AsteriskNow works with Avaya PBX (H323 IP Trunk)

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AsteriskNow works with Avaya PBX (H323 IP Trunk)

Postby gollum93 » Thu Jul 12, 2007 4:12 pm

Hi,
Just to announce that AsteriskNow b6 works with Avaya PBX through H323 IP Trunk.
I use it as answer machine and sip<-->h323 gateway
So i'm able to call avaya phone from sip softphone and sip softphone from avaya phone
If you want any help for the configuration of your avaya pbx system and your AsteriskNow, do a reply to this post
gollum93
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Postby luxy » Fri Jul 13, 2007 8:39 am

Hi, it's great! would you share us your solution? We tried to make a similar connection with an Avaya pbx, but our is failed (a month ago), and we give it up! This connection is capable to display the caller number/name? Can you send me the modified config files?
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Postby Shaggy » Sat Jul 14, 2007 2:34 am

Hi
don't you require the Avaya SIP Enablement Server to enable SIP Connectivity from the Avaya Communication Manager to the SIP end point?

I know that once you purchase this server it allows administration of SIP trunking which i guess you have used to trunk from CM to Asterisk?

Or have you managed to get a true H.323 trunk up between Asterisk and Avaya CM?

Shaggy
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Avaya <-> Asterisknow 323 trunk

Postby rstrickoff » Tue Jul 17, 2007 3:40 pm

I'm currently troubleshooting my way through trying to get an *now box trunked with an S8700. The 8700 side is set up as h323 tie, but the group type is ISDN which I doesn't make sense to me. I'm trying to use one of the CLAN boards as my peer in the h323.conf. Am I going the right direction? Can you post your configs? Any help is much appreciated.
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avay asterisk conf.

Postby bfph3t0 » Wed Jul 18, 2007 8:08 am

I can make calls from avaya to asterisk with no problem. I can not make calls from asterisk to avaya. I sure could use some help.
thanks
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Postby Shaggy » Thu Jul 19, 2007 10:45 am

I can make calls from avaya to asterisk with no problem. I can not make calls from asterisk to avaya. I sure could use some help.
thanks

Hi
I am unable to find the details on how to administer a H.323 trunk in *Now I thought it only supported SIP?

If you could post a tutorial it would be most appreciated.

Shaggy
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Re: Avaya <-> Asterisknow 323 trunk

Postby Shaggy » Thu Jul 19, 2007 10:50 am

rstrickoff wrote:I'm currently troubleshooting my way through trying to get an *now box trunked with an S8700. The 8700 side is set up as h323 tie, but the group type is ISDN which I doesn't make sense to me. I'm trying to use one of the CLAN boards as my peer in the h323.conf. Am I going the right direction? Can you post your configs? Any help is much appreciated.


rstrickoff
The group type of ISDN is correct, it allows the H.323 trunk to emulate ISDN services such as QSIG and DCS+

Have a look at "Administration for Network Connectivity for Avaya
Communication Manager" Page 150 "Administering H.323 trunks"

http://support.avaya.com/elmodocs2/comm ... 504_12.pdf

Cheers Shaggy
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Postby gollum93 » Wed Aug 08, 2007 5:09 am

=========================================
For the S8700 :
____________________________________________________
1 - Check your system-parameters customer-options : Maximum Administered IP Trunks need to be > 0 (it's better to build an ip-trunk ;p )
2 - Change node-name IP and add the IP of your Asterisk box (Asterisk 192.168.12.66)
3 - create a sig group type h.323 with near-end Node Name on of your CLAN / port 1720 and in far-end Node Name the one set in step 2; port 1720. Far-end Network Region a new one (for exemple 31). DTMF is set to out-of-band
4 - create a new trunk type isdn / service tie / cor set as system. in group member assignments, port IP / Sig Grp the sig created in step 3
5 - create an ip-network-region (for exemple 31) with codec set to 1 (or your default codec set) with all values set to yes excepted RSVP. In inter region set all to 1 (in codec set 1 you must have a G.711a entry)
6 - edit the network region of your clan (set in step 3) and your ip-phone and in inter network region put 1 for the region 31
7 - create a route with your trunk number in line 1 and ITC bothunr
8 - change your partition toute table and add an entry with your route number for whole used PGN
9 - modify your ars ana and add an entry for asterisk extensions using the route pattern set in step 8 / type pubu
10 - add an entry in toll list for your asterisk extensions

