Asterisk and AVAYA
by starprowler on Aug.15, 2009, under VoIP
Some info I posted on ChipOverclock’s blog:
Thot i’d post my experiences with the AVAYA SIP phones.
- The current version of the firmware for the 46xx phones still doesn’t seem to support the message waiting indicator (MWI) light. And as a side-effect of pending voice mail, these phones seem to think they are not registered after a seeming random period of time, refusing to allow outgoing calls. I had to disable voice mail on all extensions with AVAYA phones. Wasn’t too bad, I could live without the MWI.
- The 1.xx version of SIP firmware for the 9620/9630/9640 series of phones would register fine on Asterisk, allow outgoing calls but refuse to accept incoming calls.
- AVAYA have recently release version 2 of the SIP firmware for the 96xx phones. Thought I’d give it a try the other day. I was pleasantly surprised to find that it worked perfectly, including the MWI!! Excellent speaker-phone, and great voice quality.
I’ve been testing it for about 4 days now and am totally happy with the results.
November 19th, 2009 on 11:59 pm
We are trying to do the same (9620 phones to asterisk) and being unable to connect. Already loaded one phone with the latest sip firmware but the best we have is an “acquiring service” message after login…
Can you give us more details?
Thanks!!
November 21st, 2009 on 12:07 am
I used the SIP firmware version 2.0.3.0 and manually configured the parameters. Most of these parameters can be configured automatically using a combination of DHCP options and the 46xxsettings.txt file.
Did you configure the SIP parameters on the “Admin procedures” menu of the phone?
December 15th, 2009 on 10:23 pm
Can you please tell give the details of what i DHCP you configured for the Avaya 9620 to work with the Asterisk system. I am new to all this Asterisk stuff but very familiar with the Avaya setup.
If you would could you please email me a settings file?
Thanks
December 15th, 2009 on 10:39 pm
Alan,
You should use the DHCP option 242 for the 96xx phones and configure the HTTP server as you would with a regular H.323 AVAYA installation.
The 46xxsettings file will contain the bulk of the configuration parameters. I could post a 46xxsettings file here. I’m on vacation and don’t have immediate access to the settings file.
Needless to say, you need to load the SIP firmware on the 9620 to get it to work with Asterisk.
December 18th, 2009 on 5:18 am
Thanks. I have my settings file, and option 242 in DHCP now with the settings as option 242 ascii “sipsrvraddr=10.10.10.10,httpsrvr=10.10.10.2″
Either by using DHCP and the settings file or doing everything manually here is what I am seeing. On booting of the phone it flashes by really fast but I see dhcp then what I think is the phone hitting my http server in like half a sec. Then a long pause of :no call server” After this it goes to the username and login prompt.
I put in my extension and password I set up in the gui of Asterisk Now. And the phone either hands on logging in forever or I get a login error message depending on wether or not I have the phones SIP settings in admin set to peer to peer or proxied.
In asterisk now I set the extension, a sip alias and the sercret, thats it. Xlite softphone registers no issues. I have 9620 on firmware 2_0_5. I can not find 2.0.3 like you had.
On the phone under SIP do I use proxied or peer to peer? What about TLS or TCP? Do I have to set the domain? Avaya enviroment yes or no? User field yes or no?
On Asterisk Now is there any other/special things I need to set?
Thanks
December 19th, 2009 on 10:05 am
Alan,
Set your SIP settings under the Admin menu as below:
SIP Mode = Proxied
SIP Domain = (not relevant. You may use the call server address here or leave it blank.)
Avaya Environment = No
Transport type = UDP
SIP Proxy Server = (IP address of your call server)
User ID Field = No
Asterisk does not need any special settings. If X-Lite is able to register on this extension, the above settings will definitely work.
Let us know how this works for you.
December 22nd, 2009 on 1:51 am
Yep UDP worked. Tried everything but UDP who would had thought. so I got two phones registered now both 9620’s I can call between but if I source from one of the two phones I get no talk path. IfI call from the other phone I get two way talk paths works fine. Wierd. Thaks for the help.
