I am having a strange problem with a couple of Polycom IP 500 phones. I
know this is not related to Asterisk, but maybe someone here had the same
problem.
I configured my phones following the documentation at voip-info.org and
they are working very well. The only problem I have is that when I dial an
extension like 1100 the phone changes that to 0110 and obviously the call
fails. I have to dial slowly to get the 1100. Does this have anything to do
with the dialplan?
--
Carlos Chavez
Director de Tecnolog?a
Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
Are you using inband DTMF? There are other options but I don't know much about the polycom phones. I have noticed that sometimes when accessing voicemail, it will 'miss' some dtmf tones if they are too short. This doesn't explain the number changing, unless your dial plan is putting in the leading zero. Carlos Chavez wrote:> I am having a strange problem with a couple of Polycom IP 500 phones. I >know this is not related to Asterisk, but maybe someone here had the same problem. > > I configured my phones following the documentation at voip-info.org and >they are working very well. The only problem I have is that when I dial an >extension like 1100 the phone changes that to 0110 and obviously the call >fails. I have to dial slowly to get the 1100. Does this have anything to do >with the dialplan? > >-- >Carlos Chavez >Director de Tecnolog?a >Telecomunicaciones Abiertas de M?xico S.A. de C.V. >Tel: +52-55-91169161 Ext 2001 > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
> Are you using inband DTMF? There are other options but I don't know > much > about the polycom phones. I have noticed that sometimes when accessing > voicemail, it will 'miss' some dtmf tones if they are too short. This > doesn't explain the number changing, unless your dial plan is > putting in > the leading zero.If you are using a version of CVS HEAD from April 2005 or later, you should definitely use dtmf=rfc2833 rather than dtmf=inband. In fact, just use rfc2833 - it works with all versions of asterisk that I've tested, but inband only works with certain versions. Also, you may want to check the digit map (the pattern recognition when you dial). It is in sip.cfg. I know that by default it treats numbers that start in "11" specially, but it shouldn't really transpose numbers. That's weird. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050606/330e316b/attachment.htm
I have never used inband. I always used rfc2833. This problem is seen even if you are using dtmf=rfc2833. the line progressinband=no, fixes that. Seshu _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Noah Miller Sent: Monday, June 06, 2005 3:07 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Polycom 500... Are you using inband DTMF? There are other options but I don't know much about the polycom phones. I have noticed that sometimes when accessing voicemail, it will 'miss' some dtmf tones if they are too short. This doesn't explain the number changing, unless your dial plan is putting in the leading zero. If you are using a version of CVS HEAD from April 2005 or later, you should definitely use dtmf=rfc2833 rather than dtmf=inband. In fact, just use rfc2833 - it works with all versions of asterisk that I've tested, but inband only works with certain versions. Also, you may want to check the digit map (the pattern recognition when you dial). It is in sip.cfg. I know that by default it treats numbers that start in "11" specially, but it shouldn't really transpose numbers. That's weird. -------------------------------------------------------- NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050606/26dbc735/attachment.htm
>> Also, you may want to check the digit map (the pattern recognition >> when you dial). It is in sip.cfg. I know that by default it treats >> numbers that start in "11" specially, but it shouldn't really >> transpose numbers. That's weird. >> > > Its not that it transposes numbers,it induces a delay that seems to > screw up the order in which they are registered. I find that if you > dial without picking up the handset, this problem won't appear, but if > you pick up the handset first it does.That would definitely point to a problem in the digit map then. If you dial on-hook (with no dialtone) the phone doesn't use the digit map. It just takes whatever you put in and waits until you specifically press "Dial". -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050606/b18da0b1/attachment.htm
Yeah it has to do with the dialplan, change the following: 'dialplan.impossibleMatch-Handling' to either 1 for reorder tone, and 2 for allow user to accumulate digits and dispatch call manually with the Send soft key. The default is 0 which means that the digits entered up to and including the point where an impossible match occurred are sent to the server immediately. Also look into changing your dialplan. On 6/6/05, Carlos Chavez <cursor@telecomabmex.com> wrote:> I am having a strange problem with a couple of Polycom IP 500 phones. I > know this is not related to Asterisk, but maybe someone here had the same problem. > > I configured my phones following the documentation at voip-info.org and > they are working very well. The only problem I have is that when I dial an > extension like 1100 the phone changes that to 0110 and obviously the call > fails. I have to dial slowly to get the 1100. Does this have anything to do > with the dialplan? > > -- > Carlos Chavez > Director de Tecnolog?a > Telecomunicaciones Abiertas de M?xico S.A. de C.V. > Tel: +52-55-91169161 Ext 2001 > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >