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TROUBLESHOOTING
Troubleshooting. We have listed here a collection of the most common issues we hear about and our suggested solutions. Also be sure to check the recommended links at the bottom of the page for device specific issues: I am experiencing echo on my OnSIP line. What can Ido to fix it?
SERVICES AND SERVERS OVERVIEW OnSIP provides two VOIP servers. Each server provides termination, orgination, and registration services. Calls originate from a server with which a customer is registered. Customers may utilize more than one server. sip.jnctn.net. provides for "trunked" SIP call delivery. username/password used for termination and registrationauthentication.
3-WAY CONFERENCE CALLING 3-Way Conference Calling. Note: This feature is currently only available when using Firefox as your browser. Once logged into app.onsip.com (in Firefox), start a voice call. Once the call is active, click on the “+” sign button (the 'Conference' button) on the right side of the call handling options. Enter the phone number,extension, or
VOICEMAIL REFERENCE GUIDE To access the voicemail manager, dial *98 on your OnSIP registered phone or device. Next, enter your mailbox number, followed by your mailbox password. (Tip: Press # after your mailbox number and password to speed up the process.) If you do not know your voicemail password, please contact your OnSIP Account Administrator.PHONE NUMBERS
click Create New Resource. select Phone Number - there's an indication on this page if you have enough funds to buy a phone number as well as how much each phone number will cost. select Create a New Phone Number. from the dropdown select the Area Code (first 3 digits) from the dropdown select the Rate Center (next 3 digits) GRANDSTREAM DIGIT MAP / DIAL PLAN User can dial 92125551234 and the string will respond with 12125551234. User can dial 111 (1xx) or 222 (2xx) and phone will allow it. User can dial any number starting with an asterisk (*) User can dial *xx*xxxxxx. Thanks Chuck! From Grandstream. The dial-plan will restrict the number dialed.SONICWALL FIREWALL
Sonicwall Firewall - SIP Transformations. *if this does not resolve port timeout issues, may need to also modify the Global UDP Connection Timeout: Advanced tab = Firewall => Access Rules => LAN/WAN and increase UDP to 30 to override any inherited UDP timeout rules. o Turn on Consistent NAT. o The SIP Transformations sections should beDISABLED
BLOCK/BLACKLIST CALLS ON POLYCOM PHONES Click on the name of the unwanted call. Click on the pen/pencil icon on the right to edit. Scroll down to Auto Reject. Select "Enable", then "Save" (on left) Upper left depress the arrow to go back to Directory and select another number or exit the page. Note: A blacklisted call will not be disconnected. RESET PANASONIC PHONES TO FACTORY DEFAULT Go to the "Maintenance" tab, Click on "Reset to Defaults" in the left column, and Click on "Reset to carrier defaults". Click "Restart". Be sure to close all open browser windows before logging back into the phone. Also, be sure to reset the key in the phone resource in admin.onsip.com. Follow these instructions- Panasonic Boot Server. BLOCK/BLACKLIST CALLS ON GRANDSTREAM PHONES Block/Blacklist Calls on Grandstream Phones - unsupported. Please see attached PDF for instructions to blacklist inbound calls by number on certain Grandstream phones. The best bet is to search for "Grandstream blacklist" and your phone model number. Note: A blacklisted call will not be disconnected. The phone will respond with "Busy Here" asTROUBLESHOOTING
Troubleshooting. We have listed here a collection of the most common issues we hear about and our suggested solutions. Also be sure to check the recommended links at the bottom of the page for device specific issues: I am experiencing echo on my OnSIP line. What can Ido to fix it?
SERVICES AND SERVERS OVERVIEW OnSIP provides two VOIP servers. Each server provides termination, orgination, and registration services. Calls originate from a server with which a customer is registered. Customers may utilize more than one server. sip.jnctn.net. provides for "trunked" SIP call delivery. username/password used for termination and registrationauthentication.
