Differences

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--- setting_up_a_sip_phone_with_sdf_voip [2021/03/11 03:45] – [Setting up a SIP phone with SDF VoIP] hc9
+++ setting_up_a_sip_phone_with_sdf_voip [2023/07/03 20:01] (current) – [Setting up a SIP phone with SDF VoIP] 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.
+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]] ([[http://sdf.org/?tutorials/sdf_voip_7940g|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]]
-  * Grandstream HT801
+  * Grandstream HT801/HT802
+  * Linksys/Sipura SPA-3000
+  * Obihai/Polycom OBi200
 ===== Tested SIP clients =====
@@ Line 34: / Line 37: @@
   * [[https://www.counterpath.com/plan-select-solo/|Bria Solo]] (formerly X-lite) Windows, Mac OS X - Desktop Client
   * [[http://icanblink.com|Blink]] Windows, Mac OS X, Linux - Desktop Client
-  * [[http://ekiga.net|Ekiga]] Windows and Linux - Supports Video
+  * [[https://ekiga.org|Ekiga]] Windows and Linux - Supports Video
 ==== Android ====
@@ Line 43: / Line 46: @@
 ==== iPhone/iPad/iTouchU ====
-  * [[http://www.fring.com/default.php|Fring]] on the Apple iPhone, iPod, iPad
+  * [[https://web.archive.org/web/20110226194609/http://www.fring.com/default.php|Fring]] on the Apple iPhone, iPod, iPad
   * [[https://www.acrobits.net/sip-client-ios-android/|Acrobits Groundwire]]
@@ Line 66: / Line 69: @@
 All SIP clients must have at least 3 bits of information:
-extensionThe "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.domainThis 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.passwordThis 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.
+  - 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.
+  - 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
+    - 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 ====
-Please see the [[http://sdf.org/?tutorials/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 84: / Line 97: @@
 ----
-$Id: sdf_voip.html,v 1.43 2019/12/13 03:39:56 jasmaz Exp $ [[http://sdf.org/?tutorials/sdf_voip|Setting up a SIP phone with SDF VoIP]] - legacy link
+$Id: sdf_voip.html,v 1.43 2019/12/13 03:39:56 jasmaz Exp $ [[http://sdf.org/?tutorials/sdf_voip|Setting up a SIP phone with SDF VoIP]] - traditional link (using [[wp>Revision_Control_System|RCS]])