similar to: SIP. Call-limit dialstatus

Hi Richard, Thank you for your response. But after I remove the parameters of dial command (tTkK). The call was still not native bridge. Let me know if you have any suggestion. Best regards, Charles 2015-01-30 0:34 GMT+08:00 Richard Mudgett <rmudgett at digium.com>: > > > On Wed, Jan 28, 2015 at 8:27 PM, Charles Wang <lazy.charles at gmail.com> > wrote: > >> Hi

On Wed, Jan 28, 2015 at 8:27 PM, Charles Wang <lazy.charles at gmail.com> wrote: > Hi all, > > I want to test the Native Bridge mode of DAHDI (FXS/FXO). I use asterisk > 11.14.2 and DAHDI 2.8.0. > > I try to set callwaiting = no AND callwaitingcallerid = no in > chan_dahdi.conf. > But I can't find native bridging information from CLI(opened debug mode in >

Hi all, I want to test the Native Bridge mode of DAHDI (FXS/FXO). I use asterisk 11.14.2 and DAHDI 2.8.0. I try to set callwaiting = no AND callwaitingcallerid = no in chan_dahdi.conf. But I can't find native bridging information from CLI(opened debug mode in logger.conf). How can I test the dahdi_bridge in native bridge mode? I use normal dial command ex: Dial(DAHDI/2,30,tTkK) to dial from

I am running asterisk 1.4 and the Digium GUI SVN-branch-2.0-r4489. When one phone calls another, I see the following on the console (here, 6223 dials 6123) -- Executing [6123 at DLPN_DefaultDialPlan:1] Macro("IAX2/6223-10489", "stdexten|6123|SIP/6123&IAX2/6123") in new stack -- Executing [s at macro-stdexten:1] Set("IAX2/6223-10489",

I'm having some trouble with Polycom Soundpoint phones. I have had good luck deploying them on a local network, but now I've tried putting some in place which access their * server across the network. The * server is on a public IP and the polycoms are behind a NAT on a cable modem broadband connection. Every so often I get: May 27 16:12:08 NOTICE[29728]: Peer 'Polycom1' is now

Hi Denis, Follow the output. TOP 29723 root 20 0 1617024 487668 383560 R 99,7 6,1 54:25.11 samba Service: PID ----------------------------- dnsupdate 29734 cldap_server 29727 rpc_server 29723 rpc_server 29723 rpc_server 29723 rpc_server 29723 rpc_server 29723 rpc_server 29723

Hi Team, Is there any body got any experience on setting up a virtualized environment in which the vm's can access a fiber channel SAN storage connected to host? the host access the SAN through its own HBA, but the hba is not recognized inside the virtual machines. Please let me know the step to go through this. Regards -------------- next part -------------- An HTML attachment was

Am using Asterisk 11.2 in one location and 11.1 in another location. when I trunk between two servers, the status is unreachable. But with different server with 11.2 and 11.2 it works fine. I tried both IAX and SIP. the trunk in sip.conf what i have is, [serverb] type=friend username=serverb secret=serverb host=10.10.10.5 port=5060 context=default insecure=port,invite dtmfmode=rfc2833

i could not figure out a better topic but i converted a qcow2 image to raw and tryed to use the old qcow2 file again but no bootable device the image worked perfectly before i converted it ...................??????? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Tks Rowland, its work. But same with the new installation the server goes up for 100%. We dont have the Iptables enabled. Maybe if enable the iptables the cpu is goes down. what do you think ? 2016-08-23 16:42 GMT-03:00 Rowland Penny via samba <samba at lists.samba.org>: > On Tue, 23 Aug 2016 16:06:19 -0300 > Maiquel Consalter via samba <samba at lists.samba.org> wrote: >

Hello, Just a very quick message to let you know that I have added a higher quality setting (-h2, +10Ko) and I have added now residual coding on the 512x256 wavelet image for the higher settings.Precision is a little better.-I also still apply the post-processing function on the 512x256 wavelet image for now.- Here is the link for the binaries (as I can not link them in my blog for now):

Thanks to everybody who tryied help me and in some way.. did it. Steps: 1) I updated the wine version to wine 1.1.27 on august 12 2009. 2)After that, I installed mdac_typ.exe from http://www.microsoft.com/downloads/details.aspx?FamilyID=6c050fe3-c795-4b7d-b037-185d0506396c&displaylang=en 3) Copied the ntwlib.dll from a windows system and pasted it on system32 folder of wine. 4)executed

Hi, I am using asterisk-1.4.15, My sip configs is like [2501] type=friend username=2501 secret=2501 canreinvite=no host=dynamic dtmfmode=rfc2833 context = sip disallow=all allow=ulaw incominglimit=1 nat=1 queue.conf is like [gen-enq] joinempty = yes musiconhold = default strategy = rrmemory servicelevel = 60 timeout = 60 retry = 5 wrapuptime=5 announce-frequency = 90 announce-holdtime = yes

Does anyone know what can cause queue members to go into a status of "Unknown"? pbxtel-01*CLI> queue show cs has 2 calls (max unlimited) in 'rrmemory' strategy (24s holdtime), W:0, C:447, A:20, SL:91.7% within 60s Members: SIP/1405 (dynamic) (Unknown) has taken no calls yet SIP/1420 (dynamic) (paused) (Not in use) has taken no calls yet SIP/1442

hi! $ dovecot --exec-mail ext /usr/lib/dovecot/expire-tool.sh /usr/lib/dovecot/expire-tool.sh: 9: Bad substitution What's the problem, do I miss something? This is the first time, I trying to use expire. This is an Ubuntu 10.04 . Thank you, tamas

I'm trying to write a dialplan that will allow me to "stress" test it. I want to be able to dial an extension, or pretend that the extension is busy or out of order (so that I can see what to do) given the dialplan snippet: [outbound] exten => _X.,1,NoOp(${TEST}) exten => _X.,n,Dial(SIP/${EXTEN}) exten => Busy,1,Busy(2) exten => Busy,n,Hangup() exten =>

Hi, according to the wiki the value NOANSWER for the channel variable DIALSTATUS means: No answer. The dial command reached its number, the number rang for too long, then the dial timed out. In out dialplan we grap all these events with exten => s-NOANSWER,1,Playback(sometext) exten => s-NOANSWER,2,WAIT(1) exten => s-NOANSWER,3,Hangup() The dial commands for internal and external

Hi there, could anybody tell me if the info below is still correct: Note: In order to obtain useful DIALSTATUS information when dialing a peer you will need to have qualify=yes in that peer's definition (e.g. in sip.conf or iax.conf). http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS THANKS!! -- Regards, Rustam Kovhaev

I am having a problem with ${DIALSTATUS} and )=CDR(disposition) disagreeing. Below is a dialplan snippet and the resulting CLI output. This is running in an 'h' extension. Noop(DIALSTATUS=${DIALSTATUS}) Noop(CDR(disposition)=${CDR(disposition)}) -- Executing [h at pbxmax-dial-simple:1] NoOp("SIP/msx_01-0000005b", "DIALSTATUS=ANSWER") in new stack

Hello All, Just a very simple example. I'm trying to make a call to a busy phone number using Dial application. -- H.323 call to 12345 with codec ALAW -- Called 12345 -- OH323/L5663 is ringing -- H.323 call 'ip$localhost/5663' cleared, reason 18 (Remote endpoint is busy) -- Hungup 'OH323/L5663' Everything is fine - "Remote endpoint is