Monday, 21 April 2008

HOWTO Connect Avaya to Asterisk

Hi All,

I think that it could be nice to share my knowledge on this subject and help the others people to connect Asterisk to Avaya:

In first we need to add on Avaya a new node-name:

Make on Avaya "change node-names ip"

And attribute the name Sip_server for our example and also the IP address of your Asterisk server:


Create a new signaling group, you need to use a CLAN of your system and the node-name that you have created above, keep the fiedl "Trunk group for Channel Selection" empty, this field will be fill after creation of the Trunk.



After the creation of the signalling we can create the trunk, in my example I'll create the trunk 42 but you can use a different number:

Page 1 of the trunk:



Page 2 of the trunk:





Page 3 of the trunk:



Page 4 of the trunk:



Page 5 of the trunk:



Page 6 of the trunk:



Creation of the uniform dialplan:



AAR Table:























Route pattern:





Now the trunk has been created don't forget to go back the signalling group that you have created and add in the field "Trunk group for channels selection" the value of your trunk in my example I have created a trunk 42 so it will be the value 42 in the field.

Now on Avaya we are ready to connect on Asterisk so we need to configure Asterisk:

You just need to follow my example below:

In H323.conf

[general]
port = 1720
bindaddr = 10.148.20.26
progress_setup = 8
progress_alert = 8
faststart=yes

h245tunneling=yes
gatekeeper = DISABLE

;We need to conserve the main parameters to allow the h323 to call to the SIP phone
disallow=all
allow=alaw
dtmfmode=inband
context=internal ; name of your context


[Avaya]
type=friend
context=internal
host=10.147.9.64; IP Address of your CLAN
port=1720; port used to connect on CLAN it could be some others port regarding your configuration in signalling group.
disallow=all
allow=alaw
canreinvite=no
dtmfmode=inband




In extensions.conf

[general]
static=yes
autofallthrough=yes


[internal]
;My extension 59xxx
exten => 59000,1,Dial(SIP/59000)
exten => 59000,2,VoiceMail(59000@118218)
exten => 59000,3,PlayBack(vm-goodbye)
exten => 59000,4,Wait(2)
exten => 59000,5,HangUp()



exten => _7XXXXX,1,Dial(H323/${EXTEN}@Avaya); Avaya Extension
exten => _5XXXX,1,Dial(H323/${EXTEN}@Avaya); to call on SIP Extension
exten => _4XXXX,1,Dial(H323/${EXTEN}@Avaya); Your extension on Avaya
exten => _006XXXXXXXX,1,Dial(H323/${EXTEN}@Avaya); to call on mobile
exten => _00XXXXXXXXX,1,Dial(H323/${EXTEN}@Avaya); to call on National


In Sip.conf
[general]
context=internal
bindaddr=10.148.20.26; IP Address of your Asterisk
srvlookup=yes
videosupport=yes ; if you want activate video support
canreinvite=no

[59000]
type=friend
secret=XXXXXXXXXX ;your password
host=dynamic
dtmfmode=inband
disallow=all
allow=alaw
allow=h263 ; to use a video codec if needed
callerid="Cyril CONSTANTIN" <59000>
nat=yes

Do not forget to restart Asterisk when you have finished to configure the .conf files.


I hope that this tutorial will help lot of people, if you follow this tutorial it will be easy for everybody to interconnect the both system.

61 comments:

alamogus said...

Hi, Cyrill Constantin

I have a problem:

I dont call to Avaya:

CLI>h323 set trace 3
...
Parse error of Q931 PDU
...
P.S. When using ooh323, the same problem remains

constantin said...

Hi Alamogus,

Are you using OpenH323?? because I got too many problem with ooh323, it would be better to use OpenH323 from Nufone.

Do you have the trunk establish on Avaya side and signalling ok?

alamogus said...

I have avaya s8300+g350 and * 1.4.19+pwlib-v1_10_0+openh323-v1_18_0 (by NuFone).
When I call on extensions registered locally on avaya, all OK. But when I do calls on numbers not belonging avaya (avaya routing calls outgoing) there is a specified mistake.
Excuse for bad English

constantin said...

