similar to: Fwd: Delivery Delayed: Re: Ring back tone with asterisk

Have you thought about making them agents, they would both be reachable by dialing there agent number then, and I know only one agent can be logged in at once. Just a thought. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Saturday, April 22, 2006 8:26 PM To:

If you are really wanting to build something on Raspberry Pi or similar ARM platform, you could also take a look at Elastix for ARM. http://www.elastix.com/en/downloads/ Elastix is a fully integrated platform, and includes the majority of necessary components in one installation. The new Raspberry Pi 2 platform may be perfect for your needs in this respect, although based on your load, the B+

Hi C F no asterisk and sip device are not behind same router. actually both are in different countries. how ever when caller and callee are behind same routers voice is just fine (both ways) and i can see re-INVITEs too. but when someone calls from another router then this issue arises. caller can hear the called party but called party can not hear caller. and there are no re-invites issued

On Wed, Jul 29, 2015 at 10:16 AM, John Novack <jnovack at stromberg-carlson.org > wrote: > > > Murthy Gandikota wrote: > > > > ------------------------------ > To: asterisk-users at lists.digium.com > From: webaccounts173 at jgoettgens.de > Date: Wed, 29 Jul 2015 16:11:31 +0200 > Subject: Re: [asterisk-users] Windows Asterisk Help > > > >

The mailing list will not receive emails from the forums. What I was referring to is that Discourse does provide the ability to receive emails for posts or threads you're interested in, and you are able to respond over email to them as well. On Mon, Dec 4, 2023 at 8:38 AM John Novack <jhnovack at stromberg-carlson.org> wrote: > > > Frank Vanoni wrote: > > On Mon,

Received this on the console -- IAX2/76.21.238.129:4569-4986 requested special control 20, passing it to SIP/magicjack-08225a58 Did a Google search, but reached a dead end Can anyone explain? Something need to be changed in my configuration? The call completed satisfactorily. Inbound IAX trunk - outbound to SIP provider magicjack ( no dongle ) Asterisk 1.4.26 TIA John Novack -- Dog is my

Same about me - need to re-enable membership all the time. Annoying (( ??, 12 ???. 2017 ?. ? 15:59, John Novack <jnovack at comcast.net>: > Not just gmail > Happening as well with Comcast.net > > My Comcast address is set to forward to another domain, as Comcast seems > to now block sending mail with a non Comcast "from" address. they turned > that on a couple

OK, so this list claims to have 1000's of users on it. Let's see where they are.... I was putting myself on frappr map for something else and found an unused asterisk map. I saw Olle on that (imagine that) but no mention of it on the lists. So..... Frappr yourselves. Let's see how many Astriholics there are out there. http://www.frappr.com/asteriskusers -Brian

Hi, I didn't get any solution in the mailing list. [http://asterisk.linkx.net/asteriskusers/200409/msg01167] What should be the next step? Changing the machine??????? Is it machine dependent?... Regards, Somesh S. Shanbhag --- Tzafrir Cohen <tzafrir@cohens.org.il> wrote: > On Tue, Sep 27, 2005 at 12:13:21AM -0700, somesh s > wrote: > > Hi, > > > > I did

Hmmm, looks indeed like changing the mtu helps, strange thing is that with the bigger mtu sometimes I had no problems and often TB are copied without problems. Problem is I can't test the connection that much because the connection has been established by a third party and it is just some vlan without any hops. But thanks for the link to the nfs-test that looks interesting!

Hi Chandrakala, Logan, Can you confirm that the attached patch fixes the overflow problem? Koen, can you confirm the fix makes sense? Cheers, Jean-Marc On 11/27/2017 12:10 PM, Logan Stromberg wrote: > Sorry, long holiday weekend in America. > I can say with pretty high certainty that there is an overflow occurring > and it is flipping smooth_coef_Q16 to be negative when it probably

Sorry, long holiday weekend in America. I can say with pretty high certainty that there is an overflow occurring and it is flipping smooth_coef_Q16 to be negative when it probably shouldn't be. I had originally thought it was only an issue where it was overflowing the 15th bit but not the 16th, which might still preserve the intended value for operations that ignore the sign bit (in cases

James Cass wrote: > I picked up a cheap JS200-FX on ebay: http://x100p.com/products/js200fx.php for $30, and it works great for a home install. Very low power draw as well. > > James Cass <http://goog_987864563> > jcass78 at gmail.com <mailto:jcass78 at gmail.com> The JS-200 runs a very old ( 1.4 ) version of Asterisk I have set up more than 30 nodes using the HP thin

Hi Does Sangoma 8-port card A108 support PCIe version 2.0 ? The card is here http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a108.html And we want to use 3 such cards in this motherboard because it has 3 PCIe slots of version 2.0 http://www.intel.com/products/desktop/motherboards/DX58SO/DX58SO-overview.htm Is this a good idea ? Do you have any experience

If reducing the MTU is helping, you might look into turning on Path MTU Discovery. NFS can be fast for large transfers if you tune it. http://stromberg.dnsalias.org/~strombrg/nfs-test.html NFS is not terribly secure though - at least v2 and v3 weren't. Not sure about v4. On Wed, Mar 28, 2018 at 12:59 AM, Marc Roos via rsync <rsync at lists.samba.org> wrote: > > Kevin, Dan,

I'm trying to switch to using rsync for updating a huge software library containing binaries, text files, symlinks, and so on. We've been using something homegrown which I'm not that happy with - it's a perl script that systems cp and chmod and such. The problem I'm seeing is: target computer: directories ctime-5 and ctime-5b3 are distinct directories source computer:

Yes, yes, I can reproduce it now, but only on platforms that define a 16-bit int by default (SA_Q15 is an opus_int rather than opus_int32). What system are you compiling this for? On Tue, Nov 21, 2017 at 8:34 PM, Chandrakala Madhira < chandrakala.madhira at soctronics.com> wrote: > Hi Logan, > > Please find attached the input stream we are using testing. > > Thank you, >

Hi, Can anyone let me know if this is a bug? Thank you, Chandrakala ----- Original Message ----- From: "Logan Stromberg" <loganstromberg at gmail.com> To: "Chandrakala Madhira" <chandrakala.madhira at soctronics.com> Cc: opus at xiph.org Sent: Wednesday, November 22, 2017 12:12:39 PM Subject: Re: [opus] Reg an issue with smoothing factor in VAD

Hi, our Asterisk is connected to an E1 port. So we are using the DAHDI-Driver. Please , how do I tell the driver/Asterisk to wait for overlap digits for in-calls? I found the option "overlapdial=yes" but I did not try yet. Is that "my" option? Is there any option for setting an timeout? Thorsten

I configured dovecot on a Fedora Core 2 system - the version that comes with FC2. I was using it with a RHEL 3 NFS server. Dovecot would again and again get stuck trying to grab an fcntl lock, even though I specified that it should only use dotlocks in dovecot.conf. Sometimes it would segfault (leaving things locked?). Often there would be two dovecot imapd processes for the same user.