December 26th, 2009 on 9:02 pm
🙂
For one way talk path issues, make sure that reinvite is set to “No” on the configuration of both extensions on Asterisk.
Also might be check your RTP port settings.
January 31st, 2010 on 12:50 am
Help! This was working great with firmware 2.2. Just upgraded to 2.5 and am stuck at “Acquiring Service”. Could someone kindly post a working 46xxsettings.txt?
Any help would be appreciated.
Details
Avaya 9640
Trixbox 2.8/Asterisk 1.6
January 31st, 2010 on 5:08 pm
I haven’t upgraded to 2.5 yet, so I’m not sure. I’ll probably get around to it next weekend. I’ll post my results.
March 10th, 2010 on 11:36 am
Hey guys. Hope this is stillactive. I have an Avaya 9630. I followed all the help I could find about all the settings but no matter what I try I get either “logging in…” or “Aquiring Service” and it never logs in. If I put in a user name that is not in the extension list it does tell me no user found. So I know its at least trying to accept the password and let me in. Please Help!! I can post whatever configs you need to see. Thanks.
March 10th, 2010 on 5:58 pm
Hi,
We’re still around 🙂
A few questions:
1. What version of the SIP firmware are you running on the 9630?
2. Can you post your 46xxsettings.txt file? If manually configured, can you post your settings?
3. What PBX are you running? Asterisk? If so, can you post the console logs?
Its fairly straightforward, and I’m sure we can get your phone working quickly. 🙂
March 16th, 2010 on 6:01 pm
The firmware version IS important – from what I remember (already did this some months ago), I tried with the latest version and it didn’t work but when I flashed the 9620 with the fw 2.0.5.0 it worked right away.
Just my 2 cents 🙂
Best,
Nuno
March 19th, 2010 on 9:04 am
Totally agree with you. Thats the reason I stopped upgrading to the newest firmware when I found one that works.
November 30th, 2010 on 12:20 am
Can you tell me where the older firmward can be found? I have not had any success with 2.5 or 2.6.
November 30th, 2010 on 11:42 am
Try this link:
http://support.avaya.com/css/appmanager/public/support/Downloads/P0553
Select the version you want to download from the drop-down.
March 16th, 2010 on 9:17 pm
I’m having an issue though.. I’m unable to pickup another extension call with *8 in a avaya phone. In a softphone it works ok. For some reason the phone isn’t passing the sequence to asterisk. Any ideas?
March 19th, 2010 on 4:57 pm
Solved – don’t try to use functions that begin with *. It will not work (at least with the mentioned fw). All I had to do was to change the codes just to numbers (for example, where *8 would pickup a call I changed it to just 8).
Cheers,
Nuno.
March 19th, 2010 on 8:07 pm
Thats a useful piece of information. I’ve never tried call pickup, so did not know. Thanks for posting back, Nuno.
March 19th, 2010 on 9:45 pm
Can confirm that version 2.5 works beautifully with Asterisk. have tried various other Avaya firmwares for the 9630, and they each had different issues, such as logging out of Asterisk, or not having option enabled under menu.
March 20th, 2010 on 1:07 am
Hi
I installed today the same firmware 2.5 on the 9620 but i get authorization failure on the asterisk Server , with the older firmware was all ok.
March 20th, 2010 on 6:25 pm
Thats odd. Can you post the Asterisk debug logs which show the auth failure?
March 21st, 2010 on 1:24 am
Hi , 2.5 is runing now, but i go back to the old Release.With the new one i can´t edit the Contact list.
I read in the Avaya PCN that is since Rel.2.4 with other SIP Server as SES normal.
January 27th, 2011 on 1:26 pm
Is it possible to run 2.5 oder 2.6 with Asterisk using just a customized 46xxsettings.txt or are changes to the Asterisk server needed?
