Displaying 20 results from an estimated 4000 matches similar to: "Connecting Asterisk and Avaya Definity By E1 . Incoming work, but not outgoing"
2004 Jul 30
1
Connecting Asterisk and Avaya Definity By E1. Incoming work, but not outgoing
Hi All.
I connect asterisk and definity by manual at
www.voip-info.org/tiki-index.php?page=Asterisk%20Avaya.
(I just only have E1, not T1 card).
I see, that card work (in definity trunk status, and at asterisk
== D-Channel on span 1 up
-- B-channel 1 successfully restarted on span 1
-- B-channel 2 successfully restarted on span 1
-- B-channel 3 successfully restarted on span 1
2009 Oct 25
2
Asterisk as the recording server for Avaya Definity
Has anyone tried to replace Witness or Nice recorder with asterisk. I saw a nice article on voip-info.org on how to replace voicemail server for Avaya Definity with asterisk.
The idea behind is to record not only the external channels but also extension to extension (three way calling for which the third leg is asterisk PRI will do)
Any suggestion will highly help
Sam
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2004 Aug 02
4
First Post: Any existing AVAYA Switch -> Asterisk Voicemail configs?
This is my first post, so please feel free to direct me to another list
if needed.
I am in the early stages of researching Asterisk. I administer a small
Avaya Definity G3 switch (~400 users).
Can anyone point my to resources/documents/actual implementation notes
of using Asterisk's voicemail with an Avaya Definity switch?
Many thanks,
Brian Hudson
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2004 Aug 05
3
Avaya/Lucent Definity -> Asterisk interop question
Calling all Definity admins,
Got a few questions about Definity -> Asterisk interoperability.
1) What are the options for integration? Can I hand off extensions from the
Definity and vice versa?
2) Anybody have any working configs they would like to post?
I've found and read the legacy integration on the wiki about the two
systems. I've also googled and found a few threads that were
2006 Dec 29
2
Avaya to Asterisk via H323
I am tasked with linking an Avaya Definity switch to an asterisk box using
it's IP card that handles H.323. All my googles turn up a lot of results but
nothing recent. I am able to find instructions but they are dated from 2005,
and often fail halfway through.
What is the best way to achieve what I want, which is two way calling
between the Avaya switch and Asterisk server using h.323, and
2009 Jan 30
2
Asterisk with Avaya
Hi !
I am trying to connect Asterisk with Avaya Definity.
I use this tutorial to do this http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html
The comunication between avaya and asterisk is fine but without sound. I can call from Asterisk to Avaya and extension ring or Avaya to Asterisk and extension ring too but I cant hear anything
Example
Asterisk ---> Avaya
--
2004 Jan 19
2
Lucent and ISDN-PRI
Hi Everyone,
So I have been further exploring the integration of our asterisk server and
our lucent definity g3si system. I took the suggestion of setting up an
isdn-pri line added the two way tie trunk and the signalling group, but
can't seem to get the PRI signalling working on the asterisk correctly. I've
set pri type to network on the lucent, and pri_cpe in zapata on the
asterisk, but
2004 Apr 02
4
avaya and linux
Does anyone know if avaya voip product is running linux under the hood?
Thanks,
/glen
--
Glen Ford
gford@idiom.com
2006 Nov 05
2
Definity Asterisk CallerID Issue
I am hoping someone could shed some light and point me in the right
direction? I'm attempting to get callerid work between an Avaya Definity
PBX and Asterisk via TE110P connected via T1/PRI Crossover PRI. From the
Definity side I've searched endlessly and came with an example which we
modeled as close as we can, but still no luck. While doing PRI intense
debug span 1 in I see a couple
2009 Aug 19
2
PRI Connected to definity errors
We have setup asterisk to handle our calls before between telco and an
Avaya definity. The PRI keeps locking up every so often.
In addition I keep getting this error when trying to call the avaya:
-- Channel 0/2, span 1 got hangup request, cause 102
-- Hungup 'Zap/2-1'
When that error happens I get a fast busy (congestion) tone.
Any one can point me in the right direction?
