similar to: sip - aastra 9133i

Hello - I have just purchased an Aastra 9133i SIP phone for testing with Asterisk. Its a little flakey but overall is a far superior phone to the others in the $179 range. I have an issue regarding the message waiting indicator. The phone does not seem to respond to the "NOTIFY" command from Asterisk. Searching archives seems to indicate that this was previously an issue on the 480i

Hello - I have just purchased an Aastra 9133i SIP phone for testing with Asterisk. Its a little flakey but overall is a far superior phone to the others in the $179 range. I have an issue regarding the message waiting indicator. The phone does not seem to respond to the "NOTIFY" command from Asterisk. Searching archives seems to indicate that this was previously an issue on the

I've been pulling out my hair all day on this one. If anyone can help, I'd really appreciate it. :-( I've got an Aastra 9133i (with the latest firmware version) and a Cisco 7960 sitting behind a NAT device on my LAN. The Asterisk server is hosted offsite and has a public IP address. I've set up port-forwarding on the firewall for both phones to tunnel the SIP messages initiated

I've been looking at the two phones Aastra 9112i and 9133i for our offices now. I was just curious if anyone could tell me about the ring tones. Specifically: 1) How many ring tones do they come with? 2) Can you upload your own ring tones? 3) If so, what format and how? (wav, mp3, gsm, etc. and TFTP, web management, flash, etc.) Thanks for any help. -- Tim DeBaillie 812-476-2721

Hi, I have several aastra 9133i phones, which are connected to an asterisk 1.2.6 system. I have setup MWI on the phones to point to the IP of the asterisk server, but although there is a message waiting new in the mailbox, the phone's light does not light. Any thoughts?

Ok, So I just installed some aastra 9133i phones. They work great. One small problem. When I do an 'all page' anyone who is on the phone gets their call placed on hold for the duration of the page! THIS IS NOT GOOD! How do I make the page only go to people who are not on the phone, and ignore people who are on the phone?

hi i've got an AAstra 9133i ip phone, when i've bought it, i've set it to use a SIP/400 account on my asterisk, then, i've changed settings and i've set set phone to use a SIP/500 account . now, when i connect the phone to tthe network, it register itself on asterisk with both accounts!!! -- Registered SIP '500' at 192.168.100.188 port 5060 expires 120 --

Hey - I know there's some other people out there that have the 9133i ... has anyone gotten the DTMF tones to work after the far side picks up? I didn't have any problems out of the box with my SPA-841 phones... the aastra has been nicer so far, but I can't seem to get it to dial the touch tones after an auto-answer device picks up on the far side... I googled, to no avail. -Karl

I am unable to get my 9133i to register with my asterisk server. I am including config files below, this a simple test network so there's nothing secret in the config files. I have upgraded the phone to the latest software version (1.4.3) I'm not sure what the problem is. I can call the phone from a softphone, but the 9133i says "no service" on the screen and I can't dial

There is no release note, just a text file that says AASTRA TELECOM INC. February 2006 FC-000046-01-07.st - 9133i Generic SIP Firmware 1.3.1.1095 for customer release. ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gareth Owen Sent: 15 February 2006 02:00 To: asterisk-users@lists.digium.com

1.6.2.13, sip.conf: [155] type=friend context=longdistance callerid="Admin" <155> secret=test host=dynamic dtmfmode=rfc2833 allow=all defaultuser=155-trust ............ On aastra: Basic SIP Authentication Settings Screen Name Phone Number 155 Caller ID 155 Authentication Name 155-trust Password test But: WARNING[1737]:

I upgraded my office server from 1.4.22 to 1.6.0.5 on the weekend and after some testing there seems to be a compatibility problem when using Aastra phones. With 1.6.0.5 all incoming calls to all Aastra phones were dropped after a minute or so. I installed 1.6.1-rc1 and now the newer Aastra phones (5xi) work properly. The problem remains with the older phones (9112i, 9133i and 480i). If I dial

Aastra Telecom has released SIP v1.3.1 firmware for the Aastra range of IP phones (480i, 480iCT, 9112i and 9133i). The firmware and release notes (no updated admin and user guides yet) are available for download at: http://www.aastra.com/support/enterpriseip Contrary to what the version numbering would suggest, this is a significant update with many new features and bug fixes. See the release

Hello everybody, I'm having issues with calls being dropped on Aastra phones, when the call is on hold. Tested with models 6863i and 6867i. I've figured that the call is dropped by Asterisk when it reaches the rtpholdtimeout limit. I've reported the issue to Aastra, asking them to implement some kind of "RTP keep-alive" feature on their phones. Maybe the phone could send

Thank you Melissa. I love the phone but the dial keypad is a little bouncy. I was hoping for a more solid feel like on the analog PT390's or my quality standard, the Nortel 9417CW. Other than the MWI problem, I'd like more documentation on the configuration paramters. I have found little online configuration documentation other than very basic stuff on the Sayson website. I'd

Hi, Does anyone have experience with Asterisk and the aastra 9112i or 9133i phones? I am looking at purchasing some, and was curious how quality, and stability was with them.

Upgraded from 16.2.14 to 1.6.2.15 on Fedora 13, with aastra 9133i and 57i. On 9133i and 57i: #<extension># works for a blind transfer. Xfer<extension>Xfer doesn't! All this worked on 1.6.2.14. Nothing useful on cli, verbose 3, DEBUG. Here extension 169 answers an outside call, and tries to transfer it to 145 using the Xfer button: -- SIP/169-0000009c answered

I have a customer that has recently upgraded their network and now their Aastra 9133i phones are loosing their connection to the Asterisk server. They were using an external Asterisk server and now we have installed a new internal server with Asterisk 1.4.8 on a SIP/IAX implementation with no Zap cards. Only hard phones seem to be having problems keeping a connection to the server, soft phones

Fellow list members, I have run into an issue where I encounter a delay at the beginning of a phone conversation when I make outgoing calls through Asterisk with an Aastra 9133i hardphone. This is most noticable when I call a voicemail system with the Aasta and then with a land line or other VoIP phone. The first word or two of the voicemail message is generally cut off. According to

Hi, I need to make use of BLF feature on Aastra 6757i phones but its an Asterisk 1.4 using realtime architecture. Extensions are defined in realtime database and dial plan is in AEL. I am able to correctly setup hints in the dialplan, but they don't work. Did some research and found out that hints don't work work with realtime extensions. Is there any work around? On voip-info I read