similar to: MWI for non-subscribed Realtime peers?

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

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.

Hello list members, We're trying to get MWI notifications on our ATA device and we set it to send SUBSCRIBE messages to Asterisk, but it gets UNAUTHORIZED messages, despite the fact that we set the following lines in its settings in sip.conf: subscribemwi=yes mailbox=21 at from-extensions We need help in understanding how this works and what we are doing wrong. This is the SIP debug

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

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

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

To all moderators i guess, my question was probably not clear this is not a homework, i am trying to understand R by doing some exercise in my book. I will however participate a course in R in august and thought it could be good to have some knowledge before. I hoped for help from you since i have no instructor to ask, that would have been my first choice. thanks anyway Lotta

Hi, I have a small Linux server PC at home (Intel Core2 Q9300, 4 GB RAM), and I''m seriously considering switching to OpenSolaris (Indiana, 2008.11) in the near future, mainly because of ZFS. The idea is to run the existing CentOS 4.7 system inside a VM and let it NFS mount home directories and other filesystems from OpenSolaris. I might migrate more services from Linux over time, but for

1. Emma is performing an experiment that requires individual handling of some animals. The sizes of the animals are lognormally distributed: The natural logarithms of their sizes has a normal distribution with mean 3 and standard deviation 0.4. The time (in minutes) it takes to handle each animal is given by 10 + s · 1.5 + eε for animals with s ≤ 20 20 + s · 0.8 + eε for animals with s > 20

Friends, The developer team for Asterisk not only consists of coders - a very important part are the testers, those that test new code and give feedback. For a few weeks, I've been maintaining a large number of branches with various stuff in them and have gotten very little feedback, not enough to judge whether or not to move forward with these patches. Some, but not all, code is

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

I am interested in a model diagnostic for logistic regression which is normally distributed (much like the residuals in linear regression with are ~ N(0,variance unknown). My understanding is that most (all?) of the residuals returned by residuals.lrm {design} either don't have a well defined distribution or are distributed as Chi-Square. Have I overlooked a residual measure or would it be

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

ok this is a wired problem. when i use X-Lite - after i register with asterisk X-lite sends a subscribe/notify request to asterisk to determine if the account has any messages waiting. if i create a sip.conf account using: user 12345 with a voicemail box 12345 - MWI works user jwolosuk with a voicemail box 12345 MWI fails (gets a 404 not found upon a subscribe) does anyone have a clue why

Hello, I am facing a problem I can't understand. I have several realtime SIP peers and from time to time, the mailbox field is not loaded in asterisk memory. The mailbox field is correctly populated in the database, but often, after an asterisk restart, the mailbox is not associated to the peer (just to understand, if I run "sip show peer 104-TEST", I see the Mailbox empty. If I run

Hello all, I have an Asterisk installation with Kamailio using realtime integration. I have gotten my clients to register, but there is something odd about the sip message flow with some of my clients. My clients are Zoiper and Asterisk is 11.10.2. When I set 'Subscribe to MWI' value to 'both', after a normal, successful registration Asterisk begins to send REGISTER messages to

Hi all, I'm trying to configure an * box for my home in an embedded device, so I want a minimum configuration. I've already configured it to connect to my SIP provider and my IP phones and ATAs, so far so good. My SIP provider gives me voicemail service and I'm happy with it. I don't want to run voicemail services in my * box. Is it possible for asterisk to subscribe to my SIP

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

Hello everyone, My setup: Debian squeeze Asterisk 1.8, DAHDI, libpri, compiled from source TE110P, attached to a Deutsche Telekom Octopus E Modell 300/800 I'm trying to get MWI for Voicemail working. In the same server I have also got an Eicon DIVA PRI card for testing purposes (it is integrated via CAPI and the chan-capi channel driver into my Asterisk). MWI works just fine there. I

Hi list, how does one get a SNOM 200 MWI to work with * ?? When I press the MWI button it doesn't connect with voice mail on my * box. thanks