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Basic Sip to Analog Gateway setup

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I am looking for basic configuration example for a TA908e

I would like to connect a 3 channel SIP trunk to 3 FXS ports using 3 different DID's for each port

I have the SIP trunk registered with the provider however i cant figure out how to make the connection to the FXS ports

Thank you

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Contributor III
Contributor III

Re: Basic Sip to Analog Gateway setup

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in4ni,

     That is typically the way it is done.  The Adtran should read that information for it's call routing.  It could be they are sending different number of digits then what you are looking for.  If they are sending 4, 7, or 10 digits then you need to have a matching DID on your voice users.  The output of debug sip stack messages and debug voice verbose would help a lot in verifying the issue.

John Wable

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Re: Basic Sip to Analog Gateway setup

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Well I managed to get a call through both ways by using the sip URI as the FXS  extension number, However I cant seem to get the TA908 to route the incoming DID properly

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Contributor III
Contributor III

Re: Basic Sip to Analog Gateway setup

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in4ni,

    Since you have outgoing working I am going to assume you have the working SIP Trunk setup so the problem should be on the Voice User config side.  You can either create the voice user using the DID command and assigning the DID number to it or you can create a voice user with the DID as the voice user number.  If the below do not help please provide a sanitized copy of your config file, the results of debug sip stack messages, debug voice verbose during an inbound call. Two examples follow:

Voice User name is the DID:

voice user 5555551611

  connect fxs 0/2

  first-name "Line 2"

  last-name "Name"

  no special-ring-cadences

Voice User using DID Command:

voice user 1234

  connect fxs 0/2

  did "5555551611"

  first-name "Line 2"

  last-name "Name"

  no special-ring-cadences

John Wable

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Re: Basic Sip to Analog Gateway setup

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The problem seems to with the provider, They are not sending the DID with the SIP Invite, the only DID information is to the left of the @ in the "TO:" field.

Thank you for your reply

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Contributor III
Contributor III

Re: Basic Sip to Analog Gateway setup

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in4ni,

     That is typically the way it is done.  The Adtran should read that information for it's call routing.  It could be they are sending different number of digits then what you are looking for.  If they are sending 4, 7, or 10 digits then you need to have a matching DID on your voice users.  The output of debug sip stack messages and debug voice verbose would help a lot in verifying the issue.

John Wable

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Contributor III
Contributor III

Re: Basic Sip to Analog Gateway setup

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Hello,


I went ahead and flagged the "Correct Answer" on this post to make it more visible and help other members of the community find solutions more easily. If you don't feel like the answer I marked was correct, feel free to come back to this post and unmark it.  If you still need assistance, we would be more than happy to continue working with you on this - just let us know in a reply.

Thanks,

Geoff

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