Hi Alamogus,

Could you post the traces from Avaya if you do a list trace tac (tac number of your IP Trunk H323) and try a test call.

When you says "When I call on extensions registered" you mean from Asterisk to Avaya? or Avaya to Avaya?

Give me more details if you want that I help you.

Regards.

Kabin said...

We configured our AsteriskNow and Avaya 8300 using your suggestion

But we are getting error message on Avaya Side as below, Can you help us? We donot where in Asterisk side to check the logs??

list trace tac 715 Page 1

LIST TRACE

time data

17:18:30 Calling party trunk-group 15 member 1 cid 0x143
17:18:30 Calling Number & Name NO-CPNumber NO-CPName
17:18:30 active trunk-group 15 member 1 cid 0x143
17:18:30 denial event 2329: No BW, prowler <--> IP ep D1=0x7f4000d3 D2=0x143

Martin said...

add a prowler in the same network region that the c-lan ;)

alerios said...

Hello.

I have succesfully made calls from avaya to asterisk, but I have one-way-audio in calls from asterisk-to-avaya. Watching at the logs, I think its beacuse avaya has 2 address: CLAN IP address is 172.16.7.19 and the RTP address is 172.16.7.14.

How can I make asterisk to understand this?

Here is the log: pastebin.ca/1262803

Andres said...

Hi Cyrill,

I did everything you told me, but I still can´t establish a communication between both systems.

I have a CM 3.1.5 and an Elastix 1.3-2

as_bat

Matt said...

Cyrill... is there a way to pass the Avaya UCID across the H323 trunk and make it avialable in the Asterisk dialplan? Really want to avoid CTI to get this little nugget of info. Thanks.

Macaruchi said...

Hi!
Constantin
I have followed this guide to connect my Avaya 87000. I can connect to Avaya but I cant hear anything just call Avaya and I see that call is coming but I dont have audio. When I call from Avaya I can see the call in asterisk debug. i have ASterisk working. I am using a sjphone SIP

constantin said...

Hi Guys,

You have to change in your signalling group on the field "Avaya Direct IP-IP Audio Connection" to "N" then you'll have a two ways audio

Walter said...

Gentleman,

I followed instructions as posted on the blog and I’m happy to report that it works well.
My implementation was done with Trixbox 2.6.2.2 to Avaya S8720 CM 3.1.5.

Thank you…

pwf1960 said...

We've followed this setup and feel like we are mostly there. However, whilst we have sound both ways Avaya=>trixbox, if we make calls trixbox=>Avaya (extensions on both systems, not going to the outside world) we can hear the person on the Avaya side, but the trixbox extension is mute (although we can see outbound sound levels). We set the IP-to-IP settings on the trunk in Avaya (there were two). We are using Avaya Communications Manager 5.0 service pack 3, and Trixbox CD 2.8.0.

pwf1960 said...

To clarify, we set the IP-to-IP sound settings both to N. The OOH323 config is:

tbpbx*CLI> ooh323 show config
tbpbx*CLI>
Objective Open H.323 Channel Driver's Config:
IP:Port: 10.1.1.2:1720
FastStart yes
Tunneling yes
CallerId asterisk
MediaWaitForConnect yes
Gatekeeper: No Gatekeeper
H.323 LogFile: /var/log/asterisk/h323_log
Context: default
Capability: 0x6 (gsm|ulaw)
DTMF Mode: rfc2833
AccountCode: ast_h323
AMA flags: Unknown
Aliases:
100 ObjSysAsterisk

Belar said...

Hi,
I´m new in this blog. I´m a Technician.

Does I need a DS1 or other card for the configuration of the Trunk? because, where a from the port (T01291) is obtained?

Thank a lot of.

Salu2 fron Spain...

constantin said...

Hi Belar,

You need a CLAN card to make an IP Trunk between your Avaya and Asterisk.

