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(Created page with "= TN1: Configuring a SPA504g for use with IP-Phone-C = == RTP Packet Size == The IP-Phone-C SIP server only accepts the SDP attribute value a=ptime:20. Any other value will r...")
 
 
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= TN1: Configuring a SPA504g for use with IP-Phone-C =
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= Configuring a Cisco SPA504g for use with IP-Phone-C =
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[https://asahi-net.jp/en/service/ipphone-c/ IP-Phone-C] is a VOIP service provided by [https://asahi-net.jp AsahiNet]. It is a rebranding of NTT's [https://ja.wikipedia.org/wiki/%E3%83%95%E3%83%AC%E3%83%83%E3%83%84#%E3%81%B2%E3%81%8B%E3%82%8A%E9%9B%BB%E8%A9%B1 Hikari] VOIP service, so these notes may be applicable to similar services from other resellers.
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This note describes how to configure a [https://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/small-business-spa500-series-ip-phones/data_sheet_c78-548564.html SPA504g] VOIP handset with the IP-Phone-C service. This information should be applicable to other VOIP handsets, although their configuration details will vary.
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== Subscriber Information ==
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Configure an extension on the handset as follows:
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EXT > Display Name = <Your 050 Phone Number>
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EXT > User Id = <Your 050 Phone Number>
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EXT > Password = <Your Password>
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EXT > Use Auth Id = Yes
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EXT > Auth Id = <Your User Id>
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== Audio Configuration ==
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The IP-Phone-C SIP server only supports the G711u codec. Other codecs, or too many proposed codecs, may trigger a "488 Not Acceptable Here" response. Therefore configuration with just G711u is recommended.
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EXT > Preferred Codec = G711u
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EXT > Use Pref Codec Only = Yes
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== NAT Configuration ==
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If you are using NAT, enable keepalives:
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EXT > NAT Keep Alive Enable = Yes
   
 
== RTP Packet Size ==
 
== RTP Packet Size ==
The IP-Phone-C SIP server only accepts the SDP attribute value a=ptime:20. Any other value will result in a "488 Not Acceptable Here" with the warning "307 Session description parameter not understood".
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The IP-Phone-C SIP server only accepts a SDP attribute a=ptime with the value 20. Any other value will result in a "488 Not Acceptable Here" response, incorporating a "307 Session description parameter not understood" warning. This is an extremely difficult issue to debug without a priori knowledge, and the engineer involved should be ashamed of themselves for not providing a more meaningful error response.
   
The SPA504g emits this value with the following configuration:
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The SPA504g emits the correct value with the following configuration:
 
SIP > RTP Packet Size = 0.02
 
SIP > RTP Packet Size = 0.02

Latest revision as of 09:12, 25 October 2019

Configuring a Cisco SPA504g for use with IP-Phone-C

IP-Phone-C is a VOIP service provided by AsahiNet. It is a rebranding of NTT's Hikari VOIP service, so these notes may be applicable to similar services from other resellers.

This note describes how to configure a SPA504g VOIP handset with the IP-Phone-C service. This information should be applicable to other VOIP handsets, although their configuration details will vary.

Subscriber Information

Configure an extension on the handset as follows:

EXT > Display Name = <Your 050 Phone Number>
EXT > User Id = <Your 050 Phone Number>
EXT > Password = <Your Password>
EXT > Use Auth Id = Yes
EXT > Auth Id = <Your User Id>

Audio Configuration

The IP-Phone-C SIP server only supports the G711u codec. Other codecs, or too many proposed codecs, may trigger a "488 Not Acceptable Here" response. Therefore configuration with just G711u is recommended.

EXT > Preferred Codec = G711u
EXT > Use Pref Codec Only = Yes

NAT Configuration

If you are using NAT, enable keepalives:

EXT > NAT Keep Alive Enable = Yes

RTP Packet Size

The IP-Phone-C SIP server only accepts a SDP attribute a=ptime with the value 20. Any other value will result in a "488 Not Acceptable Here" response, incorporating a "307 Session description parameter not understood" warning. This is an extremely difficult issue to debug without a priori knowledge, and the engineer involved should be ashamed of themselves for not providing a more meaningful error response.

The SPA504g emits the correct value with the following configuration:

SIP > RTP Packet Size = 0.02