similar to: AW: WiFi SIP phones

I'm trying to build a SEM using the sem package. I'll attach my model specification below. When I turn debug=TRUE, it seems as if I'm getting to convergence because I get this message: Successive iterates within tolerance. Current iterate is probably solution. However, at the end of the process I get this message: Warning message: In sem.default(ram = ram, S = S, N = N,

Martin: I had seen your other post and sent you a message off-list, but I never got a response. What do you feel is the most lacking that does not make it ready for a production enviroment. - I've been using a SIP deskphone in my office and usually some sort of ATA at my house, both as the primary phone. I've also had mobile phones from almost every carrier. Each one of these devices

Has anyone done any integration work with Covad's hosted solution ? I am considering Covad's hosted solution and want to be able to use Asterisk to develop some other apps. Anyone else tried this ? how did Covad react. I know they use MGCP. Another thing, the Cisco reseller rep tells me if I have a bunch of 7960's setup for MGCP (for use with Covad) I will need to get these

Hi Was able to communicate clearly with e60 and E61 with asterisk with new access point , even though the access point security setting was of ?opennetworks? , the previous one was of ?WEP? , I feel this was a major hurdle in communication , now I can clearly accept and make calls using Nokia E60 and E61 devices Next I will be trying to find out how to make this device work

There is an option that you can add to your dhcp server option 150 IIRC. -----Original Message----- From: "Rich Adamson" <radamson@routers.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: 2/27/06 4:37 PM > Has anyone done any integration work with Covad's hosted solution ? I > am considering

I recently upgraded a Cisco 7960 to the SIP firmware, it worked fine without a "call-manager". I just put the SIP firmware and associated config files in the TFTP directory of my asterisk server so that the phone could pull the firmware off of my asterisk server via TFTP. It took me about 5 minutes maximum to get the phone working with SIP through asterisk. NO Call-manager was used in

Hello Olivier Ive been testing the E61 phone for some days now, and we need to have an inhouse asterisk server, connected to our main asterisk server, to get it to work. That means, that you cant just walk down to your local airport, and use the IP part of the phone on their network. You have to have a non nat local server, to get it to run. Other than that, the phone can accept calls both

That's just the thing, and it sucks, because the VoIP implementation actually works very good. Jon _____ Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af list mail Sendt: 8. juni 2006 02:34 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: SV: [Asterisk-Users] I can hear only one way when I use

I have noticed that Asterisk (version 1.2.13) is not hanging up a call when the wifi handset moves out of range. My setup is Nokia E61 connected to wifi access point (private IP range) and then to server on internet (public IP). I have been testing using the talking clock application, and walking out of range does not hang up the call. The call will continue for hours even though the handset

Hi, I have questions about the Polycom 601 and side card.... 1) In the side card the lights all time off... But all functions it's ok. I need help with extension module of polycom... All works fine... But lights not work.... So... I don't know when any person or extension is busy... Any ideas? , Olger On 3/27/06 11:34 PM, "asterisk-users-request@lists.digium.com"

Having scoured the web, I still am no better off .. I have 2 Aastra480i's , and 120+ cisco 7940's :) . I am trying to decide which model to use going forward when we purchase more kit. They both seem much on a par regarding features. Q1: Is there anyway of making the cisco auto-answer _without_ having to manually edit the configuration on each phone ? I've been able to get the

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

We're looking at a large wifi phone deployment, and we're looking for wifi phones that: 1. Are SIP compliant (Asterisk friendly) 2. Provision capable (ideally TFTP of own MAC address) 3. Industrial quality (no cheap plastic stuff). 4. Well documented (and none of the "only telco's get documentation" crap) Does anyone have a suggestion? Thanks, MD -------------- next

Hello Be aware that the Nokia E60, E61 and E70 does not support NAT. Just to be shure that you know that. A clever choice from Nokia, so that users has to have some local equipment from the telco. Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af John Joseph Sendt: 7. juni 2006 13:59 Til: Asterisk Users

I am trying to estimate a covariance matrix from the Hessian of a posterior mode. However, this Hessian is indefinite (possibly because of numerical/roundoff issues), and thus, the Cholesky decomposition does not exist. So, I want to use a modified Cholesky algorithm to estimate a Cholesky of a pseudovariance that is reasonably close to the original matrix. I know that there are R packages that

The problem with the extra ptime descriptions in the SDP has been fixed in Asterisk (see http://lists.digium.com/pipermail/svn-commits/2006-October/017694.html). I've got the latest version of the 1.4 branch from SVN and have verified that the codec negotiation is working again. If you don't want to try the latest SVN version, then you'll have to restrict the phones to a single codec

> > Allen Casteran wrote: > > Anyone know what the POE draw is for the Aastra 480i phones? > > We have switches that will do 15 watts on 12 ports but only do 7.7 watts on all 24 ports. > > A Cisco 3560 switch will do 15.6 watts on all 24 ports. > > Just trying to find out if we need that much power. > Drew wrote: > According to Aastra tech support, 5 watts

Hi ! I'm using PXE to boot a Linux on a DLink DFE-530TX card. Everything worked well until I installed a Windows98 on it (please, foregive me...). Since the Windows drivers for that NIC were installed, I 'm unable to boot PXE, and I get the error: PXE-E61: Media test failure, check cable Any clue on how to make PXE work as before ? Cheers, -- Ludovic DROLEZ

Hi all, Does anyone know of an IAX client for Symbian? I have an e61 and would like to make calls through my home Asterisk box from places where I have WiFi access, as NAT is in the way I suspect that it'll be a pain to get SIP working like that as the NAT router doesn't do SIP connection tracking. Thanks, Peter. This message has been comprehensively scanned for viruses, please visit

Hi, I've set up a Gizmo Project account for access on my Nokia E61 because they work through NAT. Trouble is If I include my gizmo account in an asterisk hunt group and I'm not connected (phone is off / outside wireless coverage) the gizmo project always answers. Either the call goes to voice mail or if I turn voicemail off the call gets answered by a recording saying I'm not