we installed TL on 1.4x right from the start we have had problems with the phones not being monitored,, we have asked support about this several time to not resolve..
as we started moving our customers from our old software to TL, we notice that the message waiting light and stutter tone are also not working.
it would seem to me that one might be causing the other, we have checked everything we can and have not finding anything
we have emailed support several time about this problem, but have yet to get the simplest of questions answered so I thought we'd look here for others runing the same softeware that might be having or had the same problem..
we are running Asterisk version 1.4.13 and TL version 5.0.43, we are running the lastest and as a result have other feauters broken (see other post) but have also requested serval time for a whats new / fix list and haven't recieved a reply to that either..
so the questions are
1 whats version of Asterisk and TL are you running..?
2 are your phones being monitored..?
3 is your call waiting and stutter tone working on the phones..?
note we have tested this on the SPA-941 / 942 - Cisco 7960 - Polycom 300 all with the same result..
if you have this problem AND figured out how to correct the problem please share as we are one step from moving our customres have to our old software we are getting ALOT of complaints about this..
Thanks in Advance
George
>> Perhaps you did not know
>> Perhaps you did not know but I did respond to the same question about 1.4. and MWI to your
>> colleague Chris a couple of days ago, so please be a little easier on TL support :).
no comment in the open forum
moving forward..
Here is what I found running debug on both VoIP servers
TL message waiting light and stutter tone NOT working
Scheduling destruction of SIP dialog '59068f3b33d9789c61d376951df5ba67@66.28.179.94' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 207.171.194.121:1205:
NOTIFY sip:201-gmac@192.168.1.101:5061 SIP/2.0
Via: SIP/2.0/UDP 66.28.179.94:5060;branch=z9hG4bK656439ad;rport
From: ""PRIVATE" +13035922947"
sip:"Private" +13035922947@66.28.179.94;tag=as335c0bbe
To: sip:201-gmac@192.168.1.101:5061
Contact: sip:"Private" +13035922947@66.28.179.94
Call-ID: 59068f3b33d9789c61d376951df5ba67@66.28.179.94
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88
Messages-Waiting: yes
Message-Account: sip:201@66.28.179.94
Voice-Message: 1/0 (0/0)
SN message waiting and stutter tone WORKING
Scheduling destruction of call 'cac7794f-ac2ff39f@192.168.1.107' in 15000 ms
12 headers, 3 lines
Reliably Transmitting (NAT) to 207.171.194.121:5061:
NOTIFY sip:home-111@192.168.1.107:5061 SIP/2.0
Via: SIP/2.0/UDP 66.28.179.82:5060;branch=z9hG4bK11cd6647;rport
From: "Private" sip:Private@66.28.179.82;tag=as4d6b2854
To: sip:home-111@192.168.1.107:5061
Contact: sip:Private@66.28.179.82
Call-ID: 38d832fd226e979c716c6d127c3edde3@66.28.179.82
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 93
Messages-Waiting: yes
Message-Account: sip:asterisk@66.28.179.82
Voice-Message: 1/0 (0/0)
can anyone see anything wrong with the above..?
NOTE the message board is stripping out everything in between the greater and less then arrows.. so I removed them so you could see the complete statements
George
George, Did you ever get
George,
Did you ever get your MWI working?
I am using Polycom's and it seems to be broken using the auto provisioning.
I will try a manual config. I see the SIP packet going out to the phone.
-Matt
If I do a manual config it
If I do a manual config it works fine.
I will try to see what the difference is.
-Matt
George,
Perhaps you did not know but I did respond to the same question about 1.4. and MWI to your colleague Chris a couple of days ago, so please be a little easier on TL support :).
MWI does work in 1.4 - at least with Asterisk 1.4.13 I am currently running using Snom phones I have on my desk, but the notification mechanism is the same for all the phones.
That said, you probably need to find out any specific settings for the devices you use – perhaps someone could post that. I believe that Polycom needs a setting for stutter in its configuration, but the message notification should not reqire any special settings.
I am not sure what you mean by the phones not being monitored. As far as the MWI, you may want to watch SUBSCRIBE and NOTIFY SIP messages to see what may be going wrong. You can use Asterisk command “sip set debug” for that, or just trace the network traffic.
I just want to make a simple point - Thirdlane PBX is a sophisticated management tool, but at the end it simply generates Asterisk configuration files. You should be able to see if they contain the data you expect. In fact, as replicating every config file parameter in the GUI would be simply too much, we provide "Other options" field where you can enter additional key/value pairs for the options that are not explicitly defined in the GUI.
Thirdlane software helps with Asterisk management, but PBX run time is really Asterisk and the devices on the network, and many things can go wrong. We understand that this is not always obvious and try to help as much as we can - but we do not offer unlimited free support.
I hope this may help you - here is a fragment of my sip.conf
[general]
bindport=5060
port=5060
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=gsm
callerid=anonymous
context=from-outside ; Default context for incoming calls
allowsubscribe=yes
notifyhold=yes
notifyringing=yes
limitonpeer=yes
[200-thirdlane]
qualify=yes
nat=no
callerid=snom 300 cid <200>
context=from-inside-thirdlane
call-limit=10
canreinvite=no
vmexten=200
secret=05I3LB6u
username=200-thirdlane
host=dynamic
subscribecontext=local-extensions-thirdlane
dtmfmode=rfc2833
mailbox=200@default-thirdlane
type=friend
disallow=all
allow=ulaw
allow=alaw
allow=gsm
[201-thirdlane]
qualify=no
nat=no
callerid=snom 320 <201>
context=from-inside-thirdlane
call-limit=10
canreinvite=no
vmexten=201
secret=bgNj6qCG
username=201-thirdlane
host=dynamic
subscribecontext=local-extensions-thirdlane
dtmfmode=rfc2833
mailbox=201@default-thirdlane
type=friend
disallow=all
allow=ulaw
allow=alaw
allow=gsm
Fields on the bottom of general section are for making BLF work, the way to configure that on the phone is phone specific.
Also, I moved your post to the "News and announcements" into the "General questions" forum, please don't post bug reports in the "News and announcements".
We will be adding a ticket tracking system for this purpose; in the meantime you can use the "General questions" forum.