similar to: Aastra phone dial plan

I want to use extension numbers that begin with the # key in my dialplan, but I can't get my Aastra phone (6731i) to transmit the # key to asterisk. It works fine for the * key. I've tried numerous Local Dial Plan patterns in the aastra web configuration but none of them worked. My current Local dial Plan pattern is "x+#|xx+*|#x+". Any help would be appreciated. -- Marvin

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

hello i hope i am posting to the right list, i am a totally blind user, and i want to reprogram my aastra pt 480e phone, my friend used the web configurator, but i think he programmed thw wrong codes, a few questions, is it possible to damage the phone by programming it wrong? also, how does one reset it literally to factory defaults? as unplugging it and replugging it back into the wall

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

If you own Aastra phones, here's a group dedicated to your specific needs. BTW - The Asterisk-users mailing list is great but it has way too much volume to be useful for specific problems. It needs to be broken up into smaller more manageable lists. Homepage: http://groups.google.com/group/aastra-asterisk-users Group email: aastra-asterisk-users@googlegroups.com

I have 8 aastra phones that are loosing registration. On the phone gui it says 408 as the registation error after a minute or say they register. In the cli it eill say the phone is now unreachable then it will show it registering then available. At first they did it every hour all the phones. After messing with the experation it does it every 15 nin. Any ideas on how to troubleshoot this? I tried

I've received my Aastra 390 phone. I got the unlock procedure from the vendor, and the services button now shows all four entrys as "<available>". When I give the phone "ADSIProg()" the phone displays: Asterisk PBX download refused Conflict with: <available> The asterisk.adsi hasn't been changed: DESCRIPTION "Asterisk PBX" VERSION 0x02 SECURITY

Hi, Does anyone know of some way to make a light on one of the keys of an aastra phone light up (or go off) by sending it a command from asterisk?

Hi, I have 2 phones AASTRA 57i with Asterisk 1.6. When the internet connection for some reason fall down the 2 phones go to "NO SERVICE", searching on internet i found that this is due to DNS service. Has someone solve this problem? or suggestions? Thanks in advance Antonio Supera i limiti: raddoppia la velocit? da 10 a 20 Mega! Risparmia con Tutto Incluso: telefono + adsl 20

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

Hi, 2013/4/19 Olivier <oza_4h07 at yahoo.fr> > Hello, > I've just realized that several phones display both caller name and number > while ringing but when on call, caller name is not displayed anymore. > Could you recommend a sip phone that still displays caller name during > phone call ? > Regards > I've been testing Aastra 6757i SIP phone and it appears

Hey guys, We bought a Aastra 480i phone for our boss. This looks like a nice phone, but we have an issue with it. The issue is this: the phone is able to place calls. But as soon as the phone receives a call, you are able to answer it. But then, 2-3 seconds later, the call is dropped. The extension works, because I've put his Gnet phone back until I figure out what is wrong with this

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?

For anyone who is interested I've recently created an Aastra IP Phone config generator. I don't know if one existed but thought I'd create it anyways. It can be found at http://www.lraweb.pwp.blueyonder.co.uk/. If you have any problems or stuff you want adding then please contact at the address listed on the web page. Regards Lee -------------- next part -------------- An HTML

All, I'm using Asterisk 1.6 and using Polycom 500's with SIP firmware 2.1.3. I can not seem to get the Message Waiting Indicator to work reliably (and in my opinion correctly) with voicemail. I've got the following in my phone.cfg: <reginfo> <msg msg.bypassInstantMessage="1"> <mwi msg.mwi.1.callBack="*97"

There isn't much documentation on adsi, but I called NETXUSA (the vendor of my 480e) and they helped me along. My experience: 1. I really had no experience with ADSI so I had (probably still have) some misconceptions on how the configuration is loaded onto the phone. 2. I set the following in my /etc/asterisk/asterisk.adsi (most of this is the stock asterisk.adsi script): ;

Hi Everyone, I am running DD-WRT on a router that feeds about 30 Aastra 6753i. The phones occasionally go into "No Service" mode. The POE switch doesn't seem to be the problem as it's tested fine. I think the router sometimes gives up and comes back quickly. Or something of that nature. However, the connections are maintained if a call is going on because there are peer to peer

Hi, I can't find a way to tailor DHCP/TFTP/HTTP environment so that brand new Aastra SIP phones can be auto-provisioned when config files are stored in a specific TFTP subdirectory instead of TFTP root directory. For instance, TFTP root directory is /srv/tftp. When config files are stored in /srv/tftp, a new Aastra can find its config files. When config files are stored in /srv/tftp/aastra,

Is there a command in Asterisk that will cause all Aastra phones to reboot and/or recheck for new firmware? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070312/d394c874/attachment.htm

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