Make sure you have busydetect=yes busycount=3 somewhere below your [general] context in chan_dahdi.conf (or zapata.conf depending on your asterisk version) and restart the the service.
This should be enoough to do the magic. Alyed 2010/3/21 Daniel Bareiro <daniel-lis...@gmx.net> > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hi, Gordon. > > On Sun, 21 Mar 2010, Gordon Henderson wrote: > > >> I'm testing with a Grandstream BT200 telephone and, according to I > >> read, it has a LED that blinks if for that extension messages were > >> left. > >> > >> In "Voice Mail UserID", under the "ACCOUNT" tab, I put *100 that is > >> the extension in which my Asterisk answer the voicemail service and > >> if then I press MESSAGE button, the telephone communicates with > >> Asterisk and, after to introduce the password, it indicates to me > >> that I have messages. But the luminous indicator does not work. > >> > >> It is necessary to configure something special for this? It can be > >> that it doesn't work because there is to introduce one password > >> previously? > > > There's another setting in the phone you need to set "SUBSCRIBE for > > MWI". > > Yes. I was needing to indicate the use of MWI of the side of the > configuration of the telephone. I selected the "SUBSCRIBES for MWI" > checkbox. > > > And make-sure the mailbox number is listed in the sip.conf entry for > > that phone. > > According to which I was reading, the MWI notifications become by the > option "mailbox=" in the configuration of the extension. For this > extension, the 104, had "mailbox=104" but still with MWI enabled option, > it was not working. After to think enough on this subject, I have > noticed that instead of 104 I had to put 1...@voicemail since "voicemail" > it was context that I'm using in voicemail.conf. > > With this already was working. > > However, beyond this, I was with the following situation: if I call from > a cell phone, my Asterisk take the call, it presents to the caller the > possibility to dialing an extension number and, in case of not doing it, > it transfers this call to a specific extension. > > Then, if in this extension nobody takes the call, the service of > voicemail is triggered so that the caller leaves its message from the > cell phone. But if it hangs after to let the message without have > pressed previously the pound key, the channel is taken and no longer any > other call enters the PBX from the PSTN. This does not happen if the > caller presses the pound key after to have left his message. > > As I have a box at which the cable arrives from the PSTN in which there > are two ports of derivation and in one of them it leaves the cable for > the Asterisk PBX (connected only then), after to have detected this > problem I tried connecting in the other port an analog telephone and, > indeed, it did not have tone as if never it had been hung. In addition > this was confirmed because the MWI light never blinked on the telephone. > > After restarting the Asterisk server, yes the MWI light blinks and in > addition I could corob the time in which the channel was "taken" seeing > that the message lasted more than nine minutes. > > To what this problem can be due? It has to do the call is made > specifically from cell phone through the PSTN (because if I leave a > message hanging directly without pressing the pound key from an local > extension, this does not happen)? There is some form to avoid it? > > Thanks for your reply! > > Regards, > Daniel > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (GNU/Linux) > > iEYEARECAAYFAkums0oACgkQZpa/GxTmHTcGpQCghJvfphxc5ZzZhouryA+OlwGm > 20AAoJP64a2EVeigx08D/5g5XN8oBXgf > =Hskd > -----END PGP SIGNATURE----- > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users