3-WAY CONFERENCE CALLING 3-Way Conference Calling. Note: This feature is currently only available when using Firefox as your browser. Once logged into app.onsip.com (in Firefox), start a voice call. Once the call is active, click on the “+” sign button (the 'Conference' button) on the right side of the call handling options. Enter the phone number,extension, or
VOICEMAIL REFERENCE GUIDE To access the voicemail manager, dial *98 on your OnSIP registered phone or device. Next, enter your mailbox number, followed by your mailbox password. (Tip: Press # after your mailbox number and password to speed up the process.) If you do not know your voicemail password, please contact your OnSIP Account Administrator.PHONE NUMBERS
click Create New Resource. select Phone Number - there's an indication on this page if you have enough funds to buy a phone number as well as how much each phone number will cost. select Create a New Phone Number. from the dropdown select the Area Code (first 3 digits) from the dropdown select the Rate Center (next 3 digits) GRANDSTREAM DIGIT MAP / DIAL PLAN User can dial 92125551234 and the string will respond with 12125551234. User can dial 111 (1xx) or 222 (2xx) and phone will allow it. User can dial any number starting with an asterisk (*) User can dial *xx*xxxxxx. Thanks Chuck! From Grandstream. The dial-plan will restrict the number dialed.SONICWALL FIREWALL
Sonicwall Firewall - SIP Transformations. *if this does not resolve port timeout issues, may need to also modify the Global UDP Connection Timeout: Advanced tab = Firewall => Access Rules => LAN/WAN and increase UDP to 30 to override any inherited UDP timeout rules. o Turn on Consistent NAT. o The SIP Transformations sections should beDISABLED
BLOCK/BLACKLIST CALLS ON POLYCOM PHONES Click on the name of the unwanted call. Click on the pen/pencil icon on the right to edit. Scroll down to Auto Reject. Select "Enable", then "Save" (on left) Upper left depress the arrow to go back to Directory and select another number or exit the page. Note: A blacklisted call will not be disconnected. RESET PANASONIC PHONES TO FACTORY DEFAULT Go to the "Maintenance" tab, Click on "Reset to Defaults" in the left column, and Click on "Reset to carrier defaults". Click "Restart". Be sure to close all open browser windows before logging back into the phone. Also, be sure to reset the key in the phone resource in admin.onsip.com. Follow these instructions- Panasonic Boot Server. BLOCK/BLACKLIST CALLS ON GRANDSTREAM PHONES Block/Blacklist Calls on Grandstream Phones - unsupported. Please see attached PDF for instructions to blacklist inbound calls by number on certain Grandstream phones. The best bet is to search for "Grandstream blacklist" and your phone model number. Note: A blacklisted call will not be disconnected. The phone will respond with "Busy Here" as TROUBLESHOOTING ONE-WAY AUDIO To see if your phones/firewall are sending OnSIP the proper packets, log into the Onsip interface and click on 'Users'. Click on the name of one of the users to see the User Detail. Scroll down to the 'Maintenance' block and click on ' (Show Details)' next to the SIP Registrations line. You should see a 'Contact' address like:sip:username@192
RESET PANASONIC PHONES TO FACTORY DEFAULT Go to the "Maintenance" tab, Click on "Reset to Defaults" in the left column, and Click on "Reset to carrier defaults". Click "Restart". Be sure to close all open browser windows before logging back into the phone. Also, be sure to reset the key in the phone resource in admin.onsip.com. Follow these instructions- Panasonic Boot Server. SMS / TEXT MESSAGING SMS / Text Messaging. All OnSIP phone numbers (DIDs) are enabled for SMS (texting), but OnSIP does not directly provide SMS services. Here are a few third-party vendors that may be able to help; there are many more on the Internet. Phonewire (Disclaimer: OnSIP as a FREEPBX CONFIGURATION FOR ONSIP TRUNKING Step 2: Add the OnSIP Trunking user as a SIP Trunk in FreePBX. Log in to the FreePBX Admin page. Click on "Trunks", under the "Connectivity" drop down menu at the top. Click on "Add SIP Trunk". Under the General Settings section. Complete the following: Trunk Name: OnSIP. Outbound CallerID: 15135555555. ASTERISK CONFIGURATION Configuring SIP. Go to https://admin.onsip.com and login. Go to the Configuration tab and note your VOIP username and password. Edit /etc/asterisk/sip.conf, replacing MY_USERNAME and MY_PASSWORD in the "register => " statement below with your VOIP username and password. ; ; The "general" context should already exist in sip.conf ; Add a lineto
MIKROTIK SIP ALG = SIP HELPER Mikrotik SIP ALG = SIP Helper. Created September 2015. Mikrotik SIP ALG is called a SIP Helper and is located under /IP>Firewall>Service ports. To disable, run this command from the terminal: /ip firewall service-port disable sip. Or from winbox just navigate to IP>Firewall and then click on the Service Ports tab and disable it through theGUI.
