similar to: avaya 4602

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 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

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

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

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

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

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

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 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

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]:

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

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

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 just connected an Asterisk server to an Avaya Pbx using the instructions at: http://www.voip-info.org/wiki/index.php?page=Asterisk +Avaya Everything seems to be working as I can send and receive calls. The only detail I am having a problem with is that when an extension on the Avaya calls an Asterisk extension, I only get the name part of the CID, not the number. From Asterisk to the Avaya