similar to: Avaya 5610 SIP Firmware

Does anybody know if Avaya has a test SIP firmware available for 4620 and 4640 IP phones? The 46xx SIP image from their website is a combo download with SIP for the 4602, and h323 for the the 4620 and 4640. It looks like they demo'd a SIP image for the 4640 as far back as 2004: http://www.sip.org/von/2004/boston/slides/DSC_0042.php Thanks, Andy -------------- next part -------------- An

Hi List, Ive got a bunch of Avaya IP9640 that we want to convert to SIP and then hook up to Asterisk so we can dump this overpriced Avaya system. Ive got ahold of the SIP firmware, but I cannot find anything on how to convert the phone itself to SIP, when I go into setup mode it wants a "command" which im guessing is the program code like the rest of the Avaya systems works. Has anyone

I attempted to update an Avaya 4602 phone to the latest SIP firmware and now the phone stops at the bootloader. It keeps requesting an appsip.ebin file from my HTTP server and is no longer checking my TFTP server for update files. Since no appsip.ebin file was included in the firmware update provided by Avaya, I have no idea where I am supposed to get this file. And also since it is no

I just registered an Avaya 9620 set to my Astlinux system (0.47 - Asterisk 1.2.22), using Avaya SIP Firmware version 2.0.1.34. Set mailbox=ext at default in the sip.conf Found MWI worked immediately. Turned off as expected. Have Fun! Tom -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 * 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]:

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

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

Hi all, I'm new to the list, so I apologize in advance if I'm beating a dead horse by asking this, but I read somewhere that asterisk 1.4 has MWI working for Avaya and their rather troublesome SIP firmware. Can anyone verify this before I go changing phone systems around? Thanks Derek -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have an Avaya 4602 IP phone that was previously working with Asterisk. It was being used elsewhere for several months, and I recently set it up again to work with Asterisk. Everything works fine for several minutes -- I am able to receive and make calls as expected. However, after a few minutes, and every few minutes thereafter, I get the following message on the console: -- Got SIP

Hi, I'm trying to disable the internal dial plan of an Avaya 4602 with SIP firmware 1.1 but couldn't find how to do it. Even if I configure a custom Dial Plan it keeps adding other builtin rules to my dial plan. Ex: Configured dial plan: DialPlan 19xx|7[8-9]xx|0xxxxxxxxxxxxxxxxxxx+ Reboot. On the syslog it shows: Aug 17 12:42:12 192.168.0.115 DigitMap:

Hi, I am trying to use an Avaya 4602 phone, which I just updated from a very old SIP software to the latest I could find on avaya's site (032207). The upgrade went fine and it gets registered on the Asterisk server. Now, a couple of glitches, though. - The phone's web server is not working ... so I have no easy way to configure it. It used to work with the old release of the software. I

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

Hi guys, I installed asterisk and it's working really well. For now I`m using soft phones IAX and SIP but now I want to use the regular IP phone, what I have now is Avaya 4624 and I didn't find a firmware for SIP for this particular phone I believe now is working with H.323, can please someone advice me if exist firmware for this phone to register SIP or IAX2 with my asterisk box, and to

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 think I am the first to use the $135 Avaya 4602 SIP phone, but I need some support from the community to fix one problem I have with it. The phone stops working after about 20-30mins if I have mailbox=context in Asterisk; when I do have mailbox=contect in asterisk the sip debug returns "481 extension does not exist." Anyone willing to help me figure out why? I.E. Is it an Asterisk

Hi! Have a weired problem with Avaya 9650 phones: extensions.conf exten => 0317998975,hint,SIP/0317998975 exten => 0317998975,1,Goto(0317998975-${DEVICE_STATE(SIP/0317998975)},1) exten => 0317998975,2,Hangup() exten => 0317998975-INUSE,1,VoiceMail(0317998975 at inputinterior.se,bs) exten => 0317998975-INUSE,2,Hangup() exten => 0317998975-NOANSWER,1,VoiceMail(0317998975 at

I have an Avaya 4610SW IP phone which I have upgraded to SIP firmware. I have successfully registered this phone to Asterisk BE as well as Asterisk 1.4.18 Almost everything is working well. Except for two issues. One of the problems is that the phone looses registration every now and then ad I have to re register. I have found a tip for this which I am testing if it will work, which

I've just interfaced an Avaya 4621 set to *@home. It's running the 2.2 SIP firmware released August 17th. I've run into some strange behavior with this phone. 1st, in order to get two way audio, I had to tell * that it was behind a NAT even though it is on the same subnet as the * server. Second, it doesn't seem to be able to dial any extensions or features that use the *

Hi all, I've been pulling my hair out for two days over this problem... I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem