search for: firware

...ther with Asterisk. All users are on SER and Asterisk is the gateway/voicemail. In these days I'm starting some tests using Asterisk accounts users. With this PA168S/AT320P, if I use it with a user from SER, it's ok but I can forget to use it with Asterisk users!!! I've also updated the firware at the 1.46 released the october 10th, but nothing changed. These are my user settings: ---- [221] type=friend username=221 secret=secret host=dynamic canreinvite=yes dtmfmode=rfc2833 nat=yes context=local mailbox=221@local callerid="221" <221> accountcode=221 qualify=yes ---- Any i...

...hing to do with the asterisk, because upgraded to version 1.0.7 a week ago. Though, the strange thing is that we also have some elmeg/snom190's and they do not have this transfer problem. Not being able to transfer calls is a major problem. I'm puzzled. I didn't thought upgrading a firware would distroy existing functionality. The new firware version in the BT's is 1.0.6.3, the old firmware was still one of the 1.5.x series. We are unable to downgrade the firmware. Something to do with missing files in the older versions compared to the new one. Is there anybody that has sim...

I'm reading the O'Reilly "Asterisk the definitive guide", 4th ed, with a starfish on it. In some ways, astonishing that it's not really that definitive, it's more general -- and it only clocks in at one ream of paper! In any event, I'm having some port problems on my home network: http://security.stackexchange.com/questions/81752/ I need to open ports for

I have found that the NTP server is not contacted when the phone (Budgetone 100) comes back from a power down. I must reboot the phone without powering down to get the phone to contact the NTP server for the time. It doesn't matter whether I reboot from the phone's web site or using the menu reset function: either works. I have only tested this with a private NTP server. This may

...polycom soundpoint ip 501 phone I was having problems getting some of the functionality working like the **number to pick up calls to other phones then it just started working (when it couldn't update itself for a while) Unfortunately then I got that issue sorted and it updated to the latest firware and now it doesn't go again. Is there something in the config files themselves I need to edit to get this functionality back? I am running asterisk with a trixbox installation. Thanks Kate

...ed a document contains the its protocol and a little box with rhino firmware to testing driver development. I say 5 stars to rhino support. About solis, I received a solis hardware and a Microsol Windows application sources in Delphi. This application contained the protocol, but a juggle on solis firware was obscure, its internal rotate ?calendar, so I did need to discovery by hand, to allow adjust its internal date, as you see in solis driver code. I say 4 stars to solis support. cheers, Silvino 2009/10/9 Arnaud Quette <aquette.dev at gmail.com>: > Hi Silvino, > > long time no s...

...the router ports on the LAN side are all unmanaged, just a huge Ethernet "wirenut" You SHOULD be able to communicate between devices on the LAN without any firewall issue. I have also found with some routers that the DMZ isn't what one expects, and can get in the way, depending on the firware. Does this router have any SIP ALG setting? turn it off! As an aside, I would caution you to not have SIP 5060 exposed to the public Internet, or you will soon regret it. I am sure others will have much better information though John Novack thufir wrote: > I'm reading the O'Reilly &qu...

can anybody identify why the CLI is issuing a failure message while debug shows everything is fine???? this makes no sense to me. also, why is the username being updated? this has got to be wrong from CLI -- SIP Seeding '52221' at 52221@192.168.70.26:5060 for 3600 -- SIP Seeding '52221' at 52221@192.168.70.26:5060 for 3600 Dec 24 12:16:35 NOTICE[15776]:

...Asterisk is the gateway/voicemail. > > In these days I'm starting some tests using Asterisk accounts users. > > With this PA168S/AT320P, if I use it with a user from SER, it's ok but > > > I can forget to use it with Asterisk users!!! > > I've also updated the firware at the 1.46 released the october 10th, > > but nothing changed. > > These are my user settings: > > ---- > > [221] > > type=friend > > username=221 > > secret=secret > > host=dynamic > > canreinvite=yes > > dtmfmode=rfc2833 > > nat=y...

Hi all, I just picked myself up a Mediatrix FXO SIP gateway to play around with and hook into Asterisk but have no documentation. Are there default passwords or IP's that I need to know if I do a factory reset? Or better still, would anyone have a User Manual they could send my way? Any help would be appreciated. TIA. Jamie

OK What client cards do work? For each of the following: 802.11a 802.11b Media: PCI PCMCIA USB OS: FreeBSD 4.7 Recomendations please Thanks David ----- Original Message ----- From: <john@critchley.biz> To: "vizion communication" <vizion@ixpres.com> Sent: Monday, April 14, 2003 2:04 PM Subject: Re: Annother Wireless network card annother problem > > > > I

Hi, there were some postings a few weeks ago telling that the GSM codec problem with snom100 will be fixed. But it still seems to be very quality. Will be any change in this subject? THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/2fa9d206/attachment.htm

Hello folks, I'm trying to get an 8x8 DTA-310 running mgcp to work. I get no dialtone & can't get it to ring. My mgcp.conf says: ; ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 0.0.0.0 [172.16.2.25] host = 172.16.2.25 context = default line => aaln/1 And here's the interesting bits of extensions.conf: [globals] ... TRUNK=H323/BYEXTENSION@pstn_gw ...