SNOM 300/320/360/370 Snom 320 Review. The Snom 320 (firmware 320 v7.3.30) is a 2-year old SIP business phone, and the middle tier in the company’s 300 series. It has more features than the Snom 300 including additional line displays and a better semi-graphic LCD, but at a considerable priceincrease.
CISCO 79XX – ONSIP SUPPORT Note: While Cisco 79xx phones are compatible with the OnSIP service, we recommend reselling your Cisco phones on eBay and using the funds for new Polycoms. The benefits of using Polycoms over Cisco 79xx include: Music on Hold. High Definition (HD) audio. Boot server and asset tracking via OnSIP WWW site. Automated configuration.LINPHONE FOR IPHONE
Step 6: Configure Network Settings. Select "Network" tab from Settings page. Turn 'off' ICE, 'Allow IPv6' and 'Edge optimization' settings - see image below: Once done depress the left arrow to go back to Settings page.Your registration WILL FAIL - you're almost done. NAT AND FIREWALL TRAVERSAL RECOMMENDATION Setting the UDP port timeout to anything between 45 and 120 seconds will alleviate that issue. VOIP => Settings: Turn on Consistent NAT. Disable SIP ALG (may say SIP Helper, depends on the make/model) Consistent NAT helps the device to have the same external port opened every time it connects. In this way, if the UDP port does timeout, thenext
TROUBLESHOOTING
Troubleshooting. We have listed here a collection of the most common issues we hear about and our suggested solutions. Also be sure to check the recommended links at the bottom of the page for device specific issues: I am experiencing echo on my OnSIP line. What can Ido to fix it?
SERVICES AND SERVERS OVERVIEW OnSIP provides two VOIP servers. Each server provides termination, orgination, and registration services. Calls originate from a server with which a customer is registered. Customers may utilize more than one server. sip.jnctn.net. provides for "trunked" SIP call delivery. username/password used for termination and registrationauthentication.
TROUBLESHOOTING ONE-WAY AUDIO To see if your phones/firewall are sending OnSIP the proper packets, log into the Onsip interface and click on 'Users'. Click on the name of one of the users to see the User Detail. Scroll down to the 'Maintenance' block and click on ' (Show Details)' next to the SIP Registrations line. You should see a 'Contact' address like:sip:username@192
3-WAY CONFERENCE CALLING 3-Way Conference Calling. Note: This feature is currently only available when using Firefox as your browser. Once logged into app.onsip.com (in Firefox), start a voice call. Once the call is active, click on the “+” sign button (the 'Conference' button) on the right side of the call handling options. Enter the phone number,extension, or
SMS / TEXT MESSAGING SMS / Text Messaging. All OnSIP phone numbers (DIDs) are enabled for SMS (texting), but OnSIP does not directly provide SMS services. Here are a few third-party vendors that may be able to help; there are many more on the Internet. Phonewire (Disclaimer: OnSIP as a VOICEMAIL REFERENCE GUIDE To access the voicemail manager, dial *98 on your OnSIP registered phone or device. Next, enter your mailbox number, followed by your mailbox password. (Tip: Press # after your mailbox number and password to speed up the process.) If you do not know your voicemail password, please contact your OnSIP Account Administrator.PHONE NUMBERS
click Create New Resource. select Phone Number - there's an indication on this page if you have enough funds to buy a phone number as well as how much each phone number will cost. select Create a New Phone Number. from the dropdown select the Area Code (first 3 digits) from the dropdown select the Rate Center (next 3 digits) ASTERISK CONFIGURATION Configuring SIP. Go to https://admin.onsip.com and login. Go to the Configuration tab and note your VOIP username and password. Edit /etc/asterisk/sip.conf, replacing MY_USERNAME and MY_PASSWORD in the "register => " statement below with your VOIP username and password. ; ; The "general" context should already exist in sip.conf ; Add a lineto
