setting_up_a_sip_phone_with_sdf_voip
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setting_up_a_sip_phone_with_sdf_voip [2023/07/03 20:01] – [Setting up a SIP phone with SDF VoIP] hc9 | setting_up_a_sip_phone_with_sdf_voip [2024/12/27 05:16] (current) – in” hc9 | ||
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+ | ====== Setting up a SIP phone with SDF VoIP ====== | ||
+ | |||
+ | SDF Voice over Internet Protocol Telephony is a service feature available to the [[http:// | ||
+ | |||
+ | 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 [[# | ||
+ | |||
+ | Wikipedia also has a list of notable [[wp> | ||
+ | |||
+ | ====== Tested SIP Hardware ====== | ||
+ | |||
+ | == Routers (with SIP capabilities) == | ||
+ | |||
+ | * [[http:// | ||
+ | |||
+ | ==== Desktop ==== | ||
+ | |||
+ | * Grandstream GXP2000 | ||
+ | * Grandstream DP750 with DP720 handset | ||
+ | * snom 300 | ||
+ | * Cisco SPA-525G2 | ||
+ | * [[http:// | ||
+ | |||
+ | ==== Analog Phone Adapter ==== | ||
+ | |||
+ | * [[http:// | ||
+ | * Grandstream HT801/HT802 | ||
+ | * Linksys/ | ||
+ | * Obihai/ | ||
+ | |||
+ | ===== Tested SIP clients ===== | ||
+ | |||
+ | ==== Cross-platform ==== | ||
+ | |||
+ | * [[http:// | ||
+ | * [[https:// | ||
+ | * [[http:// | ||
+ | * [[https:// | ||
+ | |||
+ | ==== Android ==== | ||
+ | |||
+ | * CSipSimple | ||
+ | * Unlocked 2.3 (Gingerbread), | ||
+ | |||
+ | ==== iPhone/ | ||
+ | |||
+ | * [[https:// | ||
+ | * [[https:// | ||
+ | |||
+ | ==== Nokia ==== | ||
+ | |||
+ | * Built-in client on Nokia phones and tablets (N810) | ||
+ | |||
+ | ==== Mac OS X ==== | ||
+ | |||
+ | * [[http:// | ||
+ | |||
+ | ==== Unix-ish ==== | ||
+ | |||
+ | * XMeeting | ||
+ | |||
+ | ==== Windows ==== | ||
+ | |||
+ | * [[https:// | ||
+ | |||
+ | ===== General Instructions ===== | ||
+ | |||
+ | All SIP clients must have at least 3 bits of information: | ||
+ | |||
+ | - extension | ||
+ | - The “username” used for authentication. | ||
+ | - Currently, the **extension is a series of numbers** such as “6000”. A [[http:// | ||
+ | - 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/ | ||
+ | - Please note: some clients infer the domain from the extension. In this case, the extension will be in the form of “extension@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 [[sdf_voip_client|client instructions]] page for specific client instructions. | ||
+ | |||
+ | ==== Voicemail ==== | ||
+ | |||
+ | Voicemail messages will be automatically mailed to your SDF email address in WAV format. | ||
+ | |||
+ | ==== PSTN ==== | ||
+ | |||
+ | Want to call the public switched telephone network? Read the info [[http:// | ||
+ | |||
+ | This tutorial is far from complete. Wanna make it better? Edit it! | ||
+ | |||
+ | ---- | ||
+ | |||
+ | $Id: sdf_voip.html, | ||