T01291 it's an IP port attributed automatically by the Avaya when you add trunk members to your IP trunk.

Best Regards.

Jayden said...

Hi Cyrill,

I have done the above setup using a SIP phone(Express Talk) on Asterisk and using an H.323 soft/hard phone on Avaya? I'm not able to call from either side. Do I need to use a SIP phone on Avaya Server?

For this setup, do we need to attach the Digium card which provides the FXS and FXO interfaces, on the asterisk server?

Please help me on this...

Thanks in advance...

tzzz said...

Hi Cyrill,

Thank you for this guide, i can follow every step but when creating the trunk groep I cannot select H323 as a Carrier medium. I can only pick ATM, IP or PRI/BRI. Any suggestions?
Thanks in advance

constantin said...

Do a display system-parameters customer-options and go on page 2 you should have some licence for the option "Maximum dministered H323 Trunks"

Please let me know

tzzz said...

Thx for the quick reaction! I have checked the option but it says No at that specific option :-( Is there a other way to connect to asterisk? I have a c-lan card...?

constantin said...

You have to use ISDN link to interconnect both equipment

f.stiriba@gmail.com said...

Bonjour,
le poste téléphonique avaya n'affiche pas toutes les informations concernant l'appelant,si quelqu'un peut me dire comment puis-je le configurer ???

constantin said...

Bonjour,

Vous devez mettre dans votre fichier sip.conf les paramètres suivant pour afficher l'extension SIP:

Par exemple:
[59000]
callerid="Cyril CONSTANTIN" <59000>

Anonymous said...

Hello,

I am having no luck with this. I get a network failure denile event. I have followed your instructions on the Avaya side exactly, but I am not very familiar with asterisk (trixbox). I have edited the h323.config file, but is there something else I need to do. Is the problem because I am using trixbox?

Thanks,

Rob

constantin said...

Hi,

I'm not familiar with Trixbox but I have been able in the past to do it, you need to configure it from the trixbox interface directly.

Regards.

Anonymous said...

Hi friends,

I need your help. I'm running * 1.4.26 and i'm having a lot of problems when i goto IVR in pstn....my ip phones ( 4610sw ) dont send DTMF codes....
could you help me.

tks in advnce
marcus

constantin said...

Can you describe a little bit more your architecture, which kind of IVR have you etc

Peter said...

Cyril,

This is a great guide. Quick question: we have an S8500 that does NOT have SIP functionality. We do have IP Med Pro cards and Avaya IP phones running H.323. Can we connect from our switch to Asterisk just over H.323? You mentionn "Sip_Server" in your config - is that just a general description?

Thanks,

Peter Hoopes

constantin said...

Hi Peter,

Sure that you can connect your S8500 to Asterisk you just need H323 option.

You can do a disp sys custo and check that you have the option "IP Trunk" and also "Maximum Administered H.323 Trunk"

Let me know

Dustin51 said...

Maybe this is a stupid question but are you passing these calls through your network or do you have a physical connection between the two systems?

constantin said...

Hi,

Like you said it trough my network like it's an IP trunk.

Best Regards.

Anonymous said...

I have done this in the past, and these steps are pritty much comperhensive. However, this connection is done over an ip trunk and has nothing to do with sip. SIP and H323 are IP protocoles and each has its own charactaristics. the one used here is the H323. so from your avaya you are simply creating an IP trunk type ISDN and carrier medium can be h323 or ip depending on your avaya version as older versions display ip for this field which is the same.
in regards to the one way voice. There are few things need to be carefully observed:

1-FIND OUT WHAT AVAYA IS SET TO IS IT alaw or ulaw this is very important as the two systems must match.

2- what codec set you are using must be allowed in the trixbox/asterisk (this is the voice coding)

3- i believe there was some thing to do with the hairpinin so pay attention to that as well. Try all options from the avaya.

FOR THE CALLER ID AND NAME TRANSLATION:

1-check your puplic numbering you have to add the prefix as explained in toturial in there and adjust the number of digits.