January 27th, 2011 on 9:19 pm
Sure, its possible, both versions of the AVAYA firmware are compatible (within limits.)
January 31st, 2011 on 3:45 pm
Hi starprowler,
thank You for your reply.
V2.2 registers fine, with any newer release it gets stuck on “Acquiring Service”. Where do I have to fiddle in the config file to make them work?
February 1st, 2011 on 9:45 pm
AFAIK, there is the SIP_CONTROLLER_LIST parameter and SIPPROXYSRVR in 46xxsettings.txt that would likely need to be configured with the IP address of your Asterisk server.
February 17th, 2011 on 5:01 am
Hi
I have the same problem! I have two Avaya phones.. a 9620 and a 9620C. The only way I can get the phone to work is using firmware 2.2. Anything above (2.4/2.5/2.6) and it hangs on “Acquiring Service” or “Logging In”. I get no messages on the asterisk console at all however, if I log into the phone with a user that doesn’t exist I get 403 error on the phone and a message on the asterisk console saying “peer not found” which means the phone is communicating with the server!
Rolling back to v2.2 is perfect however this will not load on the 9620C phones!
Any help would be gratefully received and could save another sleepless night!
February 24th, 2011 on 12:06 am
Hi,
Using a 9630 with asterisk everything is fine…
With SIP 2.6 it worked great but no contacts!!!
So revert back with SIP 2.2 and it works very good and the contacts too.
There are a lot of parameters in the 46xxsettings.txt. But for the same models they can be different parameters depending on the sip version (2.2, 2.5 & 2.6 or 2.4). So be sure to set them all with a correct value, even if you think it’s redondant!
My first try whith 2.6 was very difficult to achieve and i ended finishing the config with the “craft” menu (and then SIP settings/proxy settings) and bingo asterisk tells me that the peer was ok!)
Anybody know of a good “contact manager” program in WML/PHP that could be used with a SIP 2.6 version (because the local application do not work).
These Avaya are wonderful phones with a amazing sound quality!
February 24th, 2011 on 11:37 am
I think the important thing is this note from the file 46xxsettings.txt (but, as i write before, it can be set manually using the admin “craft” menu when the phones hangs on on “Acquiring Service” or “Logging In”) :
## SIPPROXYSRVR sets the IP address or Fully-Qualified
## Domain Name (FQDN) of the SIP Proxy server(s). The
## default is null (“”), but valid values are zero or more
## IP addresses in dotted-decimal or DNS format, separated
## by commas without intervening spaces, to a maximum of
## 255 ASCII characters. (For 96xx SIP models, this
## parameter also may be set either via LLDP or PPM.)
## Note: This parameter is supported on 96xx SIP Releases
## 1.0, 2.0, 2.2, 16CC and 1603 SIP telephones only. For SIP
## releases 2.4.1 and later this parameter is ignored and
## equivalent functionality is supported using SIP_CONTROLLER_LIST.
## Please see SIP_CONTROLLER_LIST parameter for details.
PS: the admin craft menu is always available with these keys on the phone ==>
MUTE, “2”, “7”, “2”, “3”, “8”, “#”
MUTE
February 26th, 2011 on 1:55 am
I was led to your very interesting rambles from my own problem with 9620L handsets, but in joining them to a Draytek 2820 IPPBX. Various firmwares on the handset seem to get me to various points, but never quite far enough, and never beyond ‘acquiring services’. I have confirmed that my login is actually being validated using certain firmware by deliberately using incorrect logins to verify, but that is as far as it goes. The settings listed above for use with Asterix take me to the service acquisition state, but never beyond it, or should I be leaving it for hours?
Many thanks.
March 2nd, 2011 on 12:15 am
I really appreciate the info I’ve found here. I’ve gotten close to making this work. Running 2.2 on a 9640. It registers, and I can make outgoing calls, but incoming calls are met with a busy tone and a “Busy/Congested” message in the Asterisk console. On a somewhat weird note, when I login on the phone, the extension seems to be listed three times… More concerned with the incoming calls, though, if anyone has any ideas.
