Displaying 20 results from an estimated 1100 matches similar to: "Where should I look for MWI settings if Aastra phones don't do it?"
2008 Aug 24
2
MWI working perfectly. Shouldn't it be broken??
I have a Sipura 962 endpoint on Asterisk 1.4 where the MWI works
perfectly, however my theory is that it should be broken.
Obviously I'm wrong but "Sip show subscriptions" does not show the
endpoint subscribing to the MWI status on Asterisk, even though all of
the other endpoints on the system DO subscribe for their respective
mailboxes, including SNOM & Polycom endpoints.
2011 Nov 23
1
MWI for non-subscribed Realtime peers?
Hi,
I have an Asterisk behind an OpenSIPS proxy. The proxy handles registrations and also SIP SUBSCRIBE for MWI. The Asterisk are configured to send NOTIFY to the proxy even when the SUBSCRIBE haven't been received. I can configure a user in sip.conf that works:
[az5134939706]
type=friend
host=xxx.xxx.xxx.xxx (IP of proxy)
port=5060
nat=no
mailbox=1234 at customer
subscribemwi=no
2008 Feb 07
2
Snom 300 MWI
I think I have my echo problem solved, now i need to tackle the MWI. I
can't seem to get it to light up. I'm using Asterisk 1.4.14. Here's a
section from my sip.conf for my test phone:
[general]
context=internal
allowguest=no
allowoverlap=no
allowtransfer=yes
notifyhold=yes
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
pedantic=yes
vmexten=9998 at internal
;vmexten=*97
2013 May 06
1
OT - Differences between Aastra 6730i and 6750i series
Hi,
What are the main differences between Aastra SIP phones 6730i and 6750i
series ?
Aastra corporate web site mentions :
"The Aastra 6730i Series offers exceptional features and flexibility in an
open-standard enterprise grade IP telephone" for one
"The Aastra 6750i Series offers features and flexibility in an
open-standards based, carrier grade IP telephone." for the
2010 Feb 03
1
aastra 9480i dtmf ?
Hi,
I just deployed new Aastra 9480i phones and when I attempt enter digits on
other systems, like host pin in a GoToMeeting, the servers on the other end
do not get my entries. I am assuming this is a DTMF issue but do not see
anything in this phones config other than turning on the display of the
digits. I have the DTMF method set to "SIP INFO". I am using AsteriskNow
w/FreePBX.
2011 Apr 06
1
MWI not working on most ATAs in Asterisk 1.6.2.17
We've had several customers report since upgrading them to our new
Asterisk 1.6.2.17 server (from version 1.4), that their MWI no longer
works. No significant changes have been made to their SIP
configuration, nor to their ATA configuration.
While not exhaustive, these are the ATAs that don't work:
Linksys SPA2102
Linksys PAP2T-3.1.15
Thomson 780
Thomson 784
Unfortunately, this
2009 Dec 05
2
How to use SIP hints and BLF for realtime extensions on Aastra phones?
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
2009 Aug 05
3
Several mailboxes on SIP peer
I have in my sip.conf the following
[jon.moore]
type=friend
mailbox=8100,8150
In voicemail.conf, both mailboxes are defined.
On my Aastra 480i phone, I only see the first mailbox
listed. I've verified this, by changing mailbox= to
reverse the order, and I then see 8150 when I go to
Services > Voicemail on the phone. I also only get
MWI events for whichever mailbox is listed
2007 Jul 03
1
lookup a anonymous internal caller
Dear list,
following problem, i have some users, who are supressing their callerid.
This setting is adjusted at the sip phone. So if these guys are calling
internal persons nobody sees the callerid. I am looking for the
following resolution:
User has set his phone to anonymous, user calls somebody internal,
Asterisk initials a lookup on the channel and generates a new callerid
for the
2005 Aug 22
0
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 480i
2005 Aug 22
0
Aastra 9133i 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
2006 Apr 18
0
Aastra 9133i Phones Asterisk 1.2.6 and MWI
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?
2007 May 21
3
Aastra MWI
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
2008 Oct 17
5
How to add contexts in asterisk realtime?
Hi everybody,
How can we add new contexts in asterisk realtime module? All I could figure
out after googling is that a new context HAS to be declared in
extensions.conf with 'switch => Realtime/@<databasetable>' under the context
name declaration. This works fine as long as we are adding extensions only
to this one context, but doesn't give the freedom to add new contexts for
2013 May 06
0
OT - Question on Aastra 6735i - Was: Sip phone displaying caller name while on call
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
2015 Sep 24
2
same sip username with realms and chan_sip
Hi,
How have the same sip username in several realms ?
For now, I must add the realm prefix in the sip username of chan_sip.
For example:
[lg_2540]
amaflags = default
call-limit = 10
host = dynamic
language = en_US
context = lg_default
callerid = "LG" <2540>
secret = XXXXXXXXXXXXXXXXXXXXXXXXXX
type = friend
subscribemwi = no
mohsuggest = default
qualify = yes
2006 Nov 13
0
MWI not working in 1.4
Before I open a bug I'll ask again if anyone else is having trouble with
receiving MWI on SIP devices in 1.4. My configuration was working fine
in 1.2 but as soon as I change to any build of 1.4 I don't get
notification on any of several SIP devices. I can post my configuration
but since it was working I can only assume it would break if something
in voicemail.conf has changed or
2010 Aug 11
0
Aastra 6739i Support
All,
I have multiple Asterisk servers in various locations running various
1.4 and 1.6 versions (lab and production) and am having trouble with a
new Aastra 6739i (3.0.1.2015) registering. Below is my request to
support and they have looked it over and don't see anything wrong:
Support, Can not get a 6739i to register with 3 different Asterisk
servers with varying configurations/versions
2014 Sep 08
1
Asterisk removes ice lines in sdp when calling between webrtc clients
Hello,
I have a problem with a call between 2 webrtc clients. Asterisk removes the
ice-related lines from the sdp when it sends the INVITE out, and the called
webrtc client rejects the INVITE due to the missing ice lines. Both webrtc
clients are defined exactly the same way, same values in all fields except
the number of the peer.
There's probably something I've changed that causes this
2006 Jan 05
1
Incoming PSTN Calls
Hi all,
I am having difficulty getting incoming PSTN calls working. I have set
up an account with a third party provider. In my system, the user
register with SER and use Asterisk for PSTN access, voicemail etc
My provider told me to change my sip.conf as follows
register => username:password@sip.blueface.ie/2093
; To receive incoming calls specify this block and