similar to: Re: Avaya IPSoftPhone

Krutaw, do you have the list of DLL's you had to register to get the Avaya IP softphone to work? Can you also tell me which version of IP Agent you got to at least load? I'm getting a tapi error. I'm knew to Wine. Just looking to see if someone has done this before.

Has anyone had any luck getting the Avaya IP SoftPhone to work in wine? I get these errors on Ubuntu 8 Code: preloader: Warning: failed to reserve range 00000000-60000000 wine: creating configuration directory '/home/microchp/.wine'... preloader: Warning: failed to reserve range 00000000-60000000 preloader: Warning: failed to reserve range 00000000-60000000 err:dosmem:setup_dos_mem

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:

Hi All, I need help with integrating AVAYA 8300, the avaya can do outbound calls but cannot do inbound calls, im sending calls from sip to avaya using E1 ISDN line. My config was based on aspect dialer it's working with aspect but not with avaya. My config and error is below. zaptel.conf span=1,1,1,ccs,hdb3 bchan=1-15,17-31 dchan=16 zapata.conf group=0 context=avaya switchtype=euroisdn

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

On Wed, Jun 16, 2010 at 4:35 PM, Shina Owolabi <shinacalypse at gmail.com>wrote: > Hi! > I've installed Asterisk 1.4.32 with freepbx-2.6.0 in an attempt to provide > a conference bridge for an existing Avaya PBX. I have no control over the > Avaya system, but I am able to speak with the admin in charge when I need > stuff done. I am running all this in a VirtualBox

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

Hello, I'm trying to create a H.323 trunk between Asterisk 11 and Avaya. I have done this before between Asterisk 1.6 and Avaya but had some issues placing external calls from the Asterisk to the Public network which is connected to Avaya. I'm trying to create that trunk on Asterisk 11 because the 1.6 is outdated and has no support. On the Asterisk side I have Aastra 6731i SIP phones

I'm trying to connect an Asterisk system to an Avaya Partner ACS R6 system. The problem I'm having is that I cannot get the partner system to get CallerID over the T1 modlue. The partner is using the T1 with E & M signalling (which I don't think can be changed), and whatever I tried didn't work. My only option right now is to get FXS ports on the Avaya side plugged into the

I'm using the latest SIP firmware from Avaya. The phone receives the 46xxsettings.txt OK, and then after entering extension and password it goes to the home screen saying 'Registering'. When I check options->ViewIPSettings->IPAddresses on the phone, the registrar and SIP Proxy fields are blank. I have both lines: SET SIPREGISTRAR "67.1XX.XX.XX" and SET

I am not interested in open source solutions. I want to know how much the propriety systems cost in terms of licensing. Specially Avaya now a days per extension. Exclusive or Inclusive of the hardware for 10 agents as noted. Thanks On Thu, Sep 2, 2010 at 8:18 AM, Muhammad Shomail Haider <msh0786 at gmail.com>wrote: > Hi Bruce, > > It all depends what exactly you are in need of. A

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]

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 --

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]:

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

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

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

Dear all I have Asterisk which is connected with avaya through E1 back 2 back now i have on asterisk side G.711 codec and Avaya also useing G.711 codec everything fine. I need G.729 on my asterisk side. can i have lots of SIP phone on my lan and issue is i have 2 to 3 building so problem is LAN is congested thats why i need G.729 Now testing perpose i have download

Hi List, I would like create a H323 trunk from Avaya IPOFFICE to Asterisk, but i would like activate a "direct media path" for the RTP transit directly between the phone and the Asterisk. Now, - H323 Trunk is OK - RTP from the phone transit directly to Asterisk (activate "strictrtp=no" in rtp.conf, and "Allow Direct Media Path" option in Avaya Ipoffice) H323: Phone

I have a connection of Asterisk with Avaya by H.323 and so far everything worked well because only sent to Avaya. Now, the matter is that from Avaya will send me an IVR calls to capture credit card information, the link is active on Avaya 23 channels which is not how to configure Asterisk for those 23 simultaneous channels of Avaya's collect asterisk. Do not know if I can be with a group or