Re: [asterisk-users] Voicemail, Asterisk and Grandstream BT200
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Alyed. On Mon, 22 Mar 2010, Alyed wrote: > you are right, under [channels] is where it's supposed to be my > mistake, i guess i was thinking in sip.conf :) Perfect :-) >> However, the following doubt arises to me: it would also have had >> this problem for some originating call from a telephone that is not a >> cell phone? > yes, and this can be a really serious problem if you don't fix it. So > don't forget to include this parameters from now on. I have played > with them and found setting busycount=5 is not very efficent, so leave > it to 3 or 4 at most. That problematic. I will consider it in future configurations. Thanks for the explanation. > Good to hear your problem is solved. Thanks again for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkurK40ACgkQZpa/GxTmHTc/7gCfWbbEsqrXrWkM8ByGhQKLun1o kJ0AnA0mB+5+Rn9sWZ20iVemMXZJfhX6 =Vytb -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
Re: [asterisk-users] Voicemail, Asterisk and Grandstream BT200
you are right, under [channels] is where it's supposed to be my mistake, i guess i was thinking in sip.conf :) >However, the following doubt arises to me: it would also have had this >problem for some originating call from a telephone that is not a cell >phone? yes, and this can be a really serious problem if you don't fix it. So don't forget to include this parameters from now on. I have played with them and found setting busycount=5 is not very efficent, so leave it to 3 or 4 at most. Good to hear your problem is solved. Alyed 2010/3/22 Daniel Bareiro > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Hi, Alyed. > > On Mon, 22 Mar 2010, Alyed wrote: > > >> 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? > > > 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. > > It didn't have configured these two parameters so I added now them but > in the [channels] context since I don't have a [general] context (It > does not sound to me that in the file by default generated by Asterisk > there would not be it either, although I can be mistaken). > > Beyond that, with these two parameters, I no longer have the problem > mentioned before. Thanks! > > However, the following doubt arises to me: it would also have had this > problem for some originating call from a telephone that is not a cell > phone? > > Thanks for your reply. > > Regards, > Daniel > > -BEGIN PGP SIGNATURE- > Version: GnuPG v1.4.9 (GNU/Linux) > > iEYEARECAAYFAkunNjQACgkQZpa/GxTmHTfAbACfT8PVkcp/xESdqsiczg3YY/Dd > FGcAn1TdOqiZaKAjLg4h3SDt/34A4bKX > =37qZ > -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
Re: [asterisk-users] Voicemail, Asterisk and Grandstream BT200
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Alyed. On Mon, 22 Mar 2010, Alyed wrote: >> 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? > 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. It didn't have configured these two parameters so I added now them but in the [channels] context since I don't have a [general] context (It does not sound to me that in the file by default generated by Asterisk there would not be it either, although I can be mistaken). Beyond that, with these two parameters, I no longer have the problem mentioned before. Thanks! However, the following doubt arises to me: it would also have had this problem for some originating call from a telephone that is not a cell phone? Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkunNjQACgkQZpa/GxTmHTfAbACfT8PVkcp/xESdqsiczg3YY/Dd FGcAn1TdOqiZaKAjLg4h3SDt/34A4bKX =37qZ -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
Re: [asterisk-users] Voicemail, Asterisk and Grandstream BT200
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 > -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
Re: [asterisk-users] Voicemail, Asterisk and Grandstream BT200
-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
Re: [asterisk-users] Voicemail, Asterisk and Grandstream BT200
On Sat, 20 Mar 2010, Daniel Bareiro wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Hi all! > > 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". And make-sure the mailbox number is listed in the sip.conf entry for that phone. Gordon -- _ -- 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