March 8th, 2011 on 12:28 am
I have just been given a 9630 to play with, I can’t get the 2.2 FW the download links seem to have been removed from the avaya site. Tried 2.5 and 6 but I get an error relating to an empty proxy list when trying to login.
Can anyone share 2.2 or have any thoughts on the empty proxy list error ??
Best
Steve
March 8th, 2011 on 2:23 pm
OK, Got the 9630 running with Asterisk 1.6 all I need now is to find out about how the softkeys / voicemail keys are programmed…. Can anyone help with this ?
Thanks
S
March 10th, 2011 on 7:50 pm
Steve,
The number for the voicemail key can be set using the line “SET MSGNUM xxxx” in 46xxsettings.txt (where xxxx is your voicemail extension number)
I’m not sure the softkeys are programmable, although I could be wrong.
HTH.
March 10th, 2011 on 10:29 pm
Thanks SP,
I found that one, the phones are great but I really wished I could program the softkeys or the round buttons on the right of the phone..
Best
Steve
March 11th, 2011 on 9:47 am
If I remember right, there is a version of the firmware that does allow you a certain level of control on the softkeys and the Call Appearance buttons (the 5 buttons on the right side of the screen on a 9630 and 9640), although that version had a different problem which made me go back to an earlier version. 🙂
March 25th, 2011 on 3:03 pm
Question : Does the message light work on your phones when there is a waiting voicemail ? Mine does not work with Asterisk….
March 25th, 2011 on 6:40 pm
Yes Steve, it does work perfectly fine. I’m on version 2.2.0.7 on the 9630.
March 25th, 2011 on 9:51 pm
Hmmm Me too, but it’s not working is there something in the 46xxsettings.txt file that needs switching on ??
July 26th, 2011 on 10:50 am
in dhcp option 242, “MCIPADD=[some ip addr]” setting has me confused. When I remove it, my 9630 gets stuck in “Waiting for LLDP,” but when I set it to any ip on my network (router, WAN, and even sip server), phone gets stuck at “Discover [whatever IP I used]”
July 26th, 2011 on 11:09 am
Solved part of my problem–had h323 firmware instead of sip 2.x.x firmware. For reference, both can be downloaded from Avaya’s website (see link in comments above)
August 1st, 2011 on 12:01 am
Flashed a 9640 to v2.6 SIP and played with settings. Got it running against Gemeinschaft, Asterisk and finally my intended Fritz.box.
Okay, it can’t access the mailbox and address book. But else, it is working nicely. The original 46xx config can also be found on the Avaya website, next to the firmware. It’s well commented and I just played with the values until it worked.
January 11th, 2012 on 5:54 pm
any chance you can post your 46xxsettings file so I can see what is required ? Thanks
January 11th, 2012 on 11:28 pm
Bernie,
Here’s the 46xxsettings.txt file I use with my setup.
The internal IP address of the Asterisk server is 192.168.2.60.
November 16th, 2011 on 3:14 am
Anyone still reading this – try changing from nat=yes to nat=no on your extension. That’s what I needed. So my setup is as follows:
Manual setup:
none (you may need to set a VLAN or something)
DHCP setup:
use option 242 to set HTTP server (using a cisco router: option 242 ascii HTTPSRVR=10.11.12.5) – this can be done manually instead
46xxsettings.txt setup:
SET DISCOVER_AVAYA_ENVIRONMENT 0
SET ENABLE_CONTACTS 0
SET SIP_CONTROLLER_LIST “10.11.12.101:5060;transport=udp”
Good luck! I have the current latest SIP 2.6, running the current latest Trixbox on Asterisk 1.6.0.26, connecting to Avaya 9620L phones.
