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Twilio SIP Trunk Setup with Vodia


Travis T.

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Hello,

 

I have noticed documentation on the forum for setting up Twilio SIP Trunks with Vodia.  However, the screenshots that were linked to the OPs Dropbox account have expired and are not available for reference.  Does anyone have example documentation for setting up Vodia on a Twilio SIP trunk?

 

This would be MUCH appreciated.

 

Thank you,

 

Travis T.

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23 hours ago, Vodia PBX said:

Did you try the drop down when setting up the SIP trunk on the PBX? That should set up the right headers and everything for the Twilio trunk.

Yes, however, I am unsure what to put for the Outbound Proxy Address, Domain, Trunk ANI, User name or password.  I have tried several parameters in these fields that would make sense but nothing seems to work.  I am looking for an example configuration that would point me in the right direction.

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All of those details are given by your trunk provider and depending on the trunk (registration or gateway) that you are using. Any of the settings from the PBX's side can be seen when you click on "?" next to any of the PBX GUI pages.

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HI, Travis 

Did you create an Elastic trunk? If not here's how...

Termination URI

  1. Under Elastic SIP trunking create a trunk name under general, to name your trunk.
  2. Next is Termination, create a termination SIP URI, 
  3. Name the  Termination SIP URI for example (mycompany).pstn.twilio.com this SIP URI will be used for the outbound proxy of the trunk.

Twilio also has localized Termination URI, You can use this based on your location, so with your newly created SIP URI you can also use mycompany .pstn.us1.twilio.com etc. 

  • {example}.pstn.us1.twilio.com (North America Virginia)
  • {example}.pstn.us2.twilio.com (North America Oregon)
  • {example}.pstn.ie1.twilio.com (Europe Ireland)
  • {example}.pstn.de1.twilio.com (Europe Frankfurt)
  • {example}.pstn.sg1.twilio.com (Asia Pacific Singapore)
  • {example}.pstn.jp1.twilio.com (Asia Pacific Tokyo)
  • {example}.pstn.br1.twilio.com (South America São Paulo)
  • {example}.pstn.au1.twilio.com (Asia Pacific Sydney)

Authentication IP access control List

  1.  Authentication is used to authenticate the termination SIP URI IP address or FQDN
  • Click on the  plus button, Friendly name: My business name
  • Add an IP address range: Your PBX IP address
  • Friendly name: VodiaPBX and save

Credential list: is used authenticate the user and password of the SIP URI

  • Friendly name: mycompany.com
  • User name: The user that will be passed when authenticating the SIP request
  • Password: The password used by the user to the authenticate the SIP request

 

Origination 

Here you will create the origination URI. 

  • Origination SIP URI: sip.mypbxbuisnessname.com can also be an IP address as well.

This will be used to send traffic to the PBX, so if you don't have a FQDN you can always use your external IP address

  1. Click on the Plus button and create the (Origination SIP URI) could be IP address or FQDN

Number

Purchase a number from Twilio in order to get inbound and CID working. 

Once you have purchased your number you can point the number to your SIP trunk 

Creating the Twilio Trunk in Vodia PBX

General

  • Name your Trunk 
  • Type (SIP Gateway)
  • Direction  (inbound and outbound)

Registration 

  • Account: User name created in the credential list 
  • Domain:  SIP URI created in the Termination URI (mycompany).pstn.twilio.com
  • Username: User name created in the credential list 
  • Password:  Password created in the credential list 
  • Proxy address: SIP URI created in the Termination URI (mycompany).pstn.twilio.com
  • Explicitly list address for inbound traffic 54.172.60.3  172.18.21.17  54.172.60.3 (Optional)

Routing Redirection

  • You can choose to send calls to the Request URI or to a specific extension

Number/ Call Identification 

  • Trunk ANI: Add your Twilio Number
  • SIP Caller -ID presentation  (Custom headers)
  • Request-URI Value: Let the system decide
  • From  Value: Other <sip:{trunk-ani}@{domain};user=phone>
  • To Value: Same as the Request-URI
  • P-Asstered Identity Value: Don'use header
  • P-Preferred Identity Valye: Don;t use header
  • Remote-Party-ID value: Based on the incoming call
  • Privacy Indication value: RFC3325
  • Rewrite global numbers value "Use + symbol at the beginning

I hope this helps, I know Twilio among other providers can be complex at time......

Aldo check the attachment example. 

Thanks

 

 

Twillio example.txt

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23 hours ago, Vodia support said:

HI, Travis 

Did you create an Elastic trunk? If not here's how...

