similar to: Aastra 9133i Phone and MWI

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

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, 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?

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

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

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 --

MessageMelissa - I added the "fromuser=AnyName" to my sip.conf file stations and that, in fact, corrected the problem. The MWI now works flawlessly. I would recommend that Aastra/Sayson pursue this with the Asterisk team so that it is listed as a known issue or to have Asterisk patched to fix it. I will submit a bug on it as well. I'm copying the Users Mailing List on this. For

Is it possible to control the light on a programmable button without the blf option? I'm using a programmable button to turn call recording on and off and I would like the light to indicate the status. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david at safedatausa.com

Hello. Just rx'd the sip - aastra 9133i. Haven't done sip before. My initial attempt at setup has failed. "No Service" Anyone want to contact me off-list or on irc ? Regards...Martin

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

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?

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]:

Hi Guys, Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i, and 6730i, but none of them indicate the voic-email. Where should I look for trouble to find the root issue for MWI? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100714/d268d1f8/attachment.htm

I need to setup MWI on a few Aastra 9112's. I've tried doing so in the web interface by setting "Explicit MWI Subscription" to true, but no lights, no stutter tone. Firmware: 1.4.0.1048 Thanks! -- Warm Regards, Lee

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.

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

Having scoured the web, I still am no better off .. I have 2 Aastra480i's , and 120+ cisco 7940's :) . I am trying to decide which model to use going forward when we purchase more kit. They both seem much on a par regarding features. Q1: Is there anyway of making the cisco auto-answer _without_ having to manually edit the configuration on each phone ? I've been able to get the

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

We have a new customer with a lot of old phones like the 9133i. They won't register, and we see some very strange behavior with them. If the SIP peer exists, they simply fail silently, with no error in the CLI or the messages log. Nothing works, but no errors. If the peer does not exist, it's clear that it's registering improperly: [2013-04-28 13:34:31] NOTICE[3058] chan_sip.c: