similar to: Registrar only setup

On Fri, 13 Jan 2006 21:14 , Tomas Florian <tflorian@telus.net> sent: >Hello, > >I''m having trouble with slow IO performance under Xen 2.0.7 with 2.6 >kernel. I''m running 3 physical machine. I did hdparm -tT in my Dom0 on >both servers and this is what I get: <snip> I would check to make sure your kernels have the right ide drivers builtin or as

Hello, I have recently migrated my Win NT4 server to Linux with Samba 2.2.5 :-) One of the applications we need to run is Simply Accounting 9.0 / 2003 in multi-user mode. The company that makes Simply Accounting sais the data must be shared from Win NT,2000 or XP but sharing it through Win 95,98 or Samba could corrupt it. I know that Simply Accounting stores the data in .SDB database file

Dear all, I have an asterisk sip issue which I don't believe is unique. I use a registrar (sipgate.co.uk) where I have 3 different accounts. These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users. By using just one of these accounts I can set asterisk up to send and receive calls no problem. However, when I start to introduce an

Hi All I have posted this problem many times on the list but no reply, trying one more time may be someone will response this time When I call from 1 RTC Client to another without Asterisk everything use to be fine but when asterisk is there as a Registrar a problem use to occur in more than 90% calls, Caller can hear the voice of the receiving side but the receiver cant be able to hear the

Hello, So far everything that I'm trying with asterisk is working except for this weird thing. When I try to call voicemail and it asks me for the password I enter it in but from the debug message I can see that it thinks I didn't enter anything in. Also when I'm leaving a message it sais press pound to end, but even if I press it 10 times it keeps on recording until I hang up. It

Hi; As you probably know, SER style of handling an incoming call is : 1) try to look-up it from registrar DB 2) if not found there, try to do some thing else Is there any possibility of doing the above at "Asterisk Dial-plan"? Regards Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi, I have a really strange problem which is driving me crazy for days now. If I register my asterisk (tried all versions from 1.6 up to 13.x) with one sip registrar, everything works... calls go out and call come in... no 32 seconds limit. but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. as far as I know, there is no firewall in

Greetings, What are some registrars that members of this list have had good experience with? I was stepping through the godaddy checkout process, and being opted-in to a dozen different upsell features just left a bad impression. But I have no clue who else to go with. -Eugene -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I know this is slightly off topic but I figured the knowlege here is probably the best on the subject.. I want to setup remote offices with 4 to 6 SIP phones (SNOM 200) using ADSL and the internet to connect to the Asterisk box.. These phone will be behind an ADSL router using NAT... I don't want to setup another Asterisk system in each office so IAX is not an option.. I could use

Hi all. Is it at all possible to have a GS B101 NATed with firmware 1.0.4.40? I've tried both STUN and not STUN. The odds seems best with stun because the phone registers with right ip adress. When the connection is made * sends rtp packets to the right destination AND port, but the phone doesn't accept the packets..... Should I burn my D-LINK 604 or upgrade the GS? /t

Hi all, I am new to this list... Can I use Asterisk as a SIP Proxy and as a SIP to H.323 translator? I want to implement PC-to-Phone calls in the following topology (for signalling): SIP Softphone --> Asterisk --> Gatekeeper H.323 ---> Gateway H.323 ---> PSTN The RTP audio packets would go direct through Softphone to gateway. Does someone have a configuration file example of

Sipdiscount has replaced their asterisk servers for another thing. Then, no more iax. Ok, but I can't make calls using sip also... I'm getting a "forbidden" error when using sip1.sipdiscount.com. Anybody got it working? -- Alejandro Vargas

Hello all, I have installed the .deb packages of the Asterisk v1.8.3.3 from the upstream project on my Debian GNU/Linux Squeeze server and bought the Comodo's PossitiveSSL SSL certificate to be used for my SIP/TLS exercise. After setting up everything and trying to fix this problem, I am still getting a 401 Unauthorized SIP message. So as of this writing, I still cannot successfully REGISTER

Hi! Then I phone other phones with my Zyxel 2000W (WI-FI) it just hang up when I answer the phone I am ringing. It works fine if I call the 2000W from other phones. I have tried many sip settings. I use this now: [205] type=friend username=205 secret=passwd205 callerid="Zyxel" <205> host=dynamic context=local nat=yes canreinvite=no disallow=all allow=g729

Until now, we were tied to Amen (which delegates to Network Solutions), which was not an optimal choice. As of today, I've moved to Gandi, another French registrar, which is a kind of good citizen out there: http://www.gandi.net/whowe/ http://www.gandi.net/supports/ While I've done everything to avoid a disruption during the transfer period, we might still face some issues. If you notice

I'm using the latest SIP firmware from Avaya. The phone receives the 46xxsettings.txt OK, and then after entering extension and password it goes to the home screen saying 'Registering'. When I check options->ViewIPSettings->IPAddresses on the phone, the registrar and SIP Proxy fields are blank. I have both lines: SET SIPREGISTRAR "67.1XX.XX.XX" and SET

Hi all, I've been pulling my hair out for two days over this problem... I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem

Hi everybody,

Hi, We switched from compiling LLVM with gcc to clang (3.3) and it appears that clang (correctly I think) optimizes away the GDBRegistrar's __jit_debug_register_code() function that's used to trigger reading debug info from JIT-ted code, breaking GDB support. This patch forces it to leave the call using the method described here in the 'noinline' section:

On Fri, Dec 20, 2013 at 11:18:46AM +0100, Andrew MacPherson wrote: > This patch forces it to leave the call using the method described here in > the 'noinline' section: Use asm volatile("":::"memory") to make sure that it doesn't leave trackes. The noinline can likely go in that case... Joerg