GRANDSTREAM DIGIT MAP / DIAL PLAN User can dial 92125551234 and the string will respond with 12125551234. User can dial 111 (1xx) or 222 (2xx) and phone will allow it. User can dial any number starting with an asterisk (*) User can dial *xx*xxxxxx. Thanks Chuck! From Grandstream. The dial-plan will restrict the number dialed. BLOCK/BLACKLIST CALLS ON POLYCOM PHONES Click on the name of the unwanted call. Click on the pen/pencil icon on the right to edit. Scroll down to Auto Reject. Select "Enable", then "Save" (on left) Upper left depress the arrow to go back to Directory and select another number or exit the page. Note: A blacklisted call will not be disconnected.TROUBLESHOOTING
Troubleshooting. We have listed here a collection of the most common issues we hear about and our suggested solutions. Also be sure to check the recommended links at the bottom of the page for device specific issues: I am experiencing echo on my OnSIP line. What can Ido to fix it?
SERVICES AND SERVERS OVERVIEW OnSIP provides two VOIP servers. Each server provides termination, orgination, and registration services. Calls originate from a server with which a customer is registered. Customers may utilize more than one server. sip.jnctn.net. provides for "trunked" SIP call delivery. username/password used for termination and registrationauthentication.
TROUBLESHOOTING ONE-WAY AUDIO To see if your phones/firewall are sending OnSIP the proper packets, log into the Onsip interface and click on 'Users'. Click on the name of one of the users to see the User Detail. Scroll down to the 'Maintenance' block and click on ' (Show Details)' next to the SIP Registrations line. You should see a 'Contact' address like:sip:username@192
3-WAY CONFERENCE CALLING 3-Way Conference Calling. Note: This feature is currently only available when using Firefox as your browser. Once logged into app.onsip.com (in Firefox), start a voice call. Once the call is active, click on the “+” sign button (the 'Conference' button) on the right side of the call handling options. Enter the phone number,extension, or
SMS / TEXT MESSAGING SMS / Text Messaging. All OnSIP phone numbers (DIDs) are enabled for SMS (texting), but OnSIP does not directly provide SMS services. Here are a few third-party vendors that may be able to help; there are many more on the Internet. Phonewire (Disclaimer: OnSIP as a VOICEMAIL REFERENCE GUIDE To access the voicemail manager, dial *98 on your OnSIP registered phone or device. Next, enter your mailbox number, followed by your mailbox password. (Tip: Press # after your mailbox number and password to speed up the process.) If you do not know your voicemail password, please contact your OnSIP Account Administrator.PHONE NUMBERS
click Create New Resource. select Phone Number - there's an indication on this page if you have enough funds to buy a phone number as well as how much each phone number will cost. select Create a New Phone Number. from the dropdown select the Area Code (first 3 digits) from the dropdown select the Rate Center (next 3 digits) ASTERISK CONFIGURATION Configuring SIP. Go to https://admin.onsip.com and login. Go to the Configuration tab and note your VOIP username and password. Edit /etc/asterisk/sip.conf, replacing MY_USERNAME and MY_PASSWORD in the "register => " statement below with your VOIP username and password. ; ; The "general" context should already exist in sip.conf ; Add a lineto
GRANDSTREAM DIGIT MAP / DIAL PLAN User can dial 92125551234 and the string will respond with 12125551234. User can dial 111 (1xx) or 222 (2xx) and phone will allow it. User can dial any number starting with an asterisk (*) User can dial *xx*xxxxxx. Thanks Chuck! From Grandstream. The dial-plan will restrict the number dialed. BLOCK/BLACKLIST CALLS ON POLYCOM PHONES Click on the name of the unwanted call. Click on the pen/pencil icon on the right to edit. Scroll down to Auto Reject. Select "Enable", then "Save" (on left) Upper left depress the arrow to go back to Directory and select another number or exit the page. Note: A blacklisted call will not be disconnected. SERVICES AND SERVERS OVERVIEW OnSIP provides two VOIP servers. Each server provides termination, orgination, and registration services. Calls originate from a server with which a customer is registered. Customers may utilize more than one server. sip.jnctn.net. provides for "trunked" SIP call delivery. username/password used for termination and registrationauthentication.
