I also really need help integrating the Nortel Option 11C with asterisk. If anyone knows how to get a copy of this PDF or a new link to this document, PLEASE post it here.

The PDF that the link for the Option 11 and others is missing. Anyone got a current link or copy of the PDF?
asterisk-meridian-a1.pdf

try my response in "Re: flash transfer problem in asterisk integration with old PBX"

I've solved the problem changing the flash time in the zapata.conf file,
I've set:

flash = 200 (the defualt was 750 ms)

in the extensions.conf the code is for example:

exten => 42,1,Flash()
exten => 42,2,SendDTMF(42,250)
exten => 42,3,Hangup()

now the transfer with flash works correctly,
the call is transfered over the same channel

Hi,
I've tried to transfer a call between Asteriks and my old PBX with the Flash command using your suggestions:

_XXX,1,Flash()
_XXX,2,SendDTMF(*70w${EXTEN},250)
_XXX,3,Hangup()

but the phone where I want to transfer the call doesn't ring,
probably the old PBX doesn't interpret the characters *70w before the extension.
I don't have any manuals for this pbx, but I'm searching to understand if the characters *70w are specific for Norstar meridian
or are also correct for other PBX.

Hello there!
We have a meridian m8x24-ds ( dr5 ) and a couple of m0x16. The problem is that we don't have an auto attendant for incoming calls from 5 lines of PSTN. So every incoming calls are transfered to an operator extension. She talks to the caller and transfer the call to the destination using - FUNCTION 70 and the extension number - with her meridian phone. Ovbiously this is primitive.

We are a public university in Argentina, so we don't have budget to buy a auto attendant from norstar (and that model is very old, there is no selling of that in Argentina)

So, I want to implement asterisk as a solution for an auto attendant (and later for expansion and world domination ;) )

I'm thinking of this structure:

1-incoming calls are transfered to the extension were asterisk is:
PSTN line -> norstar -> ATA -> FXO (zap/1) asterisk

2- asterisk responds with an auto attendant and expects an input from the caller. After the input the caller is transfered to the destination

3- if the caller does not submit an input the call is transfered to the operator extension.

My question is:
Can asterisk transfer the call over the same channel?
Can asterisk (behind the ATA) do something like FUNCTION 70 and dial the extension?
Or: do I have to use another FXO port behind another ATA and transfer the call using that channel?

Thank for the help!!!

Try my response in Re: trouble: asterisk as an auto attendant for norstar meridian

I have the solution....

First: it was difficult to find the norstar manuals, I searched and found http://www.p1bcinc.com/nortel/norstar_manuals.htm

Now the solution:
In the ATA manual they explain how to do some of the functions of a digital meridian phone with an analogous phone.
So, the transfer function is:
LINK * 7 0

They say that if you don't have a link button in your phone you can press the hook switch for approximately one half of one second
In asterisk we can do that with a Flash() function.
So we end with:

_XXX,1,Flash()
_XXX,2,SendDTMF(*70w${EXTEN},250)
_XXX,3,Hangup()

Hope this works ok!

I've tried to transfer a call using the Flash command, but with my configuration it doesn't work.
I have a traditional PBX connected with a zap channel to Asterisk that acts like an IVR:

TELCO line --> traditional PBX (FXS) --> (FXO) Asterisk

From the TELCO line I can make a call to the traditional PBX and reach Asterisk, the IVR system on Asterisk answers the call and I can dial an extension (for example 42 that is on the traditional PBX). In the asterisk dialplan I've set to transfer the call using Flash() like in this example:

exten => 42,1,Flash()
exten => 42,2,Background(silence/1) wait 1 second for the traditional PBX
exten => 42,3,SendDTMF(42,250)
exten => 42,4,Background(silence/1) wait 1 second for the traditional PBX
exten => 42,5,Hangup()

When I dial the extension 42, the phone 42 on the traditional PBX rings but when I answer there isn't communication with the call from the TELCO line and after a few seconds the line hangup.
Here you can see what happen in asterisk CLI console:

      Executing Answer("Zap/4-1", "") in new stack
   — Executing BackGround("Zap/4-1", "a_suoni_plink/menu_esterno2") in new stack
   — Playing 'a_suoni_plink/menu_esterno2' (language 'it')
 == CDR updated on Zap/4-1
   — Executing Flash("Zap/4-1", "") in new stack
   — Flashed channel Zap/4-1
   — Executing BackGround("Zap/4-1", "silence/1") in new stack
   — Playing 'silence/1' (language 'it')
   — Executing SendDTMF("Zap/4-1", "42") in new stack
   — Executing BackGround("Zap/4-1", "silence/1") in new stack
   — Playing 'silence/1' (language 'it')
   — Executing Hangup("Zap/4-1", "") in new stack
 == Spawn extension (incoming, 42, 5) exited non-zero on 'Zap/4-1'
   — Hungup 'Zap/4-1'

I've tried the following changes to the dialplan in my example but transfer still doesn't work:

- I've tried to use wait(1) instead of Background(silence/1)

- I've tried without Background(silence/1) or wait(1):

exten => 42,1,Flash()
exten => 42,2,SendDTMF(42,250)
exten => 42,3,Hangup()

- I've tried without the Hangup() instructions at the end

Has anyone the same problem like me and any suggestions?

I'm new to the VoIP world. Would anyone mind answering a few questions for a newbie. <br>
We currently have a Mitel SX-200 PBX,<br>
Which (two or more port) PRI card would you recommend? (Sangoma?)<br>
How did you migrate your VoiceMail system? (Can coexistence be done?)<br>
Do you have a working configuration that I could look at? (zapata.conf etc)<br>
Can you recommend Cost Effective phones (Preferably Gigabit w/ CoS, etc)<br>
-Cheaper IP Phones (Physical)<br>
-Mid-range IP Phones (Physical)<br>
-High-range IP Phones (Physical)<br>
Which (weight effective) protocol or mix of protocols would you recommend? (MGCP/SIP/H.323/etc)<br>
Are there any warrantee/legal issues by eventually migrating away from all Mitel Equipment?<br>
Any other tips or information I should know before sticking my neck out?<br>
<br>
Thanks in advance.<br>
markbest@co.nezperce.id.us<br>