similar to: RE: SNOM 190 '486/Busy here' after upgrade to re 3.60s

Hi, Snom phones firmware 5.0 is now out. Try it if you like: http://www.snom.com/wiki/index.php/Main_Page. Regards, --------------------------------------------------------------------- Usman Tahir snom technology AG www.snom.com --------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a Snom 190 that refuses to accept calls after upgrade to firmware 3.60s, latest. I get "SIP 486/Busy Here". No change in the dialplan, nor settings in the phone. Calls out fine. I did 30 other phones yesterday with 3.60s with no problem, this is the only one. In the phone's log I get: [5]22/9/2005 17:01:00: timeout::callback: Registering with timeout of 0 ms [5]22/9/2005

I wanted to post this to everyone in case someone else ran into this problem! Firmware 6.5.1 (Current stable release) Asterisk 1.2.13 Call Completion: Off Peer to Perr call completion: On (canreinvite=no is set in sip.conf for each friend) Call join on Xfer (2 calls): Off Those settings seem to do the trick, I tested on Snom 320s with no errors at all, Im going to test on the 360s later today,

In the example I posted previous, there is an obvious gaping security hole, it would be trivial for someone to read the querystring and exploit it to make free phone calls, spoof caller ID (if you allow the CallerID to be set with a QueryString value), etc. You want to make damn sure that the URL is not publicly accessible or somehow obsfucate the querystring, or use POST. In my case, I

Looks like this phone has redirection or DND set. Anything on the display? If it still a mystery send us the settings of the phone, then it should become clear. BTW if you have a snom trouble ticket, you can also go to http://www.snom.com/onlinesupport.html (scroll down to set up an account). CS > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com >

Hi List, I cant seem to find the setting that changes this! You put a person on hold, they are on hold like normal, but after a few seconds the phone will then start having dialtone coming from the speakerphone, really weird!! Anyone know how to fix this? I see where it could be nice, but in this case, we just want them on hold is all, no dialtone! Any help would be great! Thanks! Ron

Colin, Thanks for your assistance. Reading over your advice I seem to still be a bit confused. My agents are not on the Asterisk server; it appears in your advice that my the call will travel this path: WWW interface --> agent enters their DID, platform to use, and termination DID --> AST calls agent --> Agent calls termination DID If my agents are not on the Asterisk server

Hi Guys, I am trying to add a Broadcom Physical NIC to a virtual Machine as Passthrough device. However I am getting following error. Error starting domain: unsupported configuration: host doesn't support passthrough of host PCI devices Traceback (most recent call last): File "/usr/share/virt-manager/virtManager/asyncjob.py", line 88, in cb_wrapper

Sorry , I forget it for another monitoring tool monit that we have used in our production systems to restart asterisk in case of asterisk crash or halt. I have attached a monit configuration for your reference. it will work almost in all cases This configuration will check Asterisk for following 1. will check for Asterisk process. 2. will check Asterisk via AMI 3. will check

Dear Usman, I no longer work on UPS development. You should direct your question to the nut-upsdev mailing list (cc'd above). See also http://lists.alioth.debian.org/mailman/listinfo/nut-upsdev Best wishes, -- Peter Usman Abid wrote: > > hello sir i hope you will be fine i have read your article on > Belkin Universal UPS protocol description i want to communicate with > my UPS

Anyone have a Snom they're happy with? How did you manage that? :) I have a system of: Asterisk 1.2.3 2 Wildcard TDM400P Rev I and E/F 1 Snom 360 + sidecar ~15 Sipura/Linsys SPA-841 ~15 Grandstream 101 Everything (currently) is on the same network, not a router to be seen between any two. Also everything, except the snom, is working sweetly. The main problem is ECHO.. awful echo and

Hello - 1. Is there any way to directly export R plots to MIF format for import into FrameMaker? 2. If not, how hard would it be to add such a device driver? I don't see device driver docs in the R extensions docs. Are drivers contained in their own source files with simple interfaces to the data structures with plot data? 3. If that's difficult, what workarounds are

Hi Everyone, Im usman from Pakistan Im looking for remote systems administration related work full time or part time or per project basis. I have worked for more than 6 years in Internet services providers/ISPs as Sr. Systems Engineer, Systems Admin and Network Engineer. Im using Linux since Slackware 8 and Redhat 6.x, have worked in many Linux distributions Redhat, Fedora, Slackware,

Dear R-users, Can some one help me out. I tried installing some packages in R_2.5.1 but I get an error message as shown below. This is unusual since I have been installing and updatings packages in the same version of R. > chooseCRANmirror() > update.packages(ask='graphics') Error in .readRDS(pfile) : unknown input format or > utils:::menuInstallPkgs() Error in .readRDS(pfile) :

Following new features are now supported by asterisk based telemarketing software Auto subscription / registration after call recipient press a key in voice broadcasting https://www.ictbroadcast.com/Subscription-Campaign-to-automatically-register-customers-at-website-with-Voice-broadcasting-Autodialer There will be restriction to call a number in off time accordingly to timezone of

Respected Sir, I am a MS CS scholar of Virtual University of Pakistan, I want to participate in GSoC 2016 for LLVM. Data Science, Networks, Information security, digital forensics and ethical hacking are my core areas of interest. Currently, I am working on a research project on live forensics of GPU and volatile memories like RAMs and Caches. I am looking forward your guidance to start my

Interested to know a good wholesale sip providers for 15k concurrent calls regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20231120/75b0e652/attachment.html>

Hi, I have two 3 D arrays. Both are of this form array_1<- array[n,n,k] array_2<-array[m,m,k] Lets say n=83 and m=80 Since n>m. I would like to add rows and columns to array_2 to make them equal. I want to keep the size of the third dimension fixed i.e.. k. i.e. if (nrow(array_1)>nrow(array_2)) { array_2[m:n,m:n,]<- 10^6 } But this doesn't work. I tried abind and rbind but

Though asterisk support AMD which is based on silence detection but I did not found support of tone / beep detection in asterisk to record a voice message for answering machines after detecting tone Will appreciate any help in this regard Best Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT Unified Communication Telemarketing

On Monday 20 November 2023 at 12:14:11, Tahir Almas Dhesi wrote: > Interested to know good wholesale SIP providers for 15k concurrent calls You might want to specify a bit more detail, such as: - which country are you located in - do you require inbound DDIs (if so, in which region/s)? - which countries' Caller ID/s do you need to present? Antony. -- These clients are often