similar to: External voice channels pack up

Displaying 20 results from an estimated 4000 matches similar to: "External voice channels pack up"

2016 Mar 04
3
vignette index
Dear helpers, I have multiple vignette files for a package, and I would like to have the "right" order of these files when displayed online. For instance, see below: https://cran.r-project.org/web/packages/bst/index.html The order of vignette links on CRAN is different from what I hoped for: > vignette(package="bst") Vignettes in package 'bst': pros
2005 Oct 01
2
Remote call pick-up
Hi, Does anyone have remote call pick-up working on * (either via SIP or otherwise)? If so then can you post your features.conf, sip.conf and/or zapata.conf? We can't seem to get this (seemingly simple) function to work. Cheers, Damian. -- FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz
2005 May 11
1
re:oh323 driver compiling problem
Hi. Download pwlib, openh323 Janus-patch at http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries Refer how to compile http://www.mail-archive.com/asterisk-users@lists.digium.com/msg86875.html Cheers. Kim. > > Message: 1 > Date: Wed, 11 May 2005 16:42:25 +0100 > From: Cesar Garcia <cesar.garcia@idecnet.com> > Subject: [Asterisk-Users] oh323 driver compiling
2017 Nov 23
1
[ot] Skype crackling (was: skypeforlinux lacks dependencies, won't update)
On 23/11/17 19:49, Sorin Srbu wrote: > The Skype flatpak at home worked beautifully, that is to say for about ten > minutes, then the sound (using a Koss SB/45 headset) it started crackling to > the point of not being usable anymore. I hung up and reconnected, then sound > was fine, for another ten minutes at which point the crackling returned. > Checking the sound prefs, I saw
2005 May 12
2
Best CPU config for dual-Xeon?
I have some beefy dual-Xeon servers that I will be using for Asterisk VoIP applications (i.e. no Zaptel cards). Using 2.6.11-1.14_FC3smp as the kernel (Fedora Core 3), and currently with Asterisk STABLE. My question is concerning the CPU setup, as I've seen conflicting or out-of-date suggestions: given the above config, should I have hyper-threading turned on or off? Turned on appears like 4
2003 Mar 04
2
S100U == DEAD !
We have a S100U which we were using as our sole internal phone extension. Today it died. Picking up the phone now results in a brief dialtone followed by bursts of random static and crackling noises and then a second or two of fast-busy followed by silence and more crackling noises. I've tried the hardware attached to a couple of different machines as well as a couple of different phones.
2014 Apr 18
2
On of mirrors doesn't work properly
Hello! curl 'http://64.235.47.134/?release=6&arch=x86_64&repo=os' <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2 Final//EN"> <html> <head> <title>Index of /</title> </head> <body> <html> <head> <meta http-equiv="Content-Type" content="text/html;charset=utf-8"> <title>CentOS
2006 Aug 09
2
Link to most recent
Greetings all, I have a side panel in my app that needs to have a link to the most recent 4 or 5 items in a table. How do I create this link? For example, I have a controller named widgets, and I want to create a link to :controller => "widgets", :action => "show", and then the most recent id. I want the text link to be the name of the widget. Any ideas? Obviously,
2006 Dec 14
3
IBM Server / USB Ports
Hi, I have an IBM xSeries server... and the digium card is sharing IRQ with USB and giving me crackling audio. >>> cat /proc/interrupts >>> >>> It brings up these results: >>> 0: 10566547 IO-APIC-edge timer >>> 1: 9 IO-APIC-edge i8042 >>> 2: 0 XT-PIC cascade >>> 8: 1
2003 Jul 09
2
Music on hold quality..
Hi, Does any one have any pointers on improving moh quality?? Symptoms are crackling and hissing as the sound comes and goes.. I installed mpg123 this morning.. I have tried various MP3's sampled at 160k, 128k, 32k and 8k and they all sounded terrible... The PC is a P4 so its got plenty of processing power.. I have tried a few different types of classical music (Piano, Violin and full
2005 Jun 08
1
Cisco 7960 mic generating noise on other end
Hi, I'm having a problem with one of our 7960. They all run latest 7.4 SIP firmware. The problem appears on the other end. The other end constantly hears a 'crackling' noise. I have tested using phone set, headset and speaker and the noise appears on all cases. I have other 7960 setup exactly same way (using same asterisk, firmware, etc) so it looks like a hardware issue.
