login | register
Wed 08 of Oct, 2008 [00:12 UTC]

voip-info.org

ST2030 non-working features

Created by: kurgan,Last modification on Sat 15 of Mar, 2008 [16:12 UTC] by gmaoret
  • Direct dialer software (like OutCALL, HUDlite or Web Phonebooks) - the ST2030 isn't able to accept a standard Asterisk INVITE from any of these software and make it unusable with those phonebook tools
This problem is well reported in Trixbox forum. (gmaoret - 29/01/07)
    • The direct dialer issue (aka "Click to Dial") is fixed in v1.59 which is available now (was:shortly)
      • What's the mean of "shortly"? We are waiting that from long time...
      • There is a scheduled date for this release?
      • There is a planned release date? This is the only function that we are wait for to adopt this phone...
        • Unfortunately, that can't be answered with an exact release date. When testing is complete, the version will be posted on the Thomson site. They work hard to make sure new releases don't cause more problems than they solve.

      • GMAORET- GOOD JOB!!! - Ok, I've got the updated firmware (1.59.3), updated and configured. VERY GOOD JOB!!!. We think that this is now one of the best SIP phone on the market!

  • ST2030 1.58 reject calls with hidden caller-id... (nebz - 06/02/08)
  • Extension Panel stop working when going to Function Key Table page on Web Admin (nebz - 06/02/08)

(RESOLVED) Problems with remote phonebook


The phone always gives an "service unavailable" error if the remote phonebook XML content has to be created dynamically (i.e. by a Perl or PHP script) whereas displaying the content of a static XML file actually works - even if the content is exactly the same (checked with "diff"). The only difference is the missing "Content-Length" header in the HTTP response if the content in generated dynamically (there is a "Transfer-Encoding: chunked" instead).

The problem is the same with application versions 1.42, 1.47 and 1.48b.

Is there any way to use the remote phone book with dynamically generated content?

--> You have to calculate the Content-Length and supply the header yourself, see: http://www.ip-phone-forum.de/showpost.php?p=679094&postcount=2

--> Ronald: I'm using the remote phonebook with dynamic content with all the mentioned versions with succes! Send me an e-mail to voermans at globe dot nl, and I'll send you some sample code.

--> PomTom: In fact, it was the missing "Content-Length"-Header. I can query the remote phone book now.

Reported by gaupe against version 1.42

  • HTTP-client in the phone seems to be quite broken and implements the HTTP 1.1 spec wrong. Complete URL is sent in the GET request and the Host-parameter is filled out with the ip-address of the http-server. Also it identifies itself as MSIE 6, I would rather see the firware version and MAC-address here. Also it request chinese language in the HTTP-request.

  • It is not possible to make the phone authenticate itself by other means than the phone number, auth id seems to be available, but does not seem to get used.


Comments

Comments Filter
222

333

by mlarosa, Saturday 06 of October, 2007 [09:34:07 UTC]
222

333Bug with volume in system melody

by mlarosa, Saturday 06 of October, 2007 [09:03:19 UTC]
I have a problem with the volume in before (in before) only called the external one
The volume is set up to the maximum, but when the line ring, the volume is a lot low
If it comes pressed 1 the volume turns the key vol+ goes to the corrected maximum set
This happens for every single bound together line entering to keys BLF
After to only have pressed (while the line ring) the key volume, in the successive entering calls, the volume remains ok
The volume goes ok also when it is made ring for before turns the inside
222

333Re: disable call waiting?

by raarts, Monday 25 of September, 2006 [01:49:08 UTC]
Hm, I think this is solved now, the provisioning system played tricks on me..
222

333

by Danny, Monday 01 of May, 2006 [10:11:42 UTC]
222

333Re: nothing works in MGCP

by zziiss, Wednesday 19 of April, 2006 [18:03:37 UTC]
Francesco,

What softswitch are you using?

Thomson is not planning to stop MGCP support.
But as the FW is tightly linked to the SoftSwitch you need to upload the right FW.

Stéphane


222

333Re: disable call wating?

by bringe, Tuesday 18 of April, 2006 [12:20:54 UTC]
  • About which configuration's file do you speak ?
222

333

by bringe, Tuesday 18 of April, 2006 [12:20:30 UTC]
  • About which configuration's file do you speak ?
222

333

by bringe, Tuesday 18 of April, 2006 [12:19:21 UTC]
  • About which configuration's file do you speak ?
222

333disable call wating?

by raarts, Monday 17 of April, 2006 [21:16:59 UTC]
I can't disable call-waiting from a config file. According to the docs it should be:

[customer]
sw_call_wait=0

but the phone gives me a syntax error.

222

333nothing works in MGCP

by thecalle, Tuesday 11 of April, 2006 [17:09:54 UTC]
Using MGCP v.1.18:

- no codec setting
- no CA registration
- no endpoint name setting

Did Thomson decided to shut down the MGCP support? SIP is complete and quite good working (using the correct settings).

Any idea?