TROUBLESHOOTING ONE-WAY AUDIO To see if your phones/firewall are sending OnSIP the proper packets, log into the Onsip interface and click on 'Users'. Click on the name of one of the users to see the User Detail. Scroll down to the 'Maintenance' block and click on ' (Show Details)' next to the SIP Registrations line. You should see a 'Contact' address like:sip:username@192
ASTERISK CONFIGURATION Configuring SIP. Go to https://admin.onsip.com and login. Go to the Configuration tab and note your VOIP username and password. Edit /etc/asterisk/sip.conf, replacing MY_USERNAME and MY_PASSWORD in the "register => " statement below with your VOIP username and password. ; ; The "general" context should already exist in sip.conf ; Add a lineto
FREEPBX CONFIGURATION FOR ONSIP TRUNKING Step 2: Add the OnSIP Trunking user as a SIP Trunk in FreePBX. Log in to the FreePBX Admin page. Click on "Trunks", under the "Connectivity" drop down menu at the top. Click on "Add SIP Trunk". Under the General Settings section. Complete the following: Trunk Name: OnSIP. Outbound CallerID: 15135555555. POLYCOM REALPRESENCE TRIO 8800 Admin login: Username > Polycom (note capital "P") is Password > 456. **OnSIP strongly encourages customers to change this universal password after the phone has been registered**. Step 3. Enter your User information from Step 1. Select Settings, then Lines, then Line 1 Identification: "Display Name" > whatever you would like callers tosee.
SONICWALL FIREWALL
Sonicwall Firewall - SIP Transformations. *if this does not resolve port timeout issues, may need to also modify the Global UDP Connection Timeout: Advanced tab = Firewall => Access Rules => LAN/WAN and increase UDP to 30 to override any inherited UDP timeout rules. o Turn on Consistent NAT. o The SIP Transformations sections should beDISABLED
RESET PANASONIC PHONES TO FACTORY DEFAULT Go to the "Maintenance" tab, Click on "Reset to Defaults" in the left column, and Click on "Reset to carrier defaults". Click "Restart". Be sure to close all open browser windows before logging back into the phone. Also, be sure to reset the key in the phone resource in admin.onsip.com. Follow these instructions- Panasonic Boot Server. BLOCK/BLACKLIST CALLS ON GRANDSTREAM PHONES Block/Blacklist Calls on Grandstream Phones - unsupported. Please see attached PDF for instructions to blacklist inbound calls by number on certain Grandstream phones. The best bet is to search for "Grandstream blacklist" and your phone model number. Note: A blacklisted call will not be disconnected. The phone will respond with "Busy Here" as MIKROTIK SIP ALG = SIP HELPER Mikrotik SIP ALG = SIP Helper. Created September 2015. Mikrotik SIP ALG is called a SIP Helper and is located under /IP>Firewall>Service ports. To disable, run this command from the terminal: /ip firewall service-port disable sip. Or from winbox just navigate to IP>Firewall and then click on the Service Ports tab and disable it through theGUI.
