similar to: Fwd: Re: Asterisk CLI unresponsive

Hello setting "nat=no" or omitting "nat=" in peer definition does not help either. Still no audio. Why do you think this is a NAT issue ? IP and port information in SDP-body is correct. Kind regards. On 12-08-16 09:25, ????? ?????? wrote: > > Try delete nat from 770000wrtc settings ice should do the same > > > On Aug 11, 2016 10:00 PM, "Jonas

On 11-08-16 18:03, Matt Fredrickson wrote: > On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kellens at telenet.be> wrote: >> My main reason not to upgrade to Ast 13 is because I'm afraid of losing >> functionality as there are certain functions deprecated/replaced. This can >> also cause headache :-) >> >> I will do so if there is no other option.

Hello list, I've got a whole lot of these in my debug log : [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both

Hi I am trying to implement monitoring of asterisk (all 4 spans-i want to show them line by line Up or down) using nagios using below script, but i always get the status as down and red..can anyone let me know how to read an output from nagios plugin ? nagios etc is configured already and is working PATH=/bin:/sbin:/usr/bin:/usr/sbin FAILS="" SPANS=$(asterisk -rnx "pri show

>>>>> Anupam Tyagi >>>>> on Tue, 9 Jul 2024 16:16:43 +0530 writes: > How can I do automatic knot selection while fitting piecewise linear > splines to two variables x and y? Which package to use to do it simply? I > also want to visualize the splines (and the scatter plot) with a graph. > Anupam NB: linear splines, i.e. piecewise

My main reason not to upgrade to Ast 13 is because I'm afraid of losing functionality as there are certain functions deprecated/replaced. This can also cause headache :-) I will do so if there is no other option. But still, I don't see why Ast 13 would differ so much in this case ? If ICE and NAT is working (not causing problems) why should Ast 13 bring me audio and Ast 12 don't

dear all, I apologize for my delay in replying you. Here my contribution, maybe just for completeness: Similar to "earth", "segmented" also fits piecewise linear relationships with the number of breakpoints being selected by the AIC or BIC (recommended). #code (example and code from Martin Maechler previous email) library(segmented) o<-selgmented(y, ~x, Kmax=20,

Don't be afraid about the info that I'm going to post in this mail, but I want you to give as much info as possible. Also I want to show you what I've tried. What do I want When a voicemail-message is left via the Voicemail()-application, I want the .wav-file send to my mail-address as an attachment. My mail-setup I'm not using sendmail as MTA. I have msmtp as MTA and mutt as

Hi I am trying to implement monitoring of asterisk (all 4 spans-i want to show them line by line Up or down) using nagios using below script, but i always get the status as down and red..can anyone let me know how to read an output from nagios plugin ? nagios etc is configured already and is working PATH=/bin:/sbin:/usr/bin:/usr/sbin FAILS="" STATUS=$(asterisk -rnx "pri

Zitat von Israel Gottlieb <isrlgb at gmail.com>: > If you the c option in the dial command it will send answered > else where sip message to the phone and most ip phones understand that > The cell will always display a missed call? I'm very sorry, but I can't understand what you mean... Could you explain, maybe with an example? Thanks Luca Bertoncello (lucabert at

On 16-08-16 17:45, George Joseph wrote: > > > On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > On 16-08-16 04:38, George Joseph wrote: >> >> >> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens >> <jonas.kellens at telenet.be <mailto:jonas.kellens at

To Jonas: I have an asterisk box at home and I have this line in my rtp.conf file: rtpstart=10000 rtpend=10100 And My FW is setup to forward all incoming ports of range 10000-10100 to the asterisk PC. I've never had a problem since one year, but I have never received more than two simultaneous calls with SIP clients. Message: 5 Date: Fri, 13 Sep 2013 11:49:59 +0200 From: Jonas Kellens

I've just installed centos 6.3 for the first time on a test machine and couldn't be happier - except that a critical wireless adapter driver is missing - the r8712u. I would love to convert my mythtv machine from Fedora 17 to centos 6.3 to put all that chaos and instability behind me. My mythtv machine connects only wirelessly to the rest of my network and uses a Rosewill RNX-N180UBE USB

Zitat von Israel Gottlieb <isrlgb at gmail.com>: Shalom, Israel! > Using chan_sip you need to create another ?user aand then dial both > > Using pjsip you can connect 2 devices Thank you. Unfortunately it seems that I don't have pjsip available as package on the OpenWRT where I installed Asterisk... :( I'll create another user. Thanks Luca Bertoncello (lucabert at

Israel Gottlieb <isrlgb at gmail.com> schrieb: > At the end of the Command you could use options one of them is the c (not > apital) which sends a cancel event to the phone > http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Shalom Israel, unfortunately it does not work as expected... I wrote: exten => _00493512222222,n,Dial(SIP/00493512222222&SIP/00493511111111,,Rc)

We are running asteriskNow 1.4.18 and after a few days it becomes unresponsive and inbound INVITEs timeout. We just reboot the box to resolve it. But it seems to be occurring more regularly now. I am hesitant to move to latest version, but will do if needed. Any guidance or troubleshooting modes I may use will be helpful. -------------- next part -------------- An HTML attachment was

On 16-08-16 04:38, George Joseph wrote: > > > On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > using pjproject 2.5.5 > using asterisk-certified-13.8-cert1 > > > IIRC there were API changes in pjproject 2.5 that aren't accounted for > in

Le 23/03/2017 ? 20:17, Jonas Kellens a ?crit : > Hello > > > is there any more information on how to reload/read musiconhold files ? CLI> module reload res_musiconhold -- Daniel > On 07-03-17 10:46, Jonas Kellens wrote: >> Hello >> >> I did not mention it but of course the MOH directory is listed in >> /etc/asterisk/musiconhold.conf : >> >>

Hi, I have a web server which is randomly becoming unresponsive. I've checked for hardware problems with no luck : - no ethernet errors - no disk errors - no memory errors So far I've managed to point the trouble at the web server since if I restart httpd (apache) it comes back to normal. This is not a shared server so I control (or at least like to think of) what is hosted so I have

I transferred a site from Lighty+SCGI to Lighty+Mongrel, and now I''m having problems with it becoming unresponsive. It works fine for about 10-15 minutes, but then attempts to access the site just stall forever. Nothing is written in any of lighty''s or mongrel''s logs. Here''s my lighty config: $HTTP["host"] =~ "www.xyz.com" {