=========================================
For the Asterisk Box (AsteriskNow b6 for me) :
____________________________________________________
1 - create your first extension (if in your avaya dial plan you have defined 8xxxxx as asterisk extension, create 800000 for exemple). dislplan 1 / SIP and NAT selected / disallow:all allow:alaw
2 - in extensions.conf in [default] add exten=_2XXXXX,1,Dial(H323/${EXTEN}@IP_OF_A_CLAN:1720)
_2XXXXX : avaya extentions from 200000 to 299999

Now, if you log a sip phone (extension 800000) to your asteriskbox and you call an avaya extension in the range defined, your phone must ring.
From an avaya phone, if you call the extension 0800000, the sip phone must ring
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Postby pspenn » Thu Aug 09, 2007 5:01 am

Thanks for this tip... However, I have done exactly as suggested and I am able to call to and from my Avaya but have no audio either way. I have verified on both ends that the correct codec is being used. Another strange thing that is happening is I am able to hear the DTMF from the Asterisk Box to the Avaya station. But when I hit the touch tone from the Avaya station the DTMF gets "Stuck" coming out of the audio of my softphone. (Example: Make the connection between the softphone through the Asterisk to the Avaya and the IP phone on the Avaya... Then hit any touch tone key from the softphone. You will hear it on the IP phone on the Avaya. When you hit a touch tone key from the IP phone on the avaya for just a brief second, the touch tone gets stuck and will not stop the touch tone audio coming from the Softphone until you hit another touch tone on the softphone. Then touch tone will stop working between the two phones.)

Sorry if I sound crazy... Thanks for any assistance.
Perry
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Postby gollum93 » Thu Aug 09, 2007 1:33 pm

ok you seems to have a route problem.

When you call your sip asterisk phone from Avaya phone do a status station XXXXX (XXXXX is your avaya station extension ;p )
look on the second tab and check the IP of voice ressources that are used.

Then from your asterisk console, do a ping to the voice ressource used by your avaya phone. if there is no answer then you need to add a route to the voice ressource subnet (if i remember, sudo IP route add subnet/mask gateway)
Then you must heard voice from your sip phone.

do the same test from a pc in the same subnet than the pbx avaya (from your dhcp/tftp server)

Sorry for my english... i'm french....
gollum93
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Postby gollum93 » Thu Aug 09, 2007 1:34 pm

ok you seems to have a route problem.

When you call your sip asterisk phone from Avaya phone do a status station XXXXX (XXXXX is your avaya station extension ;p )
look on the second tab and check the IP of voice ressources that are used.

Then from your asterisk console, do a ping to the voice ressource used by your avaya phone. if there is no answer then you need to add a route to the voice ressource subnet (if i remember, sudo IP route add subnet/mask gateway)
Then you must heard voice from your sip phone.

do the same test from a pc in the same subnet than the pbx avaya (from your dhcp/tftp server)

Sorry for my english... i'm french....
gollum93
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Postby gollum93 » Thu Aug 09, 2007 1:36 pm

ok you seems to have a route problem.

When you call your sip asterisk phone from Avaya phone do a status station XXXXX (XXXXX is your avaya station extension ;p )
look on the second tab and check the IP of voice ressources that are used.

Then from your asterisk console, do a ping to the voice ressource used by your avaya phone. if there is no answer then you need to add a route to the voice ressource subnet (if i remember, sudo IP route add subnet/mask gateway)
Then you must heard voice from your sip phone.

do the same test from a pc in the same subnet than the pbx avaya (from your dhcp/tftp server)

Sorry for my english... i'm french....
gollum93
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Postby gollum93 » Thu Aug 09, 2007 1:45 pm

ok you seems to have a route problem.

