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

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You can access various public WikiPhone services on any SIP client, phone, or PBX (via trunk). This can done in the form of entering a "SIP URI" on your client. Please note that you cannot reach WikiPhone staff members or departments over public SIP. You will need to either use the PSTN or a ~~collecters~~ network. For each service, we provide both a alphabetic and numeric URI so that if your device does not allow the use of ~~letters~~, you can still reach each service. Authentication is not ~~requried~~. The standard SIP port (5060) is used for all ~~URI's~~.

You can access various public WikiPhone services on any SIP client, phone, or PBX (via trunk). This can done in the form of entering a "SIP URI" on your client. For example, you can reach the WikiPhone Updates Hotline at sip:updates@telephones.wiki. Please note that you cannot reach WikiPhone staff members or departments over public SIP. You will need to either use the PSTN or a collectors network. For each service, we provide both an alphabetic and a numeric URI so that if your device does not allow the use of alphabetic dialing or if it's inconvenient, you can still reach each service using the numeric URI. Numeric URIs are the same as the code to reach the same services on telephone collectors' networks. Authentication or registration is not required. The standard SIP port (5060) is used for all URIs.

== Available Services ==

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|-

|WikiPhone Updates Hotline

|updates~~@telephones.wiki~~

|updates

|0200

|-

|Jordan U Ambient ~~Tempature~~ Line

|Jordan U Ambient Temperature Line

|jordanu~~@telephones.wiki~~

|jordanu

|0250

|-

|~~Listen to articles~~

|Articles via telephone

|articles~~@telephones.wiki~~

|articles

|0201

|}

== Available Codecs ==

Our system will accept the following codecs for SIP calls:

* g711 ulaw

* g711 alaw

* g722 HD

At this time, we do not support other codecs as these codecs are supported by nearly every SIP client; as such, we don't deem it necessary to test and support other codecs.

== Dialing URI's ==

To dial a URI on a SIP client, you will need to enter <uri>@telephones.wiki or if you don't have DNS, <uri>@135.148.32.154 into your SIP client. Example: articles@telephone.wiki or Some clients will require sip: before the URI. Some will also attempt to dial using the active SIP account if you are connected to a SIP server such as a PBX. Some will require you to enable local accounts or URI dialing. For help, find the manual or documentation for your SIP device/client.

== Softphone Clients ==

These are some softphone clients that let you dial SIP URIs without a SIP account and are commonly used.

* MicroSIP

** A commonly used SIP client that can be used to dial SIP URIs.

** Before you can dial URIs without a SIP account, you will need to go to MicroSIP settings and check "Enable Local Account".

== 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 ==

You can configure WikiPhone as a trunk in CUCM so that you can reach a WikiPhone service by dialing a DN on your phones. Since WikiPhone does not ~~requrie auth~~ or registration, this can be done without an external border element.

You can configure WikiPhone as a trunk in CUCM so that you can reach a WikiPhone service by dialing a DN on your phones. Since WikiPhone does not require authentication or registration, this can be done without an external border element as long as your CUCM server has access to the internet.

To configure WikiPhone as a trunk, go to Device -> Trunk and choose SIP as the trunk type. Name the trunk WikiPhone. Set your device pool, and set the SIP Trunk Security Profile as "Non Secure SIP Trunk Profile". And set the SIP profile to "Standard SIP Profile". Set the Destination Address to "telephones.wiki" or if you don't have DNS, you can use our static IPv4 which is "135.148.32.154". And then save the trunk and click reset.

Now you can go to Call Routing > Route/Hunt > Route Pattern and create a new pattern. Set the call pattern to what you want for the ~~perticler~~ service. And set Gateway/Route List to your WikiPhone trunk. Then under Called Party Transformations, enter the numeric ~~number~~ of the service you want this DN to route to (not the full URI). And click save. Repeat this step for every service you wanna be able to access.

Now you can go to Call Routing > Route/Hunt > Route Pattern and create a new pattern. Set the call pattern to what you want for the particular service. And set Gateway/Route List to your WikiPhone trunk. Then, under Called Party Transformations, enter the numeric URI of the service you want this DN to route to (not the full URI). And click save. Repeat this step for every service you wanna be able to access.

Once you are done, you should now be able to dial the DN you configured to reach the particular service of your choice. If you cannot hear audio, make sure your codecs are setup correctly, and that calls are being routed properly over NAT. If your call is declined, ensure you used a vaild number, and that the IP/FQDN is correct.

