similar to: Asterisk with Avaya

Sorry for the basic question - bur I ran into something I haven't noticed before and would appreciate a little more perspective on my problem. I am using R to determine if various thresholds are hit (or surpassed) in a data set.? If a threshold is surpassed, I have had no problems identifying it.? However, when the threshold is matched *exactly*, not all cases are being identified. Please

So far I have not found a way that I can register the Avaya phone with Asterisk. From what I have found so far is that Avaya phone needs the Avaya Media Server and Avaya Gateway. Looking at the h.323.conf (in Asterisk) and the file 46xxsettings.txt (avaya file located in tftpboot) there are no settings to make the phone initialize. I have sent an email to the Asterisk Users Mailing List to see

The company I work for has deployed an Avaya IP phone system. They have deployed the Avaya 4602 and 4620 IP telephones. They might be sending me one of these phones for use in my home office. Question: Can I make this IP telephone register and work with my Asterisk server? I don't know if it is a SIP phone? I searched thru the Avaya site, but can't find whether it's a SIP phone or

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

I would like to connect a Legacy PBX (Avaya IP Office 406) to an Asterisk Server. The Avaya has 3 CO Ports available. I thought buying a TDM30B card with 3 FXS ports to connect the * to the Avaya CO Ports. Is this the right approach? Does any one have experience with such a configuration? Thanks in advance for all recommandations and suggestions.... Daniel Kocher

Hi * Users, I ran into a problem when I was trying to communicate an avaya IP Office talk to asterisk with SIP Trunking. I had successful calls from asterisk to Avaya but not from avaya to asterisk. Can someone provide me insight on how to address it or the path to resolve it. The error I get is mentioned below: (dialing 32564 from avaya to asterisk) "[Nov 6 17:14:23] WARNING[6227]:

Hello, I'm posting this to the list in case others run into the same issue. I've recently been connecting * to a legacy Avaya InDEX switch over E1 ISDN PRI here in the UK. Everything was working OK, except that DTMF digits were not being recognised by * when sent by the Avaya switch to the * system. Instead, the background noise of the call centre would be silenced while

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

I was wondering if anyone can help me with a problem we have at one of our sites. We have setup a Asterisk Trunk to a Avaya PBX, ie ... Avaya <-> Asterisk (1.2.30) <-> External ISDN Network BUT They also have a Polycom VSX 7000 that with some sort of BRI converters that plugs into the Avaya. The Trunk is working well except for Video Conference Calls. The Polocom can receive

I am facing one way audio problem in sip trunking between asterisk and avaya. +-------------+ +----+ | avaya sip |-------| P1 | +-------------+ +----+ | | | +-------------+ | Asterisk | WAN

Hi! Trying to figure out what I am doing wrong... 1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256) 1 Cell phone 00733025975 attached through H323. extensions.conf exten => 975,1,Goto(975-${DEVICE_STATE(SIP/0317998975)},1) exten => 975-INUSE,1,VoiceMail(0317998975 at inputinterior.se,bs) exten => 975-INUSE,2,Hangup() exten =>

Hi I've a softphone account to a AVAYA H.323 system, basically, it has a numeric ID (which is the extension number) and a numeric password. Instead of using the default AVAYA softphone (H.323), can I make asterisk as a H.323 client and login to the AVAYA system via any one of its h323 modules? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am using H.323 to create a trunk between Asterisk and Avaya IP Office system. Everything is working correctly, Asterisk can call Avaya and vise versa. Now I create a conference room with a user pin in Asterisk. Avaya can call into the conference room, but can enter the pin number. The error message they are getting is the "invalid pin number". However the pin number works if they

Hi all,as good? It would like to know if already they had had success, in the integration of the functions of asterisk, with one pabx legacy (Avaya)for that pabx avaya, could use the voicemail of asterisk. For sample, user of pabx avaya, it would have its calls directed for not attendance and busy, for asterisk and asterisk, it would send the same one for the voicemail. Best Regards Josu?

Dear all I have confusion how to asterisk genrate tone and what ringing code use default 180 or 183 i have setup asterisk with mediant 2000 with avaya [asterisk]-----[mediant 2000]--------[Avaya] when i call from avaya side to ---> asterisk i don't got ringback Sound so how to asterisk genrate tone for calling party is there any soution and what is the problem of

Dear all I have asterisk 1.2 configuration and it is working fine but thing is that i have alread Avaya setup and i have intergrate my Linuxbox asterik with Avaya system avaya already use 4 digit dialplan (1644 example ) and in asterisk i have configure 2 digit dialplan ( 44 example ) now i want to configure 4 digit dialplan in asterik without any change in avaya or asterisk so

Hi List, I have another issue on allowing outgoing calls to PSTN on Asterisk via Avaya Phones, I hope that anyone could help me fix this issue: *When I dial through Avaya phone i just here a "good bye message" reply from asterisk server. And here is the log:* == Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so falling back to exten 's' == Starting

Hi, at the moment we have in Avaya Integral PBX with german pri (30 lines). We want to smouthly migrate to an Asterisk server. For this reason: Is it possible to route the external german pri (E1) through Asterisk server to that Avaya PBX? I think at first we need a Digium e1 card 4-Port. But how do we have to configure the routing of the whole PRI? I really would appreciate any sample

Right-- I've seen the Avaya document you cite below. It says "To administer DHCP option 242, make a copy of an existing option 176" but I don't have any example of option 176 or 242 to copy, and don't know what to do to /etc/dhcpd.conf to make it offer option 242. Then there's this long table of parameters to use with (presumably) option 242. I was hoping someone had a

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 --------------