similar to: Problems with 9133i config

This is a single aastra 9113i sip phone. asterisk 1.0.9 Why do I keep seeing this in the logs ? ------------------------------------------------------ Sep ?8 18:44:25 VERBOSE[18779]: Scheduling destruction of call '9157b7d5ef36e8ec556a68e446d4ad59@192.168.1.100' in 15000 ms Sep ?8 18:44:31 DEBUG[18779]: Setting NAT on RTP to 0 Sep ?8 18:44:31 VERBOSE[18779]: 11 headers, 2 lines Sep ?8

Asterisk 1.0.3 Sayson 480i running .78 release (problem may not be Sayson specific, it's just that's what's deployed) Problem: Asterisk rejects registrations every so often even though nothing has changed either with Sayson or Asterisk configuration (and previous registrations have succeeded) SIP trace of successful registration: =============================

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 have a terminal adaptor connected to Asterisk. I found that when it connect to Asterisk v1.0.0,The TA send register msg, and get OK msg, but it get Notify msg every 1 minute, The TA reply with "call does not exist" , usually 10 minutes later, The TA is down. But it work well with Asterisk v1.0-RC1, only register and OK msg are sent and got, no Notify msg is sent to it.

Hi, I tried to cable #21 with a thomson cable modem mta: <-- SIP read from 192.168.153.100:5060: INVITE sip:%2321@195.38.96.5:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.153.100;branch=z9hG4bK1aa77a586 Max-Forwards: 70 Content-Length: 258 To: "#21" <sip:%2321@195.38.96.5:5060> From: sip:15800115@195.38.96.5:5060;tag=da42eb89613306c Call-ID:

Hi, I'm trying to get the match_auth_username=yes sip configuration working. It's mentioned as an experimental new feature of 1.6.2.x. (I'm using 1.6.2.8) The sip.conf example states: ; if available, match user entry using the ; 'username' field from the authentication line ; instead of the From: field. But still I've been unable to authenticate using username

Currently I am having 2 issues with my Avaya 4602 phone: First, the phone registers with my Asterisk server, but when I start dialing I get a busy signal after 4 digits. I specified in the dialplan on the phone to expect 10 digits and that solved that problem, but I still immediately get a busy after the 10th digit. The phone never sends a dial command to asterisk. Second, asterisk is

Hi, I've just (earlier today) updated from CVS so that I can apply the dtmf caller id patches. Unfortunately this has had an undesired effect. I have an intertex ix66 which up until the CVS update allowed me to register my * server with the ix66 for my local domain (eg sip.mydomain.com). Now it appears that asterisk gets totally confused and tries to register with itself! Anyone got any

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'm running asterisk 1.2.6 and aastra phones 9133i with the latest firmware. The MWI light is not coming on on the phones. A) how can I debug if asterisk is actually sending the 'new message' notification? Is it possible I could have this turned off by accident? Has anyone else had success with MWI with 1.2.6 and 9133i phones?

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 --

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

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

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?

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

Hi - I've recently bought a mediatrix 1204 and have had a complete nightmare getting it up and running with an asterisk@home setup. I know this isn't a mediatrix list but I'm at my wits end and the support with this product is atrocious. (mine was even shipped with firmware that was incompatible with the win32 software it came with so I wasted a day trying to work out why the SNMP