TIA
2004 Dec 17
2
Definity PBX with a T100P & TN767E
Hey everybody,
I'm looking for some information that I'm not finding by searching the
2004 Asterisk archives.
I'm currently playing with a Digium T100P card and 2 Grandstream phones,
things are working well. I wanted to move on to linking our Definity
G3R Rev 8.2 to the T100P. Everything that I've read so far shows that
you need a TN464 to accomplish this. We have a TN767E
2004 Jan 16
1
Asterisk Integration with Lucent Definity g3 si
We have had quite a bit of success with our T100P and TE410P cards
interfacing to Nortel Meridian PBXes and also to a Livingston Portmaster 3
using ESF/B8ZS and various combinations of E&M wink and ISDN PRI (usually in
5ESS mode).
In the near future, I may also need to interface to a Definity via a T1. I
was planning to use PRI--is that an option for you on your switch?
-----Original
2007 Dec 28
1
Definity G3R and MWI
Hey everybody,
I've just spent the last two hours Googling and searching the Wiki. I'm
trying to find if there are any listings of codes for the Avaya Definity
G3R, to allow for an Asterisk system to turn on/off a phones MWI that is
attached to a G3. We are looking to use an Asterisk system as a voice
mail server.
I'm not having any luck, anybody have such information?
Thanks,
2009 Jan 22
1
Help with Avaya integration
Hi,
I'm trying to integrate my asterisk PBX to an Avaya PBX. I am using
chan_ooh323 from asterisk-addons.
I am able to make a call from SIP Phone -> Asterisk -> Avaya -> Station
(phone) and vice versa.
I am also able to make a call from SIP Phone -> Asterisk -> Avaya -> PSTN.
However I face problems when I make DID calls from the PSTN. The DID
calls are made through
2005 Jan 12
2
T1 Timing Slips
Does anyone know how to monitor * to see if they are receiving timing slips
on a span connected to a T100P card? I am seeing b-channel restarts quite
often and also getting "No D-channels available" warnings from time to time.
Yesterday I had all the b-channels crash during a MeetMe Conference. Not
good! This PRI is connected to an Avaya Definity PBX that is onsite and
located in the
2005 Aug 12
8
Incompatible destination (88) Error Message
I have connected asterisk 1.0.7 with Avaya Definity via E1 with a
TE100P Digium Card.
Inbound calls are working perfectly and I dont have any problem. But
when I try to make an outgoing call with my softphone (xlite) I am
getting the following messages.
Hungup 'Zap/13-1'
Executing Dial("SIP/IZ-bc0a", "Zap/g1/3118") in new stack
Called g1/3118
Channel 0/1, span 1 got
2005 Mar 22
4
TE405P and echo
I am using a SIP softphone (X-lite, SIPPS or Firefly) connected to an
Asterisk v 1.0.3 PBX. The PBX is also connected via a ISDN-PRI crossover
cable to a Avaya Definity Generic 3 PBX via a TE405P card. All outside of
the office calls go through the Definity. Here's the issue:
Calls to internal SIP extensions, Definity extensions, other offices within
our private network (through the
2005 Aug 04
0
h.323 Call problem asterisk to\from lucent(avaya) definity
Hello,
We want to make H323 calls between asterisk and avaya(lucent) pbx.
We create node-name,H.323 signaling group,trunk,
but we can not make H.323 calls to asterisk. Also no warnings exist in
debug.
Instead of giving the IP of Asterisk ,i give my computer's IP and run
SJPhone ith H.323 GUI.
In this time, connection is established.
SJPhone accepts H323 calls but Asterisk does not.
Do
2004 May 19
1
One-way audio with H.323 --> SIP call
Good day,
I have a puzzling issue that people in the IRC channel recommended I
post to the list so here goes :)
I am trying to call a SIP softphone from an H.323 hardphone. The
hardphone is connected to a Definity Prologix R12 PBX with a MedPro card
and a CLAN. The Avaya is setup to send any call to extension 1609 down
an H.323 trunk group that is destined for the Asterisk server. When I
call
2010 Mar 15
3
USING ASTERISK AS AVAYA DEFINITY RECORDING SERVER
Hi there
I remember to ask this question in the past but now I have thought of
something little bit difference. While I understand that asterisk dialplan
accept the call to be answered[ Answer() ] in the dialplan, I wanna know if
this is possible;
i. A call on legacy PBX, extension to extension is made.
ii. On call bridging, the legacy PBX initiate a third bridging to the
recording system via an