similar to: "sip set debug on" output to file only (not to console)

Hello everybody, from time to time, we get so-called simplex / one-way audio calls, where one party cannot hear the other. The only thing in common is that is does happen with calls via SIP trunk, not ISDN and not internal calls. Nothing strange in verbose and SIP logs. Could even be some weird intermittent firewall issue I guess. Apart from logging all traffic 24/7 via tcpdump (not really

Hi Marie Thanks! I was just worried about thread safety if I had to use a global variable, e. g. it might be set to a value by one call (since I'm using the same global for every incoming call to transfer the accountcode gotten from my HTTP endpoint to the same macro, and there can be several calls simultaneously all inserting HTTP-sourced values at more or less the same instant) and then

Hi everybody, I'm looking for a solution for the following scenario: ? Asterisk queue ? At peak hours, there will be more callers then queue members/agents, so some callers will spend some time on hold ? Agents should be able to choose which of the on hold calls to answer instead of answering the next one in queue We already have a web interface where agents can see the callers on hold, so

On Tue, Apr 14, 2015 at 08:26:07AM +0200, Sebastian Kemper wrote: > On Thu, Apr 02, 2015 at 11:33:38PM +0200, Daniel Heckl wrote: > > I do not want set allowguest=yes. The problem is, there is no official > > list with ip addresses of Telekom Germany. But I think all ip > > addresses comes from the ip range 217.0.0.0/13. > > Hello Daniel, > > Judging by the lists

Hi all I need to pass a parameter in a thread-safe manner to the Queue pickup macro. This is to know when (and who) picked up an incoming call to a queue and log that to my back-office system with a CURL to a HTTP endpoint. However, the Queue application does not appear to allow passing of parameters to the called queue pickup macro. E. g. non-working code is: [queuetest] timeout = 60 retry =

Hi, is there any way to execute scripts in R (Windows) without the script being copied to the console, so that only error messages are reported? Or to have a second console in parallel? Please. Not being able to hit the F10 button like in S-Plus seriously impairs my productivity (not only because it is slow and clutters my console history). And totally unrelated: Is there any chance that R

hi, I have a really strange problem which is driving me crazy for days now. If I register my asterisk (tried all versions from 1.6 up to 13.x) with one sip registrar, everything works... calls go out and call come in... no 32 seconds limit. but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. as far as I know, there is no firewall in

Hello, Does anyone know how to conference a call on the SNOM 200 phone? Whenever I push the cnf/tran button it just hangs up on the active call. The manual says you have to push the cnf function key but it doesn't appear in the lcd on my phone. Thanks -Matt -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear list, Sometimes I have huge data.frames and the small spreadsheetlike edit.data.frame is quite handy to get an overview of the data. However, when I close the editor all data are rolled over the console window, which takes time and clutters the window. Is there a way to avoid this? Fredrik

I cannot seem to get any transfers to work at all. The console show I have #1 amd #2 set up for Blind and Attended Transfer, but when I hit these buttons on my handset nothing happens (other than I hear the dtmf tones on the other end of the line). roo*CLI> show features Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8

Hello all, I have seen some chatter again about DTMF. I see most of the talk about DTMF around not being able to get an external IVR to recognize digits, not a big issue for me at this time but sill interesting. My issue though, is with talk off on a zap channel. It seems to be getting worse or maybe my patience is thinning. All my calls go out and come in pstn through an FXO as I do not

I have a TDM2402E card. Occasionally I have noticed that a number I call that gives and IVR the DTMF keys are not detected. All other times the DTMF works fine. /proc/interrupts is: CPU0 CPU1 0: 500798020 500749281 IO-APIC-edge timer 8: 4 9 IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 15: 4505912 4502264 IO-APIC-edge

Hi Thank you for your support. The server is actually compromised, I discovered that after making a deep trace using the audit daemon and looking for the kill signal (SIGKILL) that terminates asterisk. I discovered that there is an executable with a random name in the /boot folder that is killing and deleting asterisk !!! This executable is launched by a service in /etc/rc.d/ with the same

Hi, We are considering using a ZFS based storage as a staging disk for Networker. We''re aiming at providing enough storage to be able to keep 3 months worth of backups on disk, before it''s moved to tape. To provide storage for 3 months of backups, we want to utilize the dedup functionality in ZFS. I''ve searched around for these topics and found no success stories,

I have a patch I'd like to commit that adds commentary to asm files about which TableGen pattern generated a particular instruction. The output looks like this: cvtpd2ps %xmm0, %xmm0 # source.c:39 # Src: (intrinsic_wo_chain:v4f32 927:iPTR, VR128:v2f64:$src) # Dst: (Int_CVTPD2PSrr:v4f32 VR128:v2f64:$src)

We've moved within the last two months to Asterisk 1.4.x All remote facilities are connected via highspeed (9mbit) connections (Over OpenVPN) to a central Asterisk box, acting as a voice router, that funnels all calls into our Avaya Definity G3R via PRI. When corporate employees visit the remote facilities and try to call the G3R's voice mail system(Audix), DTMF is not recognized unless

Hi, We are trying to track some specific rules using LOG as target. Everything is working well but the problem is that iptables is flooding the console with LOG messages. We tried --log level 4 on iptables rules but it didn't work. We fixed the problem changing KLOGD_OPTIONS value in /etc/sysconfig/syslog to: KLOG_OPTIONS="-c 4" Is it the best option or we are missing something?

I am trying to integrate a Nortel Norstar system with an Asterisk service using a TDM04B card (4 fxo). So far everyting works from Asterisk to Nortel. The problem is when someone dials from the Nortel PBX to the Asterisk server. Asterisk answers the call and provides a dialtone (with DISA) but appartently the DTMF tones are not passed to asterisk and the call cannot proceed. This only

I have an entry for console/dsp in the dialplan. When I call into that extension I get connected to the soundcard and I hear myself etc... everything is fine. However, if I call in and get connected then a second call comes in they also get connected. I was expecting them to get a busy signal or something... Do I have something configured wrong? How do I only get 1 person at a time on

Hey everyone, I''m trying to figure out how to manage my /etc/fstab with puppet on all my servers. Does anybody have an idea for me how to do? I tried it with augeas but I didn''t like the results I got so far... here is what I''ve tried: define conf ( $id = "01", $spec = " ", $file = " ", $vfstype = "