Chan_ooh323 To Sip , No Connected Line Info

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Hello!

We have asterisk connected over PRI no our phone network, so I'm avaya PBX user. Asterisk connects to another avaya system over h323.

Connection can be shown as

avaya--PRI-asterisk--h323-avaya

When I do call as avaya user I see name of remote end avay user, i.e. connected line info.

As I see in debug remote side send is as

14:07:29:758 Received H.2250 Message = { 14:07:29:758 protocolDiscriminator = 8 14:07:29:758 callReference = 47 14:07:29:758 from = destination 14:07:29:758 messageType = 7 14:07:29:758 Display IE = { 14:07:29:758 Disa 14:07:29:758 }

over h323.

But now we need to connect to another asterisk over SIP.

in this case we have…

Asterisk Users 3 months ago 1 Answer

Asterisk As A Media Gateway

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I'm playing around in a lab, and I was wondering if its possible to have Asterisk act similar to that of a Avaya PBX, where we have media gateways do the heavy lifting.

This is what I was thinking of trying.

1. One asterisk server will contain the logic of the phone system (ex: queues, extensions...etc).

2. The mains server will not handle RTP traffic, it will send the RTP traffic to another system (another asterisk box?) for processing.

At the end of the day, what I am hoping for is to have 1 brain, and mutiple work horse audio gateways that…

Asterisk Users 1.5 years ago 1 Answer

Transparent Elastix 2.2 fax receiving problem

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Hi all, First off all let me apologize for my English. My company use an Avaya PBX with one E1 and about 250 extensions
(phones, faxes alarms, visa card readers etc).
My idea was to install a transparent asterisk between TELCO and PBX. I use an OpenVox DE210E.
After install Elastix 2.2.0 and run dahdi_genconfig, i edit
/etc/asterisk/dahdi-channels.conf changing span 1 group from 0,11 to 0
and span 2 from 0,12 to 1. Then from gui add all the active avaya extensions as custom extension
with dial DAHDI/i2/extension-number The phones works fine but faxes…

Asterisk Users 3.4 years ago 0 Answers

Asterisk refuses INVITE (401) and I don't know why

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Jonas, May I suggest that you present us your sip.conf entry for this peer, properly redacted, of course. That might help more. What I do for "gateways" at known addresses is to put an
entry like this into the sip.conf entry:
[peer]
type=peer
defaultip=192.168.40.123
insecure=invite,port
context=some_context On 11/22/2011 06:40 AM, Jonas Kellens wrote:
> Hello list,
>
> this is the communication between an Aastra 5000 PBX and Asterisk, where the Aastra makes a call to Asterisk. For some reason, Asterisk responds with 401-Unauthorized and I don't

Asterisk Users 3.8 years ago 10 Answers

Avaya & Asterisk FreePBX Integration Problem

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Hi, I'm currently testing my FreePbx Box to work with our Avaya PBX to allow
dialing outgoing international call and FreePBX extensions to avaya PBX
Extensions calling.
Unfortunately no luck to do it successfully. Any help would be much be
appreciated, here is the sample codes I already tried: On FreePBX GUI:
1. I created a custom Trunk called AvayaPBXTrunk with custom dial string
OOH323/$OUTNUM$/Avaya
2. Created an Outbound route called InternationalCall and select
AvayaPBXTrunk on the trunk sequence.
3. Created an Extension 1000 with dial extension OOH323/$OUTNUM$@Avaya On Asterisk CLI:

Asterisk Users 4.1 years ago 0 Answers

need to be able to pass a call to the pstn from another pbx trunk

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pstn                                                           pstn asterisk                 link between                 avaya pbx both systems tied together by 2 pri's
both have trunks out to the pstn
want to get rid of the avaya pstn trunk and send thru my asterisk box
avaya still has inbound calls on this trunk until late november (at&t
is dragging their feet doing the porting - 8 weeks between ) and still
has stations that we are not in a position to migrate to asterisk just
yet (about 500)
can get the call to show on the link between systems, but asterisk has

Asterisk Users 4.8 years ago 0 Answers