2- check in the trunk(avaya side) there is an option labled "send name" you have to make it "y" and i believe "send number" play with these.

3- In the trunk group there are two options for supplementary service a and b interchange them and see if they make any differnce.

4- one crazy thing about the route which is a fix for some denial events from avaya is to change the value at the bottom of the page under W to yes or the one next to it TSC and make it as needed.



Good luck

Hamad KSA

darrell said...

Hi Cyrill,

This seems like a very good tutorial. I'm completely new to AVAYA & Asterisk. Would all of the information be valid for a IA770 AVAYA Gateway?

Where can I read up on all the terminology, like Signaling Group, trunks etc to get a better understanding of your tutorial.

Regards,
--
Darrell

constantin said...

HI Darrell,

IA770 it's different, it's a messaging application server, you can't use it to do the same.

Regards.

darrell said...

Hi Cyril,

Apologies, its a G700 gateway & S8300 Server. Like I said, very new at this!
--
Darrell

constantin said...

Darrell,

Yes you can without problem.

Regards

Ruel said...

Hi, Cyril Constantin

I have Trixbox CE and Avaya S8700 working properly using ooh323. My problem is that Avaya can not decode the DTMF I am sending from Asterisk and vice versa. Please help. Thanks!

constantin said...

Hi,

Try to use dtmfmode=auto or dtmfmode=rfc2833.

Best Regards.

Ruel said...

[follow up]

Hello, Cyrill,

I tried both. I tried changing it in ooh323.conf and in the extensions that I am using but still it doesn't work. Is there anything else I am missing?

constantin said...

Can you send me the log file below please:

/var/log/asterisk/ooh323_log
/var/log/asterisk/messages

Anonymous said...

On the Avaya side does it require a CLAN card, or can I just use the processor network port.

I just have a S8300, SES and an asterisk box. Trying to get a H323link up and running between the two.

Kumar said...

I complied and configured. I can make call from Asterisk with a delay about 30s, but when i try from Avaya i got an error
"denial event 1220: Recovery on timer expiry D1=0x8c6d D2=0x66"

Anonymous said...

Hello cyril,
we are using trixbox 2.6.2.3
it seems we have problems when connecting to the avaya. is there a way for this version to interconnect it with avaya?

Lee said...

Hello

We have a S8300 server & G450 media gateway. To connect our Avaya system to our Asterisk box do we need a C-LAN? If so then I believe we need a TN799 module, would such a module work in a G450?

P.S. Great article.

Thank you

Cliff said...

Hi Cyrill,
Has here been any improvement on this seup? I am planning to tackle it using elastix 3.0 and my s8300. Is there a better way?

VoIP Migration said...

Thanks for sharing with stepwise information actually i am searching information about features and durability of Nortel Trixbox
and Avaya SIP and how much is this successful in telecom industries...

Santosh Rao said...
This comment has been removed by the author.
Santosh Rao said...

Is it possible for the Asterisk sip gateway to route calls to avaya LSP in case the avaya main server fails, if yes how?

gaffer said...

We have a S8300 server & G430 media gateway I have done the above setup but the avaya h323 trunk still out off service any
any help

constantin said...

Hi,

do a "display system-parameters special-options" and check on page 4 that you have H245 support with others vendors" to Yes without that you can't do it.

Best Regards

mike das said...

heres something i found out recently.. you can connect the avaya by using a sip trunk to asterisk. no need for a h323 trunk. avaya 5.x can be set up using arr or ars. I have had success with avaya 5.2 and freepbx with asterisk 1.8 g729 for all calls.

James Julius said...

Hello guys, my question may be out of topic a bit, but any help will be appreciated, Can anyone help me how connect AVAYA 1608-I to asterisk PBX?

Sreejith N K said...

Hello Guys,

I am using asterisk 1.8.32 and trying to connect with avaya g450 using h323(ooh323), i am able to receive the call from avaya to asterisk but when i tried to make call from asterisk to avaya it disconnects immedaitely.

from the logs i am getting nocircuit cahnnels available.

constantin said...

