similar to: Linksys doesn't resync properly and doesn't get provisioning from TFTP and HTTP

have adaptors compatible with Asterisk, but explicitly say in the product titles that they're unlocked, which I think is the key. On Thu, Dec 17, 2009 at 4:16 AM, Brian Cline <Brian at nw.brian.fm> wrote: > Hello, > > I'm running Asterisk v1.6.1.11 internally with a few Linksys SIP > phones and will be receiving a machine containing a Dialogic card > for a

Hi All, Anyone know the sip header to send to a Linksys to resync it's config file? Thanks. JR -- JR Richardson Engineering for the Masses

Hello! Sorry for the OT-thread, but i don't know where else too ask... Has anyone done http-provisioning of a Linksys SPA942 with client side ssl-authentication? Where do i get the CA from? I'm aware of the Sipura mass deployment howto on voip-info.org, but it doesn't cover the authentification part. Thanks Christian

Hellos, I need to send personal directory from asterisk to the ersonal directory of the linksys spa 942. Is this possible? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve

We're contemplating Linksys WIP 330 phones, but we're concerned about configuration effort. Does anyone have the file format for an XML file to configure this phone? We got auto-provisioning to D/L a file, but the XML file format seems to be a secret... Thanks, Michelle -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 -

Does anyone know how the Linksys PAP2T ATA's can be mass provisioned? Documentation seems to be sketchy, even on the Linksys web site. Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070607/3f90695c/attachment.htm

I'm going to be deploying around 30 IP phones with Asterisk in the near future. I've tentatively settled on the Linksys/Sipura SPA9xx family. I am unclear on the notion of "lines" in the context of SIP phones like these. The SPA942 model has a 2-line-to-4-line upgrade available, but I don't know why I'd need to purchase it. I have tested a SPA942 with Asterisk, and

I came up with this a few days ago, mostly used the wiki examples, didn't have time to post on the wiki yet, maybe one of you guys with a few minutes can throw it up there, really, I forgot my logon. http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom The agi script didn't work for me, wouldn't call the active hint extensions, even though they were there, no

Hello, I'm running Asterisk v1.6.1.11 internally with a few Linksys SIP phones and will be receiving a machine containing a Dialogic card for a development project (in a nutshell, the card receives analog calls while the accompanying software handles automated prompts, etc). The Dialogic card is not SIP-based but will work with an analog line, so I'm looking into adapters that act

Working on a paging system for one of my sites and running into something I can't believe is this hard. In one of the zones, they want to have three different extensions ring over the pa system, using it as a loud ringer. Now the paging system does have a loud ringer built in and I can easily have it do a simultaneous ring, but all of the extensions will sound the same over the loud

It appears as though the 489 Bad Event response to the NAT keep alive event responds to the local address, instead of responding to the NATted address. This causes Linksys phones to go amber (no registration) after a short amount of time after placing calls. Turning the Linksys NAT keep alive off is a workound, but non-ideal in may situations. Apparently the asterisk devs don't even think

Hi, i've been trying to apply the lme apprach to growth curves of children, but lme keeps running for ever and ever as soon as I use a reasonable basis. First Example: Data are 39 boys from the Berkeley growth study, each one measured 31 times at the ages of 1.00 1.25 1.50 1.75 2.00 3.00 4.00 5.00 6.00 7.00 8.00 8.50 9.00 9.50 10.00 10.50 11.00 11.50 12.00 12.50 13.00 13.50

Is anyone out there using any of the newer linksys phones since Cisco took over? I am more specifically looking at the spa-941 & 942's. Just curious about call quality, programability, and functionality with asterisk. I have read through the literature, but would like some real world feedback. Thanks

Found my answer for those who would like to know: Profile Rule: [--key $A]http://your.addre.ss/$B/$MAC.cfg GPP A: urtopsecretultrasecureaesencryptionkey GPP B: OddBallDirectory123098 Hope that helps someone! Curt -----Original Message----- From: Curt Shaffer [mailto:cshaffer@gmail.com] Sent: Wednesday, February 07, 2007 11:35 AM To: 'Asterisk Users Mailing List - Non-Commercial

Dear All, 1. any one know how to reset the 941 to factory defaults? 2. any one know how to config' the 941 via flat configuration file via TFTP? sample file? URL? (LinkSys tech support has been just a hair above worthless to me thus far) Thanks very much. www.sjobeck.com

Hi there, We have plans to install an office (not call center) with the following setup: 200 linksys 942 phones (sip + g711) on a LAN a server with a dual port E1 sangoma and a remora card with 4 fxo modules. So far when we want to setup a linksys phone, we need to use the http interface of each phone, disable/enable a lot of things and plug it into the network. this is not the best scenario for

Hi All, Has anyone gotten the polycoms or the linksys phones to accept oprtion 66 on the dhcp request for the address of the tftp config server? We have the dhcp server issuing the proper IP of the tftp server, but the phones just sit there and never try to contact the tftp server for their configs. We can see the proper option going from the dhcp to the phones with ethereal trace. Thanks JR

Hi people, I'm having problems of connection with a Linksys SPA IP PHONE 942 when I use the WAN port, most of the time when I try to connect to the network or restart the IP Phone I can't get internet connectivity . I tried using both static IP and DHCP, but the problem is the same. Some have had similar problem with this brand of IP Phone? Thanks for the help and attention. Hugs

Hi, I don't understand why a SIP hardphone can't provision itself using tftp. I'm very suspicious about my tftp daemon but I lack basic knowledge of Linux CLI to pinpoint what's going wrong and separate what belongs to SIP phone configuration from what comes from tftp server. What I would like to do is to add a given file in current /srv/tftp directory and test by hand that tftpd