PANASONIC AUTO ANSWER Simply choose the line with the up-down arrows that you want auto-answered and choose the AutoAns soft key. That will enable Auto Answer for that line and the icon for that line will change to burnt orange to indicate Auto Answer is enabled. To disable, perform the same keypresses and the icon will return to pink. Then, when that linereceives
ONSIP SUPPORT
OnSIP Trunking. 3CX Trunking. Configure Firebrick for OnSIP Trunking. Grandstream UCM6104 Configuration for OnSIP Trunking. Enabling OnSIP Trunking. Transfer An Existing Phone Number from the PSTN Gateway Service to An OnSIP Hosted PBX. Switchvox Configuration for TROUBLESHOOTING ONE-WAY AUDIO To see if your phones/firewall are sending OnSIP the proper packets, log into the Onsip interface and click on 'Users'. Click on the name of one of the users to see the User Detail. Scroll down to the 'Maintenance' block and click on ' (Show Details)' next to the SIP Registrations line. You should see a 'Contact' address like:sip:username@192
TROUBLESHOOTING
Troubleshooting. We have listed here a collection of the most common issues we hear about and our suggested solutions. Also be sure to check the recommended links at the bottom of the page for device specific issues: I am experiencing echo on my OnSIP line. What can Ido to fix it?
SERVICES AND SERVERS OVERVIEW OnSIP provides two VOIP servers. Each server provides termination, orgination, and registration services. Calls originate from a server with which a customer is registered. Customers may utilize more than one server. sip.jnctn.net. provides for "trunked" SIP call delivery. username/password used for termination and registrationauthentication.
SMS / TEXT MESSAGING SMS / Text Messaging. All OnSIP phone numbers (DIDs) are enabled for SMS (texting), but OnSIP does not directly provide SMS services. Here are a few third-party vendors that may be able to help; there are many more on the Internet. Phonewire (Disclaimer: OnSIP as a 3-WAY CONFERENCE CALLING 3-Way Conference Calling. Note: This feature is currently only available when using Firefox as your browser. Once logged into app.onsip.com (in Firefox), start a voice call. Once the call is active, click on the “+” sign button (the 'Conference' button) on the right side of the call handling options. Enter the phone number,extension, or
VOICEMAIL REFERENCE GUIDE To access the voicemail manager, dial *98 on your OnSIP registered phone or device. Next, enter your mailbox number, followed by your mailbox password. (Tip: Press # after your mailbox number and password to speed up the process.) If you do not know your voicemail password, please contact your OnSIP Account Administrator. ASTERISK CONFIGURATION Configuring SIP. Go to https://admin.onsip.com and login. Go to the Configuration tab and note your VOIP username and password. Edit /etc/asterisk/sip.conf, replacing MY_USERNAME and MY_PASSWORD in the "register => " statement below with your VOIP username and password. ; ; The "general" context should already exist in sip.conf ; Add a lineto
GRANDSTREAM DIGIT MAP / DIAL PLAN User can dial 92125551234 and the string will respond with 12125551234. User can dial 111 (1xx) or 222 (2xx) and phone will allow it. User can dial any number starting with an asterisk (*) User can dial *xx*xxxxxx. Thanks Chuck! From Grandstream. The dial-plan will restrict the number dialed. BLOCK/BLACKLIST CALLS ON POLYCOM PHONES Click on the name of the unwanted call. Click on the pen/pencil icon on the right to edit. Scroll down to Auto Reject. Select "Enable", then "Save" (on left) Upper left depress the arrow to go back to Directory and select another number or exit the page. Note: A blacklisted call will not be disconnected.ONSIP SUPPORT
OnSIP Trunking. 3CX Trunking. Configure Firebrick for OnSIP Trunking. Grandstream UCM6104 Configuration for OnSIP Trunking. Enabling OnSIP Trunking. Transfer An Existing Phone Number from the PSTN Gateway Service to An OnSIP Hosted PBX. Switchvox Configuration for TROUBLESHOOTING ONE-WAY AUDIO To see if your phones/firewall are sending OnSIP the proper packets, log into the Onsip interface and click on 'Users'. Click on the name of one of the users to see the User Detail. Scroll down to the 'Maintenance' block and click on ' (Show Details)' next to the SIP Registrations line. You should see a 'Contact' address like:sip:username@192
TROUBLESHOOTING
Troubleshooting. We have listed here a collection of the most common issues we hear about and our suggested solutions. Also be sure to check the recommended links at the bottom of the page for device specific issues: I am experiencing echo on my OnSIP line. What can Ido to fix it?
