search for: axiali

Symptom: Phone after about 15mins will stop functioning Problem: DHCP lease renewed but default route dropped Fix: Assign a static ip and problem is resolved. Upgrade to new firmware once it is released It turn's out that these phones have a few issue in 1.0.3.81 firmware. The phone may stop transmitting packets if configured with DHCP, if DHCP is being provided by certain devices. Netopia

Hi Guys, is there a client which can be used on the SonyEricsson P800/P900...? IAX would be cool, but i take anything that can connect (via bluetooth) to an asterisk-server ;-). The phone is Symbian, and can also execute java-stuff... Greez Andreas _________________________________________________________________ Find your perfect match @ http://personals.xtramsn.co.nz with XtraMSN

Hi, I know it's not really the place, but if anybody in the UK (or US) is interested, I'm clearing out lots of new Cisco stock... 7970G's (colour LCD), 7960G's, 7940G's, 7920G's (wireless IP phone), 7935's (conference phone) and 3550-24-PWR switches. I also have boxes of 7914's, the single-7914 foot stand and double-7914 foot stand (these are required to

Dear all, I'd like to know if the "core" (demo, voicemail...) asterisk prompts have ever been recorded in french (and are freely available). If not, I'm willing to have them studio-recorded by a professional speaker, and contributed back to the community. Does a message list exist ? -- Nicolas Bougues Axialys Interactive

hi all I spoke to this guy the other day, working with Cisco's VoIP system. He told me they were using a PRI/E1 to transport SMS, and could even do so from their phones. May this be possible with asterisk? I have an E100P in my primary asterisk server connected to a E1/PRI. roy

Dear all, I read across Asterisk's lists archives, and found out various discussions about how nice it would be to have a (SQL) database abstraction layer enabling the use of various SQL backends, for various purposes inside of Asterisk. As far as I see, there is no such thing yet, although there are various efforts (on CDR and voicemail, mostly) to use external databases. As I'm

I have some strange question bout the asterisk (gpl license ...) but i'm not an experienced linux user ... What happens if for example a big company buys digium , do we have a garantuee that asterisk stays opensource ??? Kind regards Michael Devenijn -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 3208

The quick question: Do the digium drivers for the Digium Wildcard TE410P (4 port T1/E1/PRI 3.3v card) , the T100P (single port T1), and the TDM400P support hot plug PCI? I am also noting that while the TDM400P doesn't state the voltage requirements, it looks like a 5v card. I hope that I am wrong on this. What type of T1 channel bank would people recoment for ISDN (for video

Hi all, Information on this topic seems a little scarce, so I thought I'd try the list.... Apart from the the coolness factor can anyone explain to me in what situation one would use TDMoE rather than IAX for communication betwwen 2 Asterisk servers?

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Well, a bit of a newbie using Asterisk and trying to get some Sipura SPA-841 phones to talk to the Asterisk server. Not doing something right apparently.... If I turn on debug for the extension in the CLI (sip debug ip xxx.xxx.xxx.xxx) I see frequent 5 packet attempts by the server to contact the phone, but seems to always be failing. The status

Asterisk Users, Developers, Gurus, Etc., AstriCon Europe 2006 is only 20 days away! Sign up now to ensure a seat at the European Asterisk event of the year! Events will be held in Berlin (June 19 - 20), Paris (June 22 - 23) and London (June 26 - 27). Don't forget to hold the date for AstriCon 2006 USA in Dallas, October 24 - 27. REGISTER: http://www.astricon.net/ TUTORIALS: - Asterisk

Hi Has anyone opened up a grandstream phone or handytone ATA to find out what is inside? What is the CPU? How much RAM? Cheers Rob

When I was trying to run mutiple E1s over TDMoE, the zaptel would drivers complain about too little memory, whenever I added more than 31 channels. Requesting 62 channels in a dynamic span gave me ... span creation failed: Cannot allocate memory upon loading the zaptel drivers. How would you go about running, 8 or 16 say, E1s over TDMoE? Would you create multiple dynamic spans or just one large

from the zaptel sample config: # "dacs" : The zaptel driver cross connects the channels starting at # the channel number listed at the end, after a colon # "dacsrbs" : The zaptel driver cross connects the channels starting at # the channel number listed at the end, after a colon and # also performs the DACSing of RBS bits dacs=1-24:48

In the setup I'm looking now there are IPv4-only and IPv6-only SIP-clients (or UA's) but the Proxies (or servers) are v4 and v6 dual-stacked (to make things a bit easier). I think there will be a lot of IPv6-only SIP-clients (e.g. mobile clients and all the existing IPv4-NAT'ed networks) and a lot of IPv4-only "old legacy" SIP-clients. We have to find ways to deal with such

Hello, Anyone using the 1.0.4.55 firmware release with any success? I have had my Budgetone running 1.0.4.50 for about a month and a half now with no problems whatsoever, and I am a little leary about upgrading. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place!

Dear all, I'd like to know if there is a way for multiple asterisk servers to share a common SIP and/or IAX registry. The setup I imagine would be something like : - several asterisk servers called sip1.isp.com, sip2.isp.com, ... - a DNS alias sip.isp.com pointing to all the addresses (thus providing a round robin resolution on each server) - each SIP client would register with sip.isp.com

Hi, I read in the Asterisk Whitepaper, that you can run two cloned servers, one as a primary, one as a backup, and have them automatically failover to the other unit when it crashes, or when you need to restart it. The primary application of course, would be ensuring calls can be made when frequent updates are being handled, or when an update must be restarted on a busy network. The term

Firstly, the problem... Ever since I installed and setup asterisk, I have had various problems, initially it was echo caused by the ISDN (isdn4linux) card I was using. So, I upgraded to the X101P from digium. I still had echo, so I figured it was also caused by the ATA186 (cisco) I was using. So, I upgraded again to the TDM40B quad FXS card. This solved pretty much all my problems, except

The freenum.org beta continues to roll forward. If you have an Asterisk or SER SIP gateway/proxy, please see if you can make some sense of the examples below and install them in your system. Your users will hopefully be able to dial toll free numbers in various nations just like they dial regular numbers in those same country codes. I'd like to get some additional people trying to make