search for: declination

Hello I have created a graph using the following commands: <<< startBReP3O1T <- diffs$BReP3O1T - diffs$diff_BReP3O1T endBReP3O1T <- diffs$BReP3O1T x <- seq(47,89, length = 10) ymin <- min(min(startBReP3O1T), min(endBReP3O1T)) ymax <- max(max(startBReP3O1T), max(endBReP3O1T)) y <- seq(ymin, ymax, length = 10) plot(x,y, type = 'n', xlab = 'Age', ylab =

Hello, I am trying to use the libvirt Java bindings (version 0.4.7) with libvirt version 0.9.12 to connect to a XenServer hypervisor. Virsh is able to connect to my XenServer, but when I try to do the same thing in Java, it won't connect. Here is the debug output: 2012-06-26 19:48:52.259+0000: 26051: info : libvirt version: 0.9.12, package: 1.fc16 (Unknown, 2012-06-26-11:43:53, flynx)

Hello everyone. I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale operation, so I configured A2Billing for not to answer the call nor play any greetings or balance notifications to the caller. I'm authenticating each customer by it's IP address, and each customer has it's own context, in which I set the following: ;=====in extensions.conf======

Hello everyone. I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale operation, so I configured A2Billing for not to answer the call nor play any greetings or balance notifications to the caller. I'm authenticating each customer by it's IP address, and each customer has it's own context, in which I set the following: ;=====in extensions.conf======

Hello Everyone. In my outbound contexts, I'm using "${DIALSTATUS}" to fail over to other routes if the chosen route rejects the call. Now, My current scenario is if I get "BUSY" back from the first provider, I send a busy back to my customer. If I get something like CHANUNAVAIL (Like a SIP 503) I advance to the next carrier and attempt the call. This works

All my calls are getting DECLINED when I am trying from xlite : CLI shows : May 18 00:00:32 WARNING[4617]: channel.c:2781 ast_channel_make_compatible: No pa th to translate from SIP/cc101-b790c1d8(4) to SIP/sip19-090e87d8(256) May 18 00:00:32 WARNING[4617]: app_dial.c:1628 dial_exec_full: Had to drop call because I couldn't make SIP/cc101-b790c1d8 compatible with

In my dialplan I'd like to send a "603 Declined" message to the user placing the call. I see the commands for the Busy and Congestion, but not the one for the Declined. Any help? Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130726/5ac93551/attachment.htm>

Hi all, Hi have one computer hosting several VM, this particular server is in need of maintenance and has to be shutdown. In order to avoid disrupting the services provided by the VM I intended to migrate the VM to another computer. Computer 1 and 2 do not share any storage. So it is necessary to migrate not only the memory but also any data on the hard-drive. Both computers are running

Hello everyone, I've had the time to examine some specific serial/parallel forking scenarios with Asterisk lately. Looking at chan_sip it appears that anytime Asterisk wants to tear down a call before it's brought up, it sends a 603 Declined: } else { /* Incoming call, not up */ const char *res;

hello everybody and happy new year! I am a newbie in libvirt use, and I use it mostrly throught OpenNebula. Anyway, I noticed that one of the nodes can be a destination of a livemigrate but not the source. This is the environment: 4 blades nicknamed(*) red3, red9, red10, red11, each with 8 Intel E5506 cores (@ 2.13GHz) running CentOS 5 and KVM as hyoervisor. (*) these are names I

Could someone help me with the following code snippet. The results are not what I expect: > Sheet1$Claims[1:10] [1] NA 1 2 NA NA NA NA NA NA NA > Sheet1[1:10,"SubmissionStatus"] [1] Declined Bound Bound Bound Bound Bound Declined Dead Declined [10] Not Taken Levels: Bound Dead Declined Not Taken > Sheet1$Claimsnum <- NA >

Hi, Is virsh able to access the VMX config of a virtual machine running on a esxi host? I would like to take backups of my virtual machine configs and the doku section "Converting from VMware VMX config to domain XML config" (http://libvirt.org/drvesx.html) comes close to this - but does not tell how to get access to the vmx file. I would prefer a solution where ssh access to the

Hi, I have some problem with my provider. While the sip registration is successful, i intermittently encounter problem in dialing out. I receive 603 Declined error in my Sjphone client. The asterisk log shows line is busy/congestion. Appreciate if help or direction can be provided. Thanks. CK -------------- next part -------------- An HTML attachment was scrubbed... URL:

Asterisk Project Security Advisory - AST-2019-004 Product Asterisk Summary Crash when negotiating for T.38 with a declined stream Nature of Advisory Remote Crash Susceptibility Remote

I'm trying to connect to vbox hypervisor on an Ubuntu 10.04 machine through libvirt java binding (libvirt-java-0.4.6) by simply invoking: Connect conn = new Connect("vbox:///session", false); but I got this exception: libvir: warning : Failed to find the interface: Is the daemon running ? libvir: Remote error : unable to connect to '/var/run/libvirt/libvirt-sock':

> On Dec 17, 2009, at 8:50 PM, dathead2 at gmail.com wrote: > > On Dec 17, 2009, at 8:42 PM, Ron Decline <rutlecorps at gmail.com> wrote: > >> Is it possible to switch the Left / Right channel when encoding in FLAC? >> (I have some flac files with incorrect left/right channel assignment that I would like to re-encode with flac) >> _____________________________

Hello! I'm facing the following scenario: - Initial call opened to asterisk: SDP g722,alaw,ulaw - Outgoing call to provider started with Invite / SDP alaw, g726 and g729. - Provider sends 183 Session progress SDP: g729, alaw - Provider sends g729 rtp packages But: there is no license to transcode g729. What is asterisk doing? Asterisk decides to stop the call at all: - Sends cancel

On Tue, Oct 23, 2018 at 1:35 PM Dan Cropp <dan at amtelco.com> wrote: > The res_pjsip_transport_websocket failing to load seems to be a conflict > with the chan_sip.so loading. > > When I make the chan_sip.so not load, res_pjsip_transport_websocket.so > does load. > > We have customers who need chan_sip and chan_pjsip, so we need to load > both. Is there a way to

...e Galaxy name in well-known catalogs 20- 21 I2 h RAh Hour of Right Ascension (J2000) 22- 23 I2 min RAm Minute of Right Ascension (J2000) 24- 27 F4.1 s RAs Second of Right Ascension (J2000) 28 A1 --- DE- Sign of the Declination (J2000) 29- 30 I2 deg DEd Degree of Declination (J2000) 31- 32 I2 arcmin DEm Arcminute of Declination (J2000) 33- 34 I2 arcsec DEs Arcsecond of Declination (J2000) 36- 40 F5.2 kpc a26 ? Major linear diameter (1) 42- 43 I2 de...

I just noticed this upon startup since updating from 15.6.1 to 16.0.0 - do any of these matter? [Oct 18 12:12:18] WARNING[4489]: loader.c:2228 load_modules: Some non-required modules failed to load. [Oct 18 12:12:18] ERROR[4489]: loader.c:2243 load_modules: res_pjsip_transport_websocket declined to load. [Oct 18 12:12:18] ERROR[4489]: loader.c:2243 load_modules: cdr_sqlite3_custom declined to