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118
Registration Date: 01.01.1970
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09.06.2008 03:57 |
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mds
Registration Date: 01.01.1970
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08.06.2009 12:18 |
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Dia
Premium Account
Registration Date: 03.03.2006
Posts: 1443
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08.06.2009 23:10 |
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118
Registration Date: 01.01.1970
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RE: Call Progress Tone Customization |
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Zitat: |
Originally posted by Diafora
Let me actually understand this request, because I must be looking at it from exactly the opposite direction.
The basic premise of SIP as a protocol, is to enable the User Agents to handle most of the call related functions as well as features, instead of relying on the SIP Proxies and Back-end servers, which is the case in traditional telephony with Class-5 switches.
So why would we elect to regionalize the call progress tones on PBXes, and carry the ring-back media (RTP packets) across the Internet, when call progress tones can be created locally on the users' handsets following a change in their settings? |
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FYI: I have customised all my Sip Devices to generate the British style ringback tone (400Hz + 450 Hz 0.4s on 0.2s off 0.4 on 2.0s off "burr burrr,,,,,,burrr bur" etc.)
This is fine when I call between extensions as the British style ringback tone is generated by the device where the call originates from. Note: This is consistent to what you are saying in your post.
However, when an outside caller calls my DID number from the PSTN and gets the PBXes Digital Receptionist, they would enter an extension number to reach one of the internal extensions. What they hear is the default Asterisk North American style ringback tone (440Hz + 480 Hz, 2.0s on 4.0s off "rrrrinnngg rrrrrirrinng" etc.).
This is happening as the PBXes Digital Receiptionst resides on the Asterisk service and not on my SIP device!
This also happens when I do a blind transfer from one extension to another by pressing *2 and then the extension number (as this is an Asterisk process). Again, the caller hears the default North American style ringtone generated by the PBXes Asterisk server.
In addition, when I dial *2 to access the transfer function, the dialtone I get is a North American dialtone (350Hz+440Hz continuous). I prefer the European dialtone of 440Hz! (which I have set all my SIP devices to generate). Again, this is coming from PBXes Asterisk service.
Right now, it seems like the only way to change the ringback tone generated by the Asterisk server is to edit the tones in indications.conf file (this is currently only possible if I get the PBXesPro).
Since I am a SOHO account user, I am not prepared to so much more just so that I can change the indications.conf file.
This post has been edited 5 time(s), it was last edited by 118 on 09.06.2009 at 01:21.
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09.06.2009 01:06 |
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mds
Registration Date: 01.01.1970
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15.06.2009 14:13 |
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Dia
Premium Account
Registration Date: 03.03.2006
Posts: 1443
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16.06.2009 05:31 |
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mds
Registration Date: 01.01.1970
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17.06.2009 15:21 |
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118
Registration Date: 01.01.1970
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17.07.2009 00:48 |
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