When you call your sip asterisk phone from Avaya phone do a status station XXXXX (XXXXX is your avaya station extension ;p )
look on the second tab and check the IP of voice ressources that are used.

Then from your asterisk console, do a ping to the voice ressource used by your avaya phone. if there is no answer then you need to add a route to the voice ressource subnet (if i remember, sudo IP route add subnet/mask gateway)
Then you must heard voice from your sip phone.

do the same test from a pc in the same subnet than the pbx avaya (from your dhcp/tftp server)

Sorry for my english... i'm french....
gollum93
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Postby pspenn » Fri Aug 10, 2007 6:07 am

Thank You very much for the advice. Your English is just fine. I took French in school and remember VERY little.

I did what you suggested but everything looks normal. I was able to ping all of my voice resources with no problems. One item I did notice was on the source to destination talk path.

My Avaya IP Phone: 172.16.12.47 connects to the Avaya Media Resource: 10.2.3.99 which is supposed to connect to the Asterisk IP Address with h.323 trunks. However, the IP address that the Media Resource is connecting to is 127.0.0.1. Is this correct?

Thank You for your help.
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Postby gollum93 » Sat Aug 11, 2007 2:18 pm

no ! 127.0.0.1 is a loopback address.

how is it possible ??

and if you ping your asterisk box from your S8700 through on of your CLAN it's ok ?? when you call a sip phone from an avaya phone what do you see in channel ? (in the asterisk gui)
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Postby pspenn » Mon Aug 13, 2007 8:43 am

I can Ping the Asterisk box no problem from the C-lan. As for the Loopback address... I was able to see that the call comes in from the Avaya IP phone in to the Media Resource Card and then back out of the Media Resource Card and out to the proper h.323 trunk using one of the trunk members.

The connection is very well defined. It is just the audio portion that is not making it. When I make a call from SIP to Avaya I see the channels come up properly in the Asterisk GUI.

I had our networking people do a port scan on the Asterisk box and they saw port 1720 open but there were no other h.323 RTP ports open. Is that because port 1720 is all that is needed until the call is established and the RTP ports will open when the call is set up?
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Postby pspenn » Mon Aug 13, 2007 12:15 pm

First let me say Thank You so much for your help.

After some trying and testing I found that I needed to put the IP Address of the Asterisk Box in the bindaddr= section of the h323.conf file.

I am working perfectly now!
Thank You again for the help!

Perry
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Postby gollum93 » Tue Aug 14, 2007 6:25 am

sorry but i do not use bindaddress in h323...

Very happy that your solution works.

If you want ask me a question about avaya or asterisk directly, give me an icq number or msn messenger account by pm
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Can you please post your config

Postby daye » Wed Aug 15, 2007 2:34 pm

Hi, Thanks, I am interested in making it work. Can you please post your config files, especially H323.conf.

Thanks.
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asterisk+avaya configuration

Postby sravanthi » Mon Jul 07, 2008 8:07 am

hi
i just need help in making calls from asterisk clients to avaya clients through h323 trunk.i'm able to call from avaya clients to my asterisk (registered) soft phones.but the other way,like from asterisk softphone clients to avaya clients its not working .even ring is not comming to my avaya clients(hard phones).can anyone help me in this regard.any help is greatfull.
thank you.
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Re: AsteriskNow works with Avaya PBX (H323 IP Trunk)

Postby dekbh99 » Fri Oct 29, 2010 10:36 am

gollum93 wrote:Hi,
Just to announce that AsteriskNow b6 works with Avaya PBX through H323 IP Trunk.
I use it as answer machine and sip<-->h323 gateway
So i'm able to call avaya phone from sip softphone and sip softphone from avaya phone
If you want any help for the configuration of your avaya pbx system and your AsteriskNow, do a reply to this post



Can I see your config files from both the Avaya and Asterisk?
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