similar to: Ericsson MD-110 and Dig-410

Hello, Most traditional PBX-es have the ability to use external audio source (e.g. radio tuner) for music on hold. This is also useful because you can let your users listen to radio by dialing some extension. I wanted to achieve the same on asterisk, and chan_alsa seemed the logical choice. I installed ALSA drivers, connected the radio to line-in and added the folowing to extensions.conf: exten

Hello, for a long time I've been having problems with analog modems and faxes communicationg over ISDN BRI lines. Now I began to suspect that this is due to * being out of sync with the PSTN. I have a quadBRI card with first two ports connected to PSTN, and defined as follows: span=1,1,0,ccs,ami span=2,2,0,ccs,ami This should mean that spans 1 and 2 are used as primary and secondary

+Lang Ultravague answer: There are a few different Orc layers for different levels of indirection needed for different levels of substitutability. One way is to indirect every call through global function pointers - so when you want to replace the function you write the new function pointer to the global variable. I forget which layers do which kinds of indirection. - Dave On Thu, May 26, 2016

Hello, I am trying to figure out how to recompile functions multiple times during run-time with ORC JIT and I'd appreciate any help/advice. My use case is t he following: every time a function of interest (annotated) is called, profiling data are gathered. Given enough data the function is recompiled using different optimizations. This happens repeatedly until the "best"

Hello everyone, I've been using LLVM for quite some time now (for writing my own toy compilers) and I've been following this mailing list for the last few months. I would like to start making contributions to the project. I have no particular area of interest right now (in the sense that I find pretty much all of LLVM fascinating). Are there any known low hanging fruit that I could tackle

Hi list! I'm trying to configure the Kirk IP 600 wireless (DECT) system. The wireless phones are regular DECT phones, the Kirk IP 600 is doing the voip part by registering/gatewaying the phones to a callmanager server. The phones do not work and I think the problem is that they do not register at asterisk but I'm not sure because I don't understand a thing of the sccp/skinny

Iam a novice R-user and I have a few questions: 1) How can R import an Excell data file? 2) I have a file.s file which it seems I can open but I can not load it i.e. when I write the file name in the command line, for e.g. >file.s it gives me an error message thank you for your consideration, Theo Nicoleris

H Theodoros, David, What I am trying to figure out is how the function pointer can be updated > to point to the newly compiled version. As Dave implied, it depends on how you've set up your ORC stack. These days, the API directly responsible for this is the IndirectStubsManager (see llvm/include/llvm/ExecutionEngine/Orc/IndirectionUtils.h). If you have access to the appropriate

Is there a way to execute specific applications at Asterisk startup (like "startup context" or something)?

I got a bunch of configfiles that look like this: mgmt-dev:/etc/nut# grep -v 'niets' * ups.conf:[ups] ups.conf:driver=newhidups ups.conf:port=auto upsd.conf:ACL all 0.0.0.0/0 upsd.conf:ACL localhost 127.0.0.1/32 upsd.conf:ACL remote 192.168.110.0/24 upsd.conf:ACCEPT remote upsd.conf:ACCEPT localhost upsd.conf:REJECT all upsd.users:[monuser] upsd.users:password = mypass upsd.users:allowfrom

Signed-off-by: Arthur Clement <aclement at linagora.com> --- scripts/ovirt-functions | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/scripts/ovirt-functions b/scripts/ovirt-functions index 7b95ea8..9e96f7b 100644 --- a/scripts/ovirt-functions +++ b/scripts/ovirt-functions @@ -186,7 +186,7 @@ find_srv() local search=$(grep search /etc/resolv.conf)

bit general linux this one but using dig is it possible to get a list of everything in the zone? Presume this is if the network you are on is allowed transfers from the bind server but if i am not?? thanks

On 9/12/2016 3:23 PM, ash-samba--- via samba wrote: > On 09/09/16 16:35, lingpanda101--- via samba wrote: >> On 9/9/2016 10:59 AM, ash-samba--- via samba wrote: >>> We appear to have some phantom DNS records on both our domain >>> controllers. >>> [...] >>> # dig _ldap._tcp.dc._msdcs.chester-dc.example.com srv @10.4.4.155 >>> [...] >>>

> And to get it editable: > > ldbedit -e nano -H /usr/local/samba/private/sam.ldb --cross-ncs > --show-binary -b > 'DC=_ldap._tcp.pdc,DC=_msdcs.samdom.example.com,CN=MicrosoftDNS,DC=ForestDnsZones,DC=samdom,DC=example,DC=com' > -s base > > All on one line, run on a DC. > > Rowland The problem is that when I save and exit (even if I make no ldif changes), I

I have been asked by a potential customer whether we can connect an Asterisk box to an Ericsson MD110 that has an E1 port with ISDN-USR. They are unable or unwilling to upgrade their E1 port to QSIG. Has anyone here had experience of successfully making such a connection? I have found a couple of hits on Google that suggest it "should" work, but I'm after something a little more

Hi, Anybody using this Ericcson analog phone with Asterisk: Ericsson dialog 4187? I was told some functionalities like CLID will only work with an Ericsson PABX but other than that I would like to hear from anybody using this phone on a FXS port. Thanks, Jose Limeres -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does anyone have info on using the Ericsson Webswitch 100 Model G4 with *? The documentation and brochures make it sound like it's some kind of specialized FXO device purely to provide legacy PBX with access to VOIP, and not a general purpose FXO-VOIP gateway. Is it possible to use it to route incoming PSTN calls into * with full callerid, and establish outgoing calls from * to the PSTN

My asterisk server has an external IP. Also my Ericsson DRG22 has an external IP. Tried setting up the Ericsson DRG22 device. *SIP Server IP (primary): MYIPNUMBER User name: drg22-2 password: password Caller ID name: drg22 Telephone number: 12345 *Telephone domain name: ?? (I only have stighost.someotherdomain.com ) Do I have to run my own dns on the Asterisk server? Port: 5060

I am trying to use TE110p card for Euro ISDN with Ericsson AMS switch. I consistently get one of the following errors: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 or PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 My zaptel.conf file: -------------------------------- span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone=us defaultzone=us

Hello, This is a little off-topic. I have an Ericsson FCT f251m, according to the specs it supports call signalling through polarity reversals and loop break, but it's currently disabled. On my PSTN line, my TelCo does send polarity switchs to signal answer and hangup (answeronpolarityswitch=yes and hanguponpolarityswitch=yes in zapata.conf), and asterisk detects it alright. I've been