 2 and 3 also register as line 1, effectively giving me a 1-line phone with 3 buttons. Also, the line name is the

Hi all, I saw that there was an auto-provisioning feature on asterisk 1.6.x for the Polycoms. But no real documentation. I would like to know how, exactly, does the network has to be configured to allow that. I used to provision my Polycom phones with the help of tftp or ftp. But if there's a way to do that directly with Asterisk, I would prefer that. So I need to know what I need to do

Hi All, I am trying to build a samba4 server, but getting an python error when provisioning. I follow the instructions in the wiki HowTo, but when I run the provision cmd: ./setup/provision --realm=CREOSYS.CO.UK--domain=CREOSYS --adminpass=PASSWORD --server-role='domain controller' I get the following error: python: Objects/stringobject.c:105: PyString_FromString: Assertion `str !=