@@ Line 1: / Line 1: @@
-You can access various public WikiPhone services on any SIP client, phone, or PBX (via trunk). This can done in the form of entering a "SIP URI" on your client. Please note that you cannot reach WikiPhone staff members or departments over public SIP. You will need to either use the PSTN or a collecters network. For each service, we provide both a alphabetic and numeric URI so that if your device does not allow the use of letters, you can still reach each service. Authentication is not requried. The standard SIP port (5060) is used for all URI's.
+You can access various public WikiPhone services on any SIP client, phone, or PBX (via trunk). This can done in the form of entering a "SIP URI" on your client. For example, you can reach the WikiPhone Updates Hotline at sip:updates@telephones.wiki. Please note that you cannot reach WikiPhone staff members or departments over public SIP. You will need to either use the PSTN or a collectors network. For each service, we provide both an alphabetic and a numeric URI so that if your device does not allow the use of alphabetic dialing or if it's inconvenient, you can still reach each service using the numeric URI. Numeric URIs are the same as the code to reach the same services on telephone collectors' networks. Authentication or registration is not required. The standard SIP port (5060) is used for all URIs.
 == Available Services ==
@@ Line 9: / Line 10: @@
 |-
 |WikiPhone Updates Hotline
-|updates@telephones.wiki
+|updates
 |0200
 |-
 |-
-|Jordan U Ambient Tempature Line
+|Jordan U Ambient Temperature Line
-|jordanu@telephones.wiki
+|jordanu
 |0250
 |-
-|Listen to articles
+|Articles via telephone
-|articles@telephones.wiki
+|articles
 |0201
 |}
+== Available Codecs ==
+Our system will accept the following codecs for SIP calls:
+* g711 ulaw
+* g711 alaw
+* g722 HD
+At this time, we do not support other codecs as these codecs are supported by nearly every SIP client; as such, we don't deem it necessary to test and support other codecs.
+== Dialing URI's ==
+To dial a URI on a SIP client, you will need to enter <uri>@telephones.wiki or if you don't have DNS, <uri>@135.148.32.154 into your SIP client. Example: articles@telephone.wiki or Some clients will require sip: before the URI. Some will also attempt to dial using the active SIP account if you are connected to a SIP server such as a PBX. Some will require you to enable local accounts or URI dialing. For help, find the manual or documentation for your SIP device/client.
+== Softphone Clients ==
+These are some softphone clients that let you dial SIP URIs without a SIP account and are commonly used.
+* MicroSIP
+** A commonly used SIP client that can be used to dial SIP URIs.
+** Before you can dial URIs without a SIP account, you will need to go to MicroSIP settings and check "Enable Local Account".
+== 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 ==
-You can configure WikiPhone as a trunk in CUCM so that you can reach a WikiPhone service by dialing a DN on your phones. Since WikiPhone does not requrie auth or registration, this can be done without an external border element.
+You can configure WikiPhone as a trunk in CUCM so that you can reach a WikiPhone service by dialing a DN on your phones. Since WikiPhone does not require authentication or registration, this can be done without an external border element as long as your CUCM server has access to the internet.
 To configure WikiPhone as a trunk, go to Device -> Trunk and choose SIP as the trunk type. Name the trunk WikiPhone. Set your device pool, and set the SIP Trunk Security Profile as "Non Secure SIP Trunk Profile". And set the SIP profile to "Standard SIP Profile". Set the Destination Address to "telephones.wiki" or if you don't have DNS, you can use our static IPv4 which is "135.148.32.154". And then save the trunk and click reset.
-Now you can go to Call Routing > Route/Hunt > Route Pattern and create a new pattern. Set the call pattern to what you want for the perticler service. And set Gateway/Route List to your WikiPhone trunk. Then under Called Party Transformations, enter the numeric number of the service you want this DN to route to (not the full URI). And click save. Repeat this step for every service you wanna be able to access.
+Now you can go to Call Routing > Route/Hunt > Route Pattern and create a new pattern. Set the call pattern to what you want for the particular service. And set Gateway/Route List to your WikiPhone trunk. Then, under Called Party Transformations, enter the numeric URI of the service you want this DN to route to (not the full URI). And click save. Repeat this step for every service you wanna be able to access.
 Once you are done, you should now be able to dial the DN you configured to reach the particular service of your choice. If you cannot hear audio, make sure your codecs are setup correctly, and that calls are being routed properly over NAT. If your call is declined, ensure you used a vaild number, and that the IP/FQDN is correct.