Hello,

Please give me your asterisk console logs in order to understand what's going wrong.

Best Regards

Sreejith N K said...

hello

I have tried with older asterisk version also (1.6.2.24) same result is observed in the h323 logs.


14:17:53:313 Receiving H.2250 message (outgoing, ooh323c_o_9)
14:17:53:313 Received Q.931 message: (outgoing, ooh323c_o_9)
14:17:53:313 Received H.2250 Message = {
14:17:53:313 protocolDiscriminator = 8
14:17:53:313 callReference = 102
14:17:53:313 from = destination
14:17:53:313 messageType = 5a
14:17:53:313 Cause IE = {
14:17:53:313 Q931NoCircuitChannelAvailable
14:17:53:313 }
14:17:53:313 h323_uu_pdu = {
14:17:53:313 h323_message_body = {
14:17:53:313 releaseComplete = {
14:17:53:314 protocolIdentifier = {
14:17:53:314 {
14:17:53:314 0 0 8 2250 0 5 }
14:17:53:314 }
14:17:53:315 callIdentifier = {
14:17:53:315 guid = {
14:17:53:315 '28a7cb057f2ae601d62e000010ac0546'H
14:17:53:316 }
14:17:53:317 }
14:17:53:317 }
14:17:53:317 }
14:17:53:317 h245Tunneling = {
14:17:53:318 FALSE
14:17:53:318 }
14:17:53:318 }
14:17:53:318 UUIE decode successful
14:17:53:318 Decoded Q931 message (outgoing, ooh323c_o_9)
14:17:53:318 }
14:17:53:318 H.225 Release Complete message received (outgoing, ooh323c_o_9)
14:17:53:318 Cause of Release Complete is 22. (outgoing, ooh323c_o_9)
14:17:53:318 In ooEndCall call state is - OO_CALL_CLEARED (outgoing, ooh323c_o_9)
14:17:53:318 Cleaning Call (outgoing, ooh323c_o_9)- reason:OO_REASON_REMOTE_CONGESTED
14:17:53:319 Closing H.245 connection (outgoing, ooh323c_o_9)
14:17:53:319 Removed call (outgoing, ooh323c_o_9) from list

Sreejith N K said...

asterisk CLI result

- Executing [1212@from-ext-test:1] NoOp("SIP/8130-0000001c", "Testing the OOH323") in new stack
-- Executing [1212@from-ext-test:2] Set("SIP/8130-0000001c", "CALLERID(num)=8130") in new stack
-- Executing [1212@from-ext-test:3] Dial("SIP/8130-0000001c", "OOH323/1212@172.16.70.5:1720,,tTR") in new stack
-- Called 1212@172.16.70.5:1720
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [1212@from-ext-test:4] Hangup("SIP/8130-0000001c", ""

Sreejith N K said...

Please someone can give a reply?

constantin said...

Hello,

Please provide Avaya trunks traces "list trace tac XXX" to see if you receive something on Avaya side.

Best Regards

Sreejith N K said...

XXX should be replaced with trunk right?

constantin said...

on your trunk you have a TAC code so use this code for the command.

Best Regards

Sagar R said...

I really appreciate information shared above. It’s of great help. If someone want to learn Online (Virtual) instructor lead live training in Avaya Messenging Server
, kindly contact us http://www.maxmunus.com/contact
MaxMunus Offer World Class Virtual Instructor led training on in Avaya Messenging Server We have industry expert trainer. We provide Training Material and Software Support. MaxMunus has successfully conducted 100000+ trainings in India, USA, UK, Australlia, Switzerland, Qatar, Saudi Arabia, Bangladesh, Bahrain and UAE etc.
For Demo Contact us:
Name : Arunkumar U
Email : arun@maxmunus.com
Skype id: training_maxmunus
Contact No.-+91-9738507310
Company Website –http://www.maxmunus.com