SERVICES AND SERVERS OVERVIEW OnSIP provides two VOIP servers. Each server provides termination, orgination, and registration services. Calls originate from a server with which a customer is registered. Customers may utilize more than one server. sip.jnctn.net. provides for "trunked" SIP call delivery. username/password used for termination and registrationauthentication.
SMS / TEXT MESSAGING SMS / Text Messaging. All OnSIP phone numbers (DIDs) are enabled for SMS (texting), but OnSIP does not directly provide SMS services. Here are a few third-party vendors that may be able to help; there are many more on the Internet. Phonewire (Disclaimer: OnSIP as a 3-WAY CONFERENCE CALLING 3-Way Conference Calling. Note: This feature is currently only available when using Firefox as your browser. Once logged into app.onsip.com (in Firefox), start a voice call. Once the call is active, click on the “+” sign button (the 'Conference' button) on the right side of the call handling options. Enter the phone number,extension, or
VOICEMAIL REFERENCE GUIDE To access the voicemail manager, dial *98 on your OnSIP registered phone or device. Next, enter your mailbox number, followed by your mailbox password. (Tip: Press # after your mailbox number and password to speed up the process.) If you do not know your voicemail password, please contact your OnSIP Account Administrator. ASTERISK CONFIGURATION Configuring SIP. Go to https://admin.onsip.com and login. Go to the Configuration tab and note your VOIP username and password. Edit /etc/asterisk/sip.conf, replacing MY_USERNAME and MY_PASSWORD in the "register => " statement below with your VOIP username and password. ; ; The "general" context should already exist in sip.conf ; Add a lineto
GRANDSTREAM DIGIT MAP / DIAL PLAN User can dial 92125551234 and the string will respond with 12125551234. User can dial 111 (1xx) or 222 (2xx) and phone will allow it. User can dial any number starting with an asterisk (*) User can dial *xx*xxxxxx. Thanks Chuck! From Grandstream. The dial-plan will restrict the number dialed. BLOCK/BLACKLIST CALLS ON POLYCOM PHONES Click on the name of the unwanted call. Click on the pen/pencil icon on the right to edit. Scroll down to Auto Reject. Select "Enable", then "Save" (on left) Upper left depress the arrow to go back to Directory and select another number or exit the page. Note: A blacklisted call will not be disconnected.TROUBLESHOOTING
Troubleshooting. We have listed here a collection of the most common issues we hear about and our suggested solutions. Also be sure to check the recommended links at the bottom of the page for device specific issues: I am experiencing echo on my OnSIP line. What can Ido to fix it?
SERVICES AND SERVERS OVERVIEW OnSIP provides two VOIP servers. Each server provides termination, orgination, and registration services. Calls originate from a server with which a customer is registered. Customers may utilize more than one server. sip.jnctn.net. provides for "trunked" SIP call delivery. username/password used for termination and registrationauthentication.