January 11th, 2012 on 2:26 am
All gerat comments guys, as this seems to be the only place with any reference to how to get this to work. Would someone with a 9630 , firmware v 2.6 and a working asterisk server like to post their 46xxsettings.txt file so I can see where I’m going wrong.
Thanks
January 11th, 2012 on 11:17 pm
Bernie,
What are the issues you have encountered? That might help us point you in the general direction of the possible solution.
Cheers.
January 12th, 2012 on 3:01 am
Hi,
just been looking into the MWI problem further and from the asterisk CLI, sip show mwi and sip show subscriptions both give empty results with 2 phones registered. Is that right ?
January 12th, 2012 on 1:23 am
Christopher’s new comment is consistent with what I’ve just discovered. nat=no will make the ‘aquiring service’ message go away on v2.6 on a 9630. So I am am up and running now. 2 things left to sort out are to get the MWI working and need some sort of contact application to work. A notify message is sent from asterisk if I set subscribemwi=yes in the conf file but the 9630 doesn’t seem to like it. Any ideas ?
January 31st, 2012 on 12:15 am
Helpful thread! I’m having trouble with mwi too. on an Avaya 9620 SIP2.5 (tried 2.6 also) to Asterisk 1.6. Asterisk gives the warning:
— Got SIP response 400 “Bad Request (Unknown Subscription State)” back from 156.26.xxx.xxx
Would love to see a snippet of sip.conf for a working extension, here is mine:
[8788]
type=friend
username=8788
secret=
context=local
host=dynamic
disallow=all
allow=ulaw
allow=g729
mailbox=3169788788@default
callerid=8788 *2 battin
canreinvite=no
Thanks!
May 3rd, 2012 on 10:28 pm
DSnook,
I get that message as well, I figured it happens when the phone tries to download contacts or possibly requests something AVAYA specific from Asterisk. Haven’t bothered to analyze it too much, as it doesn’t interfere with the functionality.
February 22nd, 2012 on 10:00 pm
Sounds like the MWI problem in the phones isn’t really fixable. Asterisk expects to push MWI notifications, but Avaya expects to pull them.
Read more on quetwo’s post here: http://www.avayausers.com/showthread.php?t=16720
I’d really like to know if anyone finds a solution though.
May 3rd, 2012 on 2:51 pm
well, not having much luck at the moment getting my 9630 to work with my Elastix PBX (Asterix)
Managed to get the SIP firmware update on (2.6) and now stuck om accquiring service.
Will try a different SIP firmware update have 2.2 to 2.6
Any help appreciated, i.e where do I add the line NAT = NO, I dont see this in my current 46xx settings file
If any one is still here it would be good to chat
cheers
Ed
May 3rd, 2012 on 10:33 pm
Ed,
NAT=NO should feature under the context of the extension in users.conf
May 1st, 2012 on 6:30 pm
HI All
Just getting into Asterix, have installed Elastix 2.3 and have a basic working system connected to my Voip provider (IAX trunk).
I have around 15 Avaya 9630 H.323 IP deskphones and 5 Avaya 1608 H.323 Ip deskphones.
I have been looking for a method to update the firmware to the SIP version.
I have been reading this ‘ramble’ with keen interest and I am in the process of documenting all the steps I require to update to the SIP firmware.
I have been on the Avaya support site and found various SIP firmaware files, V6-sp7 seems to be the latest.
Does any one operate a 9630 with SIP V2.6-sp7?, if not can any one let me have anu older versions that work (V2.2)
I will see how I go but I have a feeling that I will be back: 🙂
Many thaks
Ed
May 3rd, 2012 on 10:40 pm
Ed,
I use the SIP firmware version 2.6.5 on my phones
October 3rd, 2012 on 9:11 pm
I have asterisk 1.8 with a 9650 phone with 2.6.8 sip firmware. I’ve set the 46xxsettings file up, the phone is working, but 1/2 of the outbound calls from it are getting fast busy signal. It’s every other outbound call. Any recommendations?
