Difference between revisions of "TN1"
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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 == |
== Subscriber Information == |
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| + | Configure an extension on the handset as follows: |
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| − | EXT > Display Name = <Phone Number> |
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| − | EXT > |
+ | EXT > Display Name = <Your 050 Phone Number> |
| − | EXT > |
+ | EXT > User Id = <Your 050 Phone Number> |
| + | EXT > Password = <Your Password> |
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EXT > Use Auth Id = Yes |
EXT > Use Auth Id = Yes |
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| − | EXT > Auth Id = <User Id> |
+ | 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 |
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== RTP Packet Size == |
== RTP Packet Size == |
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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. |
+ | 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: |
The SPA504g emits the correct value with the following configuration: |
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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