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setting_up_a_sip_phone_with_sdf_voip

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setting_up_a_sip_phone_with_sdf_voip [2022/11/16 20:37] – [General Instructions] hc9setting_up_a_sip_phone_with_sdf_voip [2024/12/27 05:16] (current) – in” hc9
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 SDF Voice over Internet Protocol Telephony is a service feature available to the [[http://sdf.org/?join|ARPA]], [[http://sdf.org/?join|META]] and Validated [[http://sdf.org/?join|USERS]] membership levels. It allows an SDF user to make voice and video calls to other SDF users and to linked Telephony services via a SIP client on a 3G or WIFI cellphone, WIFI enabled mobile device, or an Internetworked computer. SDF Voice over Internet Protocol Telephony is a service feature available to the [[http://sdf.org/?join|ARPA]], [[http://sdf.org/?join|META]] and Validated [[http://sdf.org/?join|USERS]] membership levels. It allows an SDF user to make voice and video calls to other SDF users and to linked Telephony services via a SIP client on a 3G or WIFI cellphone, WIFI enabled mobile device, or an Internetworked computer.
  
-SDF VoIP set up and server configuration can be accessed via 'maint' at the shell ('maint' → v → p).+SDF VoIP set up and server configuration can be accessed via ‘maint’ at the shell (‘maint’ → v → p).
  
 Below is a list of known-to-work [[#tested_sip_clients|sip clients]], and [[#general_instructions|general instructions]] on configuring and setting up a SIP client. There may also be [[sdf_voip_client|client-specific instructions]] as well. Below is a list of known-to-work [[#tested_sip_clients|sip clients]], and [[#general_instructions|general instructions]] on configuring and setting up a SIP client. There may also be [[sdf_voip_client|client-specific instructions]] as well.
  
-Wikipedia also has a list of notable [[http://en.wikipedia.org/wiki/List_of_SIP_software#SIP_clients|SIP clients]].+Wikipedia also has a list of notable [[wp>List_of_SIP_software#Clients|SIP clients]].
  
 ====== Tested SIP Hardware ====== ====== Tested SIP Hardware ======
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   * Grandstream DP750 with DP720 handset   * Grandstream DP750 with DP720 handset
   * snom 300   * snom 300
-  * [[http://www.cisco.com/en/US/products/hw/phones/ps379/ps1854/index.html|Cisco 7940G]] ([[:using_a_cisco_7940g_or_compatiable_phone_with_sdf_voip|Client Instructions]])+  * Cisco SPA-525G2 
 +  * [[http://www.cisco.com/en/US/products/hw/phones/ps379/ps1854/index.html|Cisco 7940G]] ([[using_a_cisco_7940g_or_compatiable_phone_with_sdf_voip|Client Instructions]])
  
-==== Analog Phone Adaptor ====+==== Analog Phone Adapter ====
  
   * [[http://www.grandstream.com/products/ht_series/ht286/ht286.html|Grandstream HandyTone 286]]   * [[http://www.grandstream.com/products/ht_series/ht286/ht286.html|Grandstream HandyTone 286]]
-  * Grandstream HT801+  * Grandstream HT801/HT802 
 +  * Linksys/Sipura SPA-3000 
 +  * Obihai/Polycom OBi200
  
 ===== Tested SIP clients ===== ===== Tested SIP clients =====
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   - extension   - extension
-    - The "username" used for authentication. Currently, the **extension is a series of numbers** such as "6000". A [[http://sdf.org/?tutorials/sdf_voip_ext|directory]] of users is also available. The assigned extension will be given in an email. Do not use your SDF username.+    - The “username” used for authentication. 
 +    - Currently, the **extension is a series of numbers** such as “6000”. A [[http://sdf.org/?tutorials/sdf_voip_ext|directory]] of users is also available. 
 +    - The assigned extension will be given in an email. 
 +    - Do not use your SDF username.
   - domain   - domain
-    - This may also be known as the server to connect with. Use **sip.sdf.org** as the server; use the default port of **5060/udp**. Please note: some clients infer the domain from the extension. In this case, the extension will be in the form of "extension@domain", or "sip:extension@domain"; eg, "6000@sip.sdf.org" would be a valid username that includes the domain.+    - This may also be known as the server to connect with. Use **sip.sdf.org** as the server; use the default port of **5060/udp**. 
 +    - Please note: some clients infer the domain from the extension. In this case, the extension will be in the form of “extension@domain”, or “sip:extension@domain”; eg, “6000@sip.sdf.org” would be a valid username that includes the domain.
   - password   - password
-    - This is the password used to authenticate to the SIP server. This will be provided in the email as well, but may change very soon in the future. The password can be reset via the maint command.+    - This is the password used to authenticate to the SIP server. This will be provided in the email as well, but may change very soon in the future. 
 +    - The password can be reset via the maint command.
  
 ==== Client Instructions ==== ==== Client Instructions ====
  
-Please see the [[:sdf_voip_client|client instructions]] page for specific client instructions.+Please see the [[sdf_voip_client|client instructions]] page for specific client instructions.
  
 ==== Voicemail ==== ==== Voicemail ====
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 ==== PSTN ==== ==== PSTN ====
  
-Want to call the public switched telephone network? Read the info [[http://sdf.org/?tutorials/voip_pstn|here]]+Want to call the public switched telephone network? Read the info [[http://sdf.org/?tutorials/voip_pstn|here]] ([[voip_pstn|wiki copy of this page]])
  
 This tutorial is far from complete. Wanna make it better? Edit it! This tutorial is far from complete. Wanna make it better? Edit it!
setting_up_a_sip_phone_with_sdf_voip.1668631035.txt.gz · Last modified: 2022/11/16 20:37 by hc9