Differences

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--- setting_up_a_sip_phone_with_sdf_voip [2022/11/16 20:42] – [Analog Phone Adaptor] hc9
+++ setting_up_a_sip_phone_with_sdf_voip [2024/12/27 05:16] (current) – in” hc9
@@ Line 3: / Line 3: @@
 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.
-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 ======
@@ Line 20: / Line 20: @@
   * Grandstream DP750 with DP720 handset
   * 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 Adapter ====
   * [[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 =====
@@ Line 68: / Line 70: @@
   - extension
-    - The "username" used for authentication.
+    - 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.
+    - 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
     - 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.
+    - 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
     - 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.
@@ Line 81: / Line 83: @@
 ==== 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 ====
@@ Line 89: / Line 91: @@
 ==== 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!