ANNOUNCEMENT
Creating an Announcement. In the Admin Portal, click on the "Apps" tab. Then, click on "Create New Apps" > "Announcement" > "Create a new Announcement". Give the announcement an appropriate, descriptive name. Choose the recording to play to the caller. Select the destination to forward the caller on to after playing the recording (the "TransferPHONE NUMBERS
click Create New Resource. select Phone Number - there's an indication on this page if you have enough funds to buy a phone number as well as how much each phone number will cost. select Create a New Phone Number. from the dropdown select the Area Code (first 3 digits) from the dropdown select the Rate Center (next 3 digits) GRANDSTREAM DIGIT MAP / DIAL PLAN User can dial 92125551234 and the string will respond with 12125551234. User can dial 111 (1xx) or 222 (2xx) and phone will allow it. User can dial any number starting with an asterisk (*) User can dial *xx*xxxxxx. Thanks Chuck! From Grandstream. The dial-plan will restrict the number dialed. ASTERISK CONFIGURATION Configuring SIP. Go to https://admin.onsip.com and login. Go to the Configuration tab and note your VOIP username and password. Edit /etc/asterisk/sip.conf, replacing MY_USERNAME and MY_PASSWORD in the "register => " statement below with your VOIP username and password. ; ; The "general" context should already exist in sip.conf ; Add a lineto
RESET PANASONIC PHONES TO FACTORY DEFAULT Go to the "Maintenance" tab, Click on "Reset to Defaults" in the left column, and Click on "Reset to carrier defaults". Click "Restart". Be sure to close all open browser windows before logging back into the phone. Also, be sure to reset the key in the phone resource in admin.onsip.com. Follow these instructions- Panasonic Boot Server.LINPHONE FOR IPHONE
Step 6: Configure Network Settings. Select "Network" tab from Settings page. Turn 'off' ICE, 'Allow IPv6' and 'Edge optimization' settings - see image below: Once done depress the left arrow to go back to Settings page.Your registration WILL FAIL - you're almost done. MIKROTIK SIP ALG = SIP HELPER Mikrotik SIP ALG = SIP Helper. Created September 2015. Mikrotik SIP ALG is called a SIP Helper and is located under /IP>Firewall>Service ports. To disable, run this command from the terminal: /ip firewall service-port disable sip. Or from winbox just navigate to IP>Firewall and then click on the Service Ports tab and disable it through theGUI.
BLOCK/BLACKLIST CALLS ON GRANDSTREAM PHONES Block/Blacklist Calls on Grandstream Phones - unsupported. Please see attached PDF for instructions to blacklist inbound calls by number on certain Grandstream phones. The best bet is to search for "Grandstream blacklist" and your phone model number. Note: A blacklisted call will not be disconnected. The phone will respond with "Busy Here" as* Admin
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* Router Configuration * Ubiquiti EdgeRouter Lite 3Follow
UBIQUITI EDGEROUTER LITE3/SECURITY GATEWAY In the UniFi controller after version 5.16, the SIP module can be disabled in the controller- Disable the H.323 and SIP modules But in the meantime, for the next couple of weeks the process that Ubiquiti recommends to disable SIP ALG is as follows:SSH into USG
the commands you need to run are:configure
set system conntrack timeout udp stream 30 set system conntrack timeout udp other 30 set system conntrack modules sip disablecommit
save
exit
SIP ALG can be disabled by config.properties config.ugw.voip.sip_alg_disable=true _===================================/__August 2016_
To disable SIP ALG (application level gateway) on this device: Open the Command Line InterfaceSSH into the router
ssh* configure
* set system conntrack modules sip disable* commit
* save
Exit
_NOTE: The information provided above is from another OnSIP customer offering these settings for other customers with a similar device. OnSIP does not sell nor monitor equipment and/or it's firmware updates/etc. The settings of routers can change and are out of the control of OnSIP. For best results, reach out to the manufacturerdirectly._
------------------------- Optimize your network for business VoIP with the right router Download Business Routers Guide Was this article helpful? 1 out of 1 found this helpfulCOMMENTS
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FULLER JR. RICK May 11, 2016 10:17 Can anyone tell me the show command to verify this is already set? It seems to have reverted on the gateway pro I am working on. Comment actions Permalink*
FULLER JR. RICK May 11, 2016 11:18 Ah...If anyone else runs into this, it should show that command if its set when you run >show configuration commands Comment actions PermalinkRELATED ARTICLES
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* PFSense Firewall Settings for VoIP * Configure UniFi phones from Ubiquiti via UniFi VoIP Controller Can't find what you're looking for? Submit a request. ------------------------- Design © 2020 OnSIP, Inc. All rights reserved.Details
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