October 3rd, 2012 on 10:33 pm
I’ve never come across this strange behaviour. Best bet is to do a debug on the asterisk console and analyze the SIP conversation. That should point you in the right direction. You may post the logs here if you wish, and I can take a look at it.
October 4th, 2012 on 11:54 pm
Nevermind! I moved back to 2.5 from 2.6.8, and the trouble cleared right up immediately(regarding 1/2 the time outbound getting fast busy).
March 25th, 2013 on 8:46 pm
I ran into this same issue.. looking at the sip headers, it appears that the phone is reusing the NONCE for the second call, and does not retry when it gets the unauthorized header. Lame.
This is on 2.6.9.
October 10th, 2012 on 10:29 am
Ran into a new issue. Not the same as the last. I’ve upgraded to asterisk 1.8, freepbx 2.10 and incredible pbx 4.0. On an Avaya 9650 phone with 2.5 and 2.6 firmware, the hold button produces a busy signal in the receiver, while the far end hears music on hold. There is no way to get the call back because as soon as you hang up, the line disconnects, as if it is not fully putting the call on hold. So hold is broken, but MWI works now. Oh and btw, 2.6 still gives 50% fast busy on all outbound calls. I have logs if needed. I did see a 401 unauthorized
October 11th, 2012 on 1:48 am
To be more concise, 2.6.8 gives fast busy 50% on outbound, 2.6.7 is clear of that. I’ve troubleshot further, and it appears that when I make a call from the 9650 phone(regardless of firmware) the call control soft buttons such as conference, transfer and hold work. It’s when it receives an internal or external call that the buttons will cause a flashing red light on the call appearance that’s affected, the phone receiver or speaker gets a busy signal, and the far end hears music on hold, until they are cut off(15 or so seconds) from the disconnect timer. Also if I hang the receiver or press the speaker button to hang up, it immediately hangs up. I’m not able to get back to the line that called, it just hangs it, until it disconnects or I hang up. traces show music on hold turning on, but then disconnecting. Any takers?
October 24th, 2012 on 3:40 am
Can anyone post a working 46xxsettings.txt file example?
i’m stuck in “acquiring service” , it is very annoing because i’ve tried all firmware versions with the same results.
I’m thinking now that the problem is not in the phone configuration.
Any suggestion to troubleshoot this?
October 24th, 2012 on 6:29 am
solved addind nat=no to sip.conf
October 26th, 2012 on 1:56 pm
hi,
had the same issue again “acquiring service” after a router (where asterisk and configuration was located as well) and my search on google placed me again at this site.
i tried everything, but the important hint is missing: in asterisk configuration 2 ports are defined, one for sip and one for tcp (bindport / tcpbindport).
at this point i assumed that the tcpbindport is used when using tcp-protocol. but this is NOT the case, instead the bindport is still needed to be configured! after that everything was fine.
next issue: language (Mlf_German.xml is not used…?).
regards,
andre
October 26th, 2012 on 2:51 pm
solved the language issue as well: signature didn’t match the language-file. now it works.
regards,
andre
December 13th, 2012 on 10:15 pm
Hi,
I just recently acquired two Avaya phones a 9630 and a 9641G. I was able to get both of upgraded to the SIP firmware. the 9630 2.6.7 and the 9641g 6.2.0. Anyways I was able to use the 46xxsettings.txt to configure them both to use voip.ms.
I also use Milkfish on my WRT54G to act as an outbound proxy which I have to say is the only was I was able to make the Avaya phones register otherwise I was greeted with the lovely “Acquiring Service” message. I am sure because it handles the NAT that is why it works. Everything seems to be working except the MWI function. In Xlite I can see the notifications but with the Avaya phones I do not see the notifcations.
Now I know I am not using this with Asterisk but I have poured though many forums and kept an eye on this site to see how to get this to work. I have tried to change the SET MWISRVR to both my outbound proxy 192.168.1.1 and also my provider proxy and both have not worked. From my research the provider does support the SIP NOTIFY function but not sure how to make it work with my setup.
