similar to: thomson speedtouch ST2030

Hello, After reading all the docs and going through the menus, I still can't find the voicemail access button or menu sequence on the ST2030 (http://www.voip-info.org/wiki/view/Thomson+ST2030) Also I can't get phone provisionning through tftp to work. Configuration files are loaded but the phone seems to ignore them. Any idea?

Hi all, i'm trouble with codec setup on an asterisk machine 1.4.18 and some Thomson ST2030 as extensions. In the users.conf file for internal extension i have: disallow=all allow=g729 allow=alaw allow=ulaw Without any codec installed (i mean with original g729 of asterisk) all go fine, calling from an extension to one other: Peer User/ANR Call ID Seq (Tx/Rx) Format

Hello, I'm trying to upgrade a Thomson ST2030 phone froms its default 1.42 firmware to the latest version (1.56) through tftp. The phone loads the .inf file, then the correct firmware file (as stated in the ST2030S.inf), then it reboots and loops doing these same things again and again. The firmware version on the phone stays at 1.42. Is there a special intermediate firmware version to

Hello, I've got strange problems trying to run asterisk with MGCP ip phones (Thomson ST2030). Situation: "user A" <----- pstn -------> ASTERISK <----- mgcp ------> "user B" "User A", connected behind a PSTN, tries to call "User B". After dialing "User B"'s number, call comes to ASTERISK, ASTERISK contacts

Hi! I wish you all e Happy New Year first! Allthough, I'm relative new to Asterisk, I got our server up and Running, Softphones, ISDN, and a brand new Snom 821 are working flawlessly. :) Platform is Debian 8/Asterisk Packages (11) from Debian Repo. But I am running into problems setting up 2 older Hardphones, Thomson 2030S. :( with in my sip.conf, I have got for this hardphone: [...]

Hello. Once again, I came up with a problem for which I can't seem to find a solution. I'm not able to make BLF work with Thomson ST2030 phones and Asterisk (1.2.13). I've set up hints in dialplan, as well as Subscibe keys on the phone. The LED status gets updated according to the associated line status. However, when a phone is ringing, If I try to pickup the call by pressing the

Hello, I'm using Asterisk 1.6.2.0 and I like to use escape characters with SendText, because I can just delete the message from my phone (Thomson Speedtouch ST2030) display by sending a return-char (\n). But \n is not escaped: I tried already: exten => 222, n, SendText(\n) exten => 222, n, SendText("\n") exten => 222, n, SendText('\n') exten => 222, n,

Hello, I am looking for a replacement for my Thomson ST2030SIP. My specifications are as follows : - 2 lines. - 6 BLF keys. - PoE. Can you give me a return on the models you use ? Thanks. Sil

Patrick Laimbock <patrick at laimbock.com> schrieb: > On 12/30/15 12:24, Luca Bertoncello wrote: > > Ishfaq Malik <ish at pack-net.co.uk> schrieb: > > > >> Do you have a link to the user guide for your exact phone model? > > > > Unfortunately not... > > I have a Thomson ST2022, but I can just find in Internet manual for the > > ST2030...

Ishfaq Malik <ish at pack-net.co.uk> schrieb: > Do you have a link to the user guide for your exact phone model? Unfortunately not... I have a Thomson ST2022, but I can just find in Internet manual for the ST2030... Regards Luca Bertoncello (lucabert at lucabert.de)

MWI-related SUBSCRIBE message to send NOTIFY messages changing phones MWI status. This is fine for me but I'm wondering what if I were using SIP hardphones refusing any such NOTIFY without prior SUBSCRIBE (does such phones exist ?) ? 1. In this case, which URI shall use a hardphone to build its SUBSCRIBE message ? Here is a hand written example. Which value should I substitute to foo (in this

Hi, Is anyone aware of SIP Diversion header ? It seems currently supported by Comverse (formely NetCentrex) softswitch and some hardphones (Thomson ST2030). An old draft (draft-levy-sip-diversion-08.txt) mentions this header. ha I'm wondering if this could be used -------------- next part -------------- An HTML attachment was scrubbed... URL:

Taken from http://www.twice.com/html/pagebeta.cfm?InputKey=2853 ------------------------------------------------------------------------- Thomson, Fraunhofer Developing Enhanced MP3 Codec Called mp3PRO Jan. 7, 2001 By Joseph Palenchar Las Vegas -- Thomson and the Fraunhofer Institute, the co-developers of MP3, are working with a third company to develop an improved MP3 codec intended to deliver

Hi all, I am looking for phones witch support POE, with a good relation between quality and price to work with asterisk. I just see the Thompson st2030 and the Linksys SPA 922 an SPA 942. Witch of this phones or another ones gave you the best results in a productivity enviroment? Thanks in advance. VoipCrazy. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm using a SIP phone (Thomson ST2030) which is able to display text received though Asterisk's SendText() application. I'm using this to display from Asterisk "Forwarded to 0123456789" whenever a user forwards his calls to another number or extension. Test is displayed with white letters on black background. What I can't do at the moment is erasing this

Hi, RFC3680 defines a SIP event package for registration. This event package which can be used through NOTIFY-SUBSCRIBE methods, seems very useful for free sitting or presence applications. This package is supported in various SIP phones (at least Thomson ST2030) : when turned on, this feature adds a new login/logout menu among other things. It can also be used to send Welcome notices to mobile

Hy all of you, I've successfully installed a freepbx solution with 10 extensions : - 5 on Linksys SPA922 - 1 on Linksys SPA942 - 1 on Thomson ST022 Everything seems to work fine with all the hardphones excepts last week. The thomson has a strange behaviour. It can reach french mobile cell phones but when it reaches "fix" phones, the correspondant can't hear the caller. What

Am 07.01.2016 um 10:55 schrieb Frank: > On Wed, 2016-01-06 at 17:03 +0100, Juergen Sauer wrote: Thx, 4answer. :) >> with in my sip.conf, I have got for this hardphone: >> [...] >> [hard1] >> username=hard1 >> secret=correct-and-three-times-checked-4-digit-pin > > In most cases, there is no need to set the "username=" option. The

My colleague left our company, then I have to manage all our phones lines and asterisk: please, apologize me because I'm 'absolute beginner' about voip/asterisk!! Well... all seems work fine; we have some queues and some agents; the "music on hold" works fine when the agent press the hold button on the phone (thomson); the agents have the 'autoanwser' flag

We're having a horrid problem with our asterisk setup. Sometimes calls just go dead - we can't hear what the other end is saying. (I think they can't hear us either). The call doesn't hang up until one of the callers gets bored. Internaly we use Thomson ST2030 SIP phones. Externaly we have 3 ISDN BRI lines (6 channels total), connected to an Eicon Diver server card (4BRI).