similar to: asterisk-users Digest, Vol 125, Issue 33

Hi, (please excuse me for lack of proper jargon usage and the vagueness of description...) i use Asterisk 11.12.1, (well... as included in FreePBX), I have several extensions that can register 2 separate devices (chan_sip) ( FreePBX calls this Devices & Users mode : Users are extension/internal number, devices are the 'SIP Accounts' for the internal 'endpoints' ) (this

I have the very same situation in one of my networks. To solve this you can dial out from the softphone and to move call to the phone you can simply transfer call to the same user (just if you were transferring call to yourself and the other device will ring. While, as you notice, you cannot dial a device, you can surely call your user to tranfer from a device to another. Please note that call

Greetings to one and all on this fine list; We have the current system: Meridian 1 Option 11 +-------------------+ | | ISDN/30 (DASS/2) ===> |NTAK79BB (2MB Pri) | | |<-->4x16 port Digital / 1x16 port Analogue ISDN/30 (EUROIDSN) ===> |NTBK50AA (2MB Pri)

Hi all If I have the following in my dialplan: exten=>25001,hint,SIP/25001 Doing a core show hint 25001 results in 25001 at local : SIP/25001 State:Idle Watchers 0 1 hint matching extension 25001 in the Asterisk CLI. What does the Watchers 0 mean? I use the hints table output via core show hints for logic in my dialler application - but

Hi, Any of you have any experience with SIP softphone dialler that capable of local recording? (recording to files in harddrive) So far I only know eyeBeam and Express talk. eyebeam fine but there are known error with recording. Express talk recording looks ok, but sometime it doesn't have incoming voice with *. Cheers Benni-

For some reason we are seeing "Avoiding deadlock for channel" in our Asterisk logs, the logs are getting filled up with an amazing speed around 12000 lines a second, and all of them are "Avoiding deadlock". What could be the potential reason for this to be happening? The Asterisk is used as auto dialler, therefore different channel types are involved SIP, DAHDI, Local's.

On Mon, Dec 29, 2014 at 7:26 AM, Lukasz Sokol <el.es.cr at gmail.com> wrote: > As the handsets have no LCD's to show the dialled number, > I want to give the workforce the ability to dial OUT using the softphone, > (as in, copy/paste the number from the CRM software into softphone then > *immediately* transfer the originated call 'endpoint' to the handset of > the

Sounds like a job for TAPI. Google TAPI for Asterisk or Asterisk TSP I've been playing with SIPTAPI and it works pretty well. It's very simple to install and set up. Hope this Helps PC... From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ryan Wagoner Sent: Monday, December 29, 2014 9:52 PM To: Asterisk Users Mailing

Microsip (Windows) is free and small. 2.5Mb download, 10Mb RAM usage, does everything I need and configuring TLS is a doddle. http://www.microsip.org/ On 16 February 2017 at 13:04, Max Grobecker <max.grobecker at ml.grobecker.info> wrote: > Hello, > > I'm a big fan of PhonerLite. > It's more poplar in Germany, but also available in English language. > This client

We are having a situation on our dialler here where our agents are claiming that when they receive a call because it has been answered, it seems as if the call had been answered several seconds earlier - IOW, they are hearing "hello ? Hello ?" and often hear the phone being put down as an initial part of the call. We have verified this by checking the voice recordings. Yet, the logs of

I had my samba printer shares working after many hours of struggling with the samba.conf file. I was happy. Now the printer shares have quit working. I checked the yum log to see if anything had been changed. Nothing applicable seems to have been changed. I've been surfing the net for the last couple of days trying to find a solution to the problem but I find many outdated and

Hello, can someone recommend a good and free Softphone for Windows which does not display advertisments inside the program? We have X-Lite but free version display advertisments. thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140608/c3a9e887/attachment.html>

Hello List, I'm working on an autodialer project. At the moment I use the Originate application then I "throw" it to an extension where I Dial() the other party and then both legs are bridged. The problem is that the Dial() will only run after the Originate finish its bit and I have lots of wasted time or even worse, the remote party hanging the call because instead of a human

Hi, I am using XML::readHTMLTable and getting the below error. Does anyone know why? Does this function not work with https? I didn't see anything in help about that. > library(XML) > wampage<-readHTMLTable('https://hr-workforce-analytics.llnl.gov/wf_pi_pop.html',1) Error in htmlParse(doc) : File https://hr-workforce-analytics.llnl.gov/wf_pi_pop.html does not exist Dan

CentOS 6.2 [mlapier at mushroom ~]$ uname -a Linux mushroom.patch 2.6.32-220.4.1.el6.i686 #1 SMP Mon Jan 23 22:37:12 GMT 2012 i686 i686 i386 GNU/Linux [mlapier at mushroom ~]$ Applicable sections of smb.conf: [global] workgroup = MYGROUP server string = Samba Server Version %v log file = /var/log/samba/log.%m max log size = 50 security = user

Would greatly appreciate any input into this currently-unanswered question on the forum: http://forums.asterisk.org/viewtopic.php?f=1&t=96496 I posted it on Jan 6th, have tried so many things, so much forum/list searching and late nights since, but have had to admit defeat. Rather than duplicate it all here, I've posted my logs and conf files on that thread, too. Problem is that while

Hi first post, so hope I'm not violating protocol. Been using Asterisk as home phone and hobby use for nearly 10 years. I love this project. Anyway, would someone mind verifying my pjsip.conf ? This seems to work well for 14.3.1 but I get no rtp into my natted Linphone when I upgrade to 14.4.1. Other than that the phone registers properly on 14.4.1. I can provide a pjsip log as well,

Hi, I keep getting the below error regarding rJava which is required by package xlsx (I have used it in the past to directly import data from Excel 2010). I was on R version 2.15.0 when I was getting this error this morning. So I upgraded to 2.15.1 but still the same problem. I tried unstalling and reinstalling xlsx and even rJava directly from the source as indicated here:

I have a samba server acting as a domain controller. Is there a way that I can Have a script that delete the shortcuts on the desktop,quicklaunch and startmenu for Internet Exploder. At the same time installing Mozilla Fire Fox. Maybe like a little vbscript or something that gets ran from the server when they login. Thanks

Hello using asterisk 1.8.32.3 I am not able to make a call with video support. I do not know what I am missing to make this video call. Codec h264 should be supported. sip*CLI> core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INT BINARY HEX TYPE NAME