Lastly does anyone know if there is a way to configure more than one line on these phones much like a cisco 7960? With Milkfish I can configure internal lines to each phone can have an extension but that does not permit you to call outside lines so if I had the option to choose the lines that would be great.
Also with regards to the NAT=no in the sip.conf. Is that another file independent of 46xxsettings.txt or is it a setting contained within that txt?
Thanks,
Jameel
January 9th, 2013 on 11:40 pm
Jameel,
As far as I know and have tested, the are only one or two versions of the Avaya 9600 SIP firmware that handle the standard MWI NOTIFY on the 9600 series phones. The newer firmware versions don’t work well with the MWI notify message.
The NAT=yes is a parameter is on the Asterisk extension configuration, not on the Avaya.
Hope this helps.
January 28th, 2013 on 11:48 pm
Thanks for the follow up with that. I know that my cisco 7960 works with the MWI and also Xlite. The SIP trace and the NOTIFY SIP message is displayed:
——
NOTIFY sip:[email protected]:5060;avaya-sc-enabled;transport=udp SIP/2.0
Via: SIP/2.0/UDP 174.137.63.206:5060;branch=z9hG4bK2f28a703;rport
Messages-Waiting: yes
Message-Account: sip:[email protected]
Voice-Message: 1/0 (0/0)
——
So even if I set the “MWISRVR” setting to a) VOIP.ms server or b) SIP router address I do not seem to get this working. Should I be using that setting? I have a mix of 9630 and 9641G’s so what firmware would you recommend in the “2.x” range for the 9630’s and what for the 9641g (or is this phone SOL for MWI).
Thanks
January 6th, 2013 on 12:25 am
Hi every one, anyone tried firmware 2.6.9 ?
I’ve read that there is a auto-answer functionality maybe some paging/intercom possibility?
ps : still missing/searching a good WML contact application…
January 9th, 2013 on 11:43 pm
Haven’t tried it yet….hmmm, maybe something for the weekend 🙂
January 19th, 2013 on 1:42 pm
Hi,
Just tried the 2.6.9 firmware and nothing extraordinary…
I can’t even make work my message button (SET MSGNUM 1234) !
I see some exchange info (but i think they were here on 2.5/2.6)
I have problem’s with the push functionality (Push Status: 402 Push security failure) when tried to use the /ScreenSaverSample/pushScreensaver.html samples !
I’m going to return to my good old 2.2/2.4 ….
January 19th, 2013 on 6:27 pm
😀
Agree, the older firmware was more compatible with Asterisk. I’ve done the same with some of my phones.
August 3rd, 2013 on 2:23 am
Starprowler, did you get the Transfer, Conference, Hold buttons working? Thanks!
August 19th, 2013 on 12:57 am
Yes, the Transfer, Conference and Hold buttons work… or used to work when I last tested it. AFAIK, only the MWI light does not work. So you have no visual indication that you might have voice mails waiting.
August 9th, 2013 on 12:23 pm
Hi guys – HELP – HELP
Is it possible, to connect an avaya 9650 SIP with a Fritzbox from AVM? My Fritzbox gives the username for avaya 620 with a personal password.
Can someone help me to configure the avaya phone?
I’m stuck in “acquiring service” 🙁
Thanks
August 19th, 2013 on 1:01 am
I’ve never tried the AVAYA with the Fritzbox, so not sure of the SIP implementation on it.
Configuring the AVAYA phone is quite simple, by pressing * when the phone boots up and manually entering the IP and SIP parameters.
August 20th, 2013 on 7:13 pm
Use the SIP Firmware 2.2.0 and it will work with the Fritzbox.
September 3rd, 2013 on 9:55 am
ciao a tutti , per far funzionare i telefoni 9608 avaya devete togliere la spunta nat dal utente zip de asterisk e funziona no dice più servizio di acquiring