WikiPhone:Phone Access:Dial in via SIP: Difference between revisions

No edit summary
No edit summary
Line 42: Line 42:
** Before you can dial URIs without a SIP account, you will need to go to MicroSIP settings and check "Enable Local Account".   
** Before you can dial URIs without a SIP account, you will need to go to MicroSIP settings and check "Enable Local Account".   
** Written for Windows, can be used on Unix-like systems with WineHQ.
** Written for Windows, can be used on Unix-like systems with WineHQ.
== Configuring a trunk in Asterisk ==
You can configure an Asterisk switch or PBX so that it can reach a WikiPhone service over SIP. We only provide copy-and-paste templates for Chan_PJSIP. Chan_SIP is deprecated and should not be used anymore.
To configure WikiPhone as an available SIP endpoint in pjsip.conf:
<pre>
[WikiPhone]
type=endpoint
transport=transport-udp-5060 ;This is the standard transport name for UDP PJSIP. If yours is diffrent, you will want to replace this.
disallow=all
allow=ulaw,g722
dtmf_mode=rfc4733
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=no
context=default
aors=WikiPhone
[WikiPhone]
type=aor
contact=sip:telephones.wiki:5060 ;If you don't have DNS, replace with 135.148.32.154
[WikiPhone-identify]
type=identify
endpoint=WikiPhone
match=telephones.wiki ;If you don't have DNS, replace with 135.148.32.154
</pre>
Then, in your extensions.conf, where your dialplan is, we can now configure dialing out to WikiPhone. For example:
<pre>
[from-internal]
exten => #200,1,NoOp()
  same => n,Dial(PJSIP/articles@WikiPhone,5)
  same => n,Hangup()
</pre>
In this example, when we dial #200 on our phone, we will reach the articles by telephone service. You can configure this however you wish. If you have Asterisk, you should already be familiar with dialplans.


== Configuring a trunk in CUCM ==
== Configuring a trunk in CUCM ==