Jump to content

Dial Plan ???


AG1
 Share

Recommended Posts

I am reading through the Snom One book and I see I should be able to make entries in a dial plan in tandem to ensure a call goes through even if I lose my internet connection.

 

I currently have 2 SIP trunks and I also have a Sangoma gateway with 4 FXO ports.

 

In the book they use 1978* in the Pattern field and again in the Replacement field

 

Is this the correct code or pattern to use to accomplish redundancy?

Link to comment
Share on other sites

I am reading through the Snom One book and I see I should be able to make entries in a dial plan in tandem to ensure a call goes through even if I lose my internet connection.

 

I currently have 2 SIP trunks and I also have a Sangoma gateway with 4 FXO ports.

 

In the book they use 1978* in the Pattern field and again in the Replacement field

 

Is this the correct code or pattern to use to accomplish redundancy?

 

It is important that the preference numbers in the dial plan are distinct, because the processing continues with the next higher preference after the one that triggered the failover.

 

If you have set the country code to "1" then you should be using the pattern "978*".

Link to comment
Share on other sites

It is important that the preference numbers in the dial plan are distinct, because the processing continues with the next higher preference after the one that triggered the failover.

 

If you have set the country code to "1" then you should be using the pattern "978*".

 

That most probably depends on the SIP Header setup. Try to use "No Indication" in the trunk for the Sangoma card. This is absolutely okay as the analog line has the caller-ID tied to the cable anyway.

Link to comment
Share on other sites

  • 2 months later...

I have a similar problem.

 

I'm trying to setup a Vega 50 so that in the event of a SIP failure the customer can make inbound and outbound.

Currently the Vega 50 is sending the POTS fine to the PBX, so inbound is fine when the trunk goes down.

The trouble I'm having is trying to work-out how to get the PBX to dial out when the SIP Out fails. Currently I'm getting "Request Timeout".

 

I have the following settings which may shed some light on what is wrong if you can help?

 

Trunk A

Routing

Destination for incoming calls: Sends calls to...

Failover behaviour: Always, except....

Request Timeout: 10

Accept Redirect: No

Error message: 500 Line....

 

Trunk B (Vegastream)

Routing

Destination for incomming calls: Send all calsl to...

Default account: xxx

Failover Timeout: 10

Accept Redirect: No

Error message..:500 line...

Redirect destination.. : 0201 (this is the FXO port)

 

Dialplan

Name: Outbound Call Dialplan

Pref 100 Trunk A Pattern * Service Flag unassigned

Pref 300 Trunk B Pattern * Service Flag unassigned

 

Standard Dialplan

Pref 200 Trunk A Pattern * Service Flag 9am-5pm M-F

Link to comment
Share on other sites

At first glance this looks good.

 

I would not use any timeouts on the Vegastream (the Ethernet connection to the Vega is not supposed to go down). But that is not the problem I guess.

 

Next step is to dig deeper into the logs. Make sure that you have "trunks" set for logging on level 8 or 9...

Link to comment
Share on other sites

Here are the logs when trying to make an outbound call.

As you say it's almost there but until I can make an outbound call this option of failover with the Vega 50 isn't an option just yet.

 

Your help as always is greatly appreciated.

[5] 2013/07/24 12:10:15: Identify trunk (domain name match) 4

[8] 2013/07/24 12:10:15: To is <sip:##########@pbx.mydomain.com>, user 0, domain 1

[9] 2013/07/24 12:10:15: Generating hf header using {from}

[9] 2013/07/24 12:10:15: Generating ht header using {to}

[9] 2013/07/24 12:10:15: Generating hpai header using {trunk}

[9] 2013/07/24 12:10:15: Generating hf header using {trunk}

[9] 2013/07/24 12:10:15: Generating ht header using {request-uri}
Link to comment
Share on other sites

  • 4 weeks later...

I still cannot get my 870's to dial out on my Nextiva SIP trunks

 

The 820s work fine so it has to be an 870 firmware issue....does it not? If so does the beta version 8.7.4.8 fix this problem?

 

 

Is anyone else having this problem with 870's or am I just the lucky one again?

Link to comment
Share on other sites

Join the conversation

You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.

Guest
Reply to this topic...

×   Pasted as rich text.   Paste as plain text instead

  Only 75 emoji are allowed.

×   Your link has been automatically embedded.   Display as a link instead

×   Your previous content has been restored.   Clear editor

×   You cannot paste images directly. Upload or insert images from URL.

Loading...
 Share

×
×
  • Create New...