Termination URI

  1. Under Elastic SIP trunking create a trunk name under general, to name your trunk.
  2. Next is Termination, create a termination SIP URI, 
  3. Name the  Termination SIP URI for example (mycompany).pstn.twilio.com this SIP URI will be used for the outbound proxy of the trunk.

Twilio also has localized Termination URI, You can use this based on your location, so with your newly created SIP URI you can also use mycompany .pstn.us1.twilio.com etc. 

  • {example}.pstn.us1.twilio.com (North America Virginia)
  • {example}.pstn.us2.twilio.com (North America Oregon)
  • {example}.pstn.ie1.twilio.com (Europe Ireland)
  • {example}.pstn.de1.twilio.com (Europe Frankfurt)
  • {example}.pstn.sg1.twilio.com (Asia Pacific Singapore)
  • {example}.pstn.jp1.twilio.com (Asia Pacific Tokyo)
  • {example}.pstn.br1.twilio.com (South America São Paulo)
  • {example}.pstn.au1.twilio.com (Asia Pacific Sydney)

Authentication IP access control List

  1.  Authentication is used to authenticate the termination SIP URI IP address or FQDN
  • Click on the  plus button, Friendly name: My business name
  • Add an IP address range: Your PBX IP address
  • Friendly name: VodiaPBX and save

Credential list: is used authenticate the user and password of the SIP URI

  • Friendly name: mycompany.com
  • User name: The user that will be passed when authenticating the SIP request
  • Password: The password used by the user to the authenticate the SIP request

 

Origination 

Here you will create the origination URI. 

  • Origination SIP URI: sip.mypbxbuisnessname.com can also be an IP address as well.

This will be used to send traffic to the PBX, so if you don't have a FQDN you can always use your external IP address

  1. Click on the Plus button and create the (Origination SIP URI) could be IP address or FQDN

Number

Purchase a number from Twilio in order to get inbound and CID working. 

Once you have purchased your number you can point the number to your SIP trunk 

Creating the Twilio Trunk in Vodia PBX

General

  • Name your Trunk 
  • Type (SIP Gateway)
  • Direction  (inbound and outbound)

Registration 

  • Account: User name created in the credential list 
  • Domain:  SIP URI created in the Termination URI (mycompany).pstn.twilio.com
  • Username: User name created in the credential list 
  • Password:  Password created in the credential list 
  • Proxy address: SIP URI created in the Termination URI (mycompany).pstn.twilio.com
  • Explicitly list address for inbound traffic 54.172.60.3  172.18.21.17  54.172.60.3 (Optional)

Routing Redirection

  • You can choose to send calls to the Request URI or to a specific extension

Number/ Call Identification 

  • Trunk ANI: Add your Twilio Number
  • SIP Caller -ID presentation  (Custom headers)
  • Request-URI Value: Let the system decide
  • From  Value: Other <sip:{trunk-ani}@{domain};user=phone>
  • To Value: Same as the Request-URI
  • P-Asstered Identity Value: Don'use header
  • P-Preferred Identity Valye: Don;t use header
  • Remote-Party-ID value: Based on the incoming call
  • Privacy Indication value: RFC3325
  • Rewrite global numbers value "Use + symbol at the beginning

I hope this helps, I know Twilio among other providers can be complex at time......

Aldo check the attachment example. 

Thanks

 

 

Twillio example.txt 1.27 kB · 2 downloads

I have tried all of this and still no luck.  Not sure what else to do.

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3 hours ago, Vodia support said:

Thanks for the update. Can you exactly tell what you have tested. For example when you make an outbound call what happens? 

A busy signal.  Inbound calls don't work as well.  I was starting to think it is the dial plan, but I am not sure.  I am a North America user.

 

Here is the text view of the dial plan I currently have.

 

1;Twilio;;^([0-9]{10})@.*;"sip:\1@\r;user=phone";;false
2;Twilio;;^([0-9]{11})@.*;"sip:\1@\r;user=phone";;false
3;Twilio;;[911|411];1*;;false

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  • 4 weeks later...

I have discovered (after many failed calls and much testing) when Twilio SIP dials to PSTN they have been using the header "remote-party-id".  This header was defined in a failed RFC : http://tools.ietf.org/html/draft-ietf-sip-privacy-00 . Although this header is supported by some SIP devices and PBXs, this was never accepted as a standard.  Therfore, it should not have been used because it may cause call processing problems.

In fact, it causes certain providers to return a "Busy" and hangup incoming calls. I found out after many days of looking for the cause of failed calls.  Hope Twilio quicking fixed this issue.

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