How to assigned a T1 port to an specific extension

[jjff2001]

Some information...

I have configured the PBX G3r from AVAYA to the Asterisk box through a T1 link, in the PBX side I dont have a trunk group, I use each one of the channel ports from the T1 as an station, this means that 1 port is know in the PBX as an extension, example: extension 3000 is the channel 1 from the 23 channels in the T1, 3001 is the channel 2 from the 23 channels in the T1....

So when someone call from one extension in the PBX example 4000 to the extension 3000 the call goes to the asterisk box, so the question here is: how can I assigned each one of the port from the T1 to an especific sip extension in the asterisk box, example from the full path call from an extension in the PBX to an specific extension in the asterisk box:

AVAYA extension that is calling ---> AVAYA DS1FD extension(T1 port) ---> Specific Sip extension
Example
4000 (From AVAYA PBX) ----> 3000(Virtual Extension Port 1 T1 link) ---> 2000 (Sip extension in the Asterisk)

I have all configurated, the only part I am looking is how to assigned the specific port to a specific sip extension in the Asterisk box.

Other option is to create a pool extension in the asterisk to send the call to a bounch of extension.
If you have any idea just let me know it.

Best regards

 

[dufus2506]

All I can think of is to create specific context for each port on the Asterisk box. Then assign that context to the SIP station.

If you do that, you can make sure that each station only uses the specific port that is assigned to it.

It's restrictive, but it will work.

Carpe dialem! (Seize the line!)

 

[dufus2506]

So in your configuration files instead of having ports assigned as:

context=incoming
channel => 1-24

It would be:

context=2000
channel => 1

context=2001
channel => 2

etc.

etc.

Then assign the channel context to the correct station.



Carpe dialem! (Seize the line!)

 

[jjff2001]

Do I need to do this over the sip.conf file?

 

[dufus2506]

No. This would be done in the chan_dahdi.conf file

http://www.voip-info.org/wiki/view/chan_dahdi.conf

You would assign each sip station to the port you wanted it to use by assigning each sip station it's own context.

However, I would not have done it this way myself.

I would have created a single trunk group between the Avaya PBX, and the Asterisk system. Then I would have used UDP to route a group of phone numbers to the Asterisk system.

You could then use any trunk in the trunk group to complete a call to any station on the Asterisk box, and you wouldn't have to worry about a call placed from the Asterisk box blocking an incoming call to an Asterisk station.

Carpe dialem! (Seize the line!)

 

[jjff2001]

The idea is to use a DSF1D extension in the PBX:

STATION

Extension: 0010 Lock Messages? n BCC: 0
Type: DS1FD Security Code: TN: 1
Port: 02B0601 Coverage Path 1: COR: 95
Name: Asterisk test Coverage Path 2: COS: 1
Hunt-to Station: Tests? y

STATION OPTIONS
Loss Group: 4
Off Premises Station? y
R Balance Network? n

This means that if I dial from my extension 4020 to the DS1FD extension the call is route to the asterisk.

I added the context into the chan_dahdi.conf:

group = 1
echocancel = yes
echotraining = yes
echocancelwhenbridged=yes
rxgain=-1.0
txgain=-4.0
signalling = pri_net
switchtype = 4ess
usecallerid = yes

context = 4000
channel => 1

context = incoming
channel => 1

context = incoming
channel => 2-23

The context is on the dahdi show channels:

tp2asterisk01*CLI> dahdi show channels
Chan Extension Context Language MOH Interpret Blocked State
pseudo default default In Service
1 4000 en default In Service
2 incoming en default In Service
3 incoming en default In Service
4 incoming en default In Service
5 incoming en default In Service
6 incoming en default In Service
7 incoming en default In Service
8 incoming en default In Service
9 incoming en default In Service
10 incoming en default In Service
11 incoming en default In Service
12 incoming en default In Service

But I need when I call from my extension 4020 to the DS1FD extension in the AVAYA PBX 2000 rings directly to a SIP extension, I added the configuration into the chan_dahdi.conf but it looks like I need to add something else, because from the AVAYA I will never dial a sip extension, so I need to tell asterisk that all calls from the port 1 from the T1 route them to an specific sip extension, I think this last part needs to be load it into the extensions.conf, please let me know your comments.

 

[jjff2001]

Typo:

I need when I call from my extension 4020 to the DS1FD extension in the AVAYA PBX (Extension 2000) rings directly to a SIP extension, I added the configuration into the chan_dahdi.conf but it looks like I need to add something else, because from the AVAYA I will never dial a sip extension, so I need to tell asterisk that all calls from the port 1 from the T1 route them to an specific sip extension, I think this last part needs to be load it into the extensions.conf, please let me know your comments.

 

[dufus2506]

You're right. You need to assign the port in the chan_dahdi.conf to the SIP station in extensions.conf.

The reason I said you should assign a different context to each port, is to make sure that when a SIP station places an outbound call, it always uses it's own port.

If you just send all of the calls to a port group, then it could use a port from another station. That would block incoming calls to that station.

I still think you should setup the Asterisk system to accept 4 digit DNIS. Then build the DS1 card as a TIE trunk group, and send the 4 digit DNIS to the Asterisk system using UDP. That would allow you to call the SIP extensions from the Avaya PBX directly. It would also allow you to have more than 24 stations on the Asterisk system.

Another advantage is that you don't use up station licenses on the Avaya PBX. The only downside is you'd have to use the Asterisk system for voicemail.





Carpe dialem! (Seize the line!)