2006 Aug 25
3
RJS Error: Element.update is not a function
Hi, I have implemented some RJS code and I have the same exact code in 2 different places. In one place I get the error "Element.update is not a function" and the other place works fine. Any idea why this error is coming? Breaking my head. -Vinod --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on
2008 Nov 01
9
Wine 1.1.7: Sound regression?
I have recently formatted my system in order to install Intrepid Ibex, and I am now using it with Wine 1.1.7. I did a backup of the prefixes of the games I was using in Hardy / Wine up to 1.1.5, and imported them in my new installment. The games work normally, but the sound is crackling all the time for each of them; something which was not happening in my former installment. I have not noticed
2006 May 17
2
IAX crackilng
I apologize about doubling these up, I forgot the subject! I have a cisco VPN from router to router over a Data T-1. The ping times are consistently 32ms with random ping responses of 295ms -408ms about every 30 secs to a minute, I have jitter buffer enabled. The connection goes like this... Mitel SIP phone to Asterisk A, IAX trunked to Asterisk B and then T-1 to a PBX, the calls are internal
2006 Mar 18
1
GS BT102 dual ethernet port -bandwidth impact
FYI for anyone using the dual ethernet ports on a Grandstream BT102. I'm using a BT102 connected to an HP2524 10/100 switch, which has an asterisk box connected directly to it. No VLANs defined or in use. Measured bandwidth: PC -> HP Switch -> Asterisk : actual throughput measured at 94.1 mbps. PC -> BT102 -> HP Switch -> Asterisk : actual measured at 8.86 mbps. The
2008 May 16
1
trixbox, sangoma a200, dell poweredge 2550 issue
Hi all, I have setup a Dell PowerEdge 2550 with a Sangoma A200 card with 2xFSO and 1XFS modules. The PowerEdge specs are 1 x P3 1133MHz, 512MB RAM. Sangoma A200 has 3 analogue PSTN lines connected. This server is based in Office 1, with 5 users all with a Linksys SPA942 VoIP Handset. There is another Office (Office 2) connected to here using VPN. There are two users in Office 2 with the
2005 Jan 21
3
IAX Inbound Sound Quality
I have a couple of DID's through VP Connect and have been having sound quality issues on incoming calls. During the call, the calling parties voice sometimes sound like it is crackling, in other words it is not very crisp. I would liken it to listening to a radio with a blown speaker. This sound defect comes and goes throughout the call. The other person is always audible but it just isn't
2006 Jun 16
3
Echo and crackle
We are running asterisk with a single POTS line for local calls and a voip line for long distance. Whenever we receive a call on the POTS line it is more than likely, but not always, going to have significant distracting echo. In addition to that there is occasional heavy crackle or static. I have tried to follow the guidelines at :
2012 Jun 26
4
ActionView::Template::Error (undefined method `strftime' for nil:NilClass)
Hello, I''m a newbie. I need help resolving this issue. I recently added a pdf to the newsletter admin section of the website and now I can no longer view page 2 of the list of pdf''s. Nor can I login to see the newsletters as a student. I''m using Ruby 1.9.3, Rails 3.2.1. Here''s the information from the log file. ActionView::Template::Error (undefined
2013 Nov 20
5
Movistar sip Mexico
Hello, I have a problem with movistar in Mexico with a sip calls. Movistar send to me T38 and G729 in the INVITE and they say that I have to ignore T38 and use G729 in the voice call. When a fax call is made Movistar send only T38 in the INVITE. Invite example: v=0 o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2 s=sip call c=IN IP4 192.168.1.2 t=0 0 m=audio 6370 RTP/AVP 18 101 a=fmtp:18