similar to: FW: re: Asterisk Outbound with Failover, alarm notification, dial status and hangupcause capturing to CDR from Dialplan

Greetings list, Over the last few days I've been gearing up to replace a couple of our servers with 1.6 as something of a testbed, but I'm encountering a few problems, and wondering if anyone can help... In extensions.conf, there are a number of contexts defined for each group of users, along the lines of: [groupa] [groupb] etc. In each of those, there's a command include =>

Hello. Browsing accross subnets fails because "route2-3" can't sync its local browse list with the domain master browser "classix", which is also the WINS server. Everything else is fine. There is 1 and only 1 WINS server (classix) and no NT server. The error message in nmb log of route2-3 is: ==================================

https://bugzilla.netfilter.org/show_bug.cgi?id=1386 Bug ID: 1386 Summary: nftables.py cmd doesn't read updated counter values after first read Product: nftables Version: unspecified Hardware: x86_64 OS: Gentoo Status: NEW Severity: major Priority: P5 Component:

https://bugzilla.netfilter.org/show_bug.cgi?id=1382 Bug ID: 1382 Summary: nftables.py cmd leaking memory when ruleset contain mapping ip length to range with high limit 65535 Product: nftables Version: unspecified Hardware: x86_64 OS: Gentoo Status: NEW Severity: major

Hi all, I have setup two multipath route tables on my system for doing failover routing, What I want it''s that if GW at route1 of the MP is dead, traffic goes by route2, for doing that I have created the multipath routes as follows: ip route add table mail.traffic proto static nexthop via ${GW1} dev eth1 weight 1 nexthop via ${GW2} dev eth1 weight 250 But it does not run as I

Bill Venables writes: > Are the scoping differences between R and S set out precisely and > definitively somewhere? This would be useful. In the source code perhaps? :-) You can find a pretty precise description in the article Robert and I did in JCGS. Actually its pretty simple. Functions have access to the variables which were in effect when the function was defined. f <-

Hello, Fast question: from DomU I cannot ping Dom0, I only get 19:17:33.573370 arp who-has 192.168.10.1 tell 192.168.10.150 19:17:34.573421 arp who-has 192.168.10.1 tell 192.168.10.150 using tcpdump in Dom0. Why? --------------- Detailed question: I''m setting up a Xen virtual server. The environment is: -HVM -AMD64 -Debian Etch and using Xen from Debian repository (so Xen 3.0.3,

Dear list, I'm trying to write my first looping function in R. After many hours of searching help files and previous posts, I'm at wits end. Please forgive my programming ignorance...any help is greatly appreciated. I need to sort through a vector (x) and identify the point at which 2 successive values become smaller than the previous value. I've written a "while"

Hi Can you please help me make this into one statement... It doesn't work if I say _9000[1-9]0[1-8]. Also would like to be able to achieve _9000[1-9]0[1-8]XXXXXXXX, Asterisk 1.4 exten => _900010[0-8].,1,Goto(route1,${EXTEN:5},1) exten => _900010[0-8].,2,Hangup exten => _900020[0-8].,1,Goto(route,${EXTEN:5},1) exten => _900020[0-8].,2,Hangup exten =>

> > > > We may need some additional info. What kind of additional info? > I haven't put a ton of thought into > this, but I'm hoping we can either (a) use debug info as is or add some > extra (valid) debug info to support this, or (b) add an extra > debug-info-like section to instrumented binaries with the information we > need. > I'd try this data

Hello all, I am new to Linux Ethernet bridging. Let me first start with what I am trying to achieve. Well you see - I am attempting to have 2 main firewall running at the same time - one as a master and the other one as a slave. Yes, I would like to make use of Ethernet bridging in this scenario - as I understand it, all I need are two machines and STP enabled. I am running Debian

I just tried to setup another domain on a server running 1 domain. When i try to start the domain it gives me an error Error: Device 0 (vif) could not be connected. Backend device not found. The entry in my xen config file for the disks is disk = [''phy:/dev/xen001/xen2-fun,sda1,w'',''phy:/dev/xen001/xen2-fun-swap,sda2,w''] Looking at /var/log/xend.log, i found

Hi, Is there any way to tell Asterisk not to generate additional headers like: X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 I can't find any relevant option in sip.conf file :-( Thanks for help. Chris

Hi, I'm trying to get HANGUPCAUSE on my cdr the problem I'm facing is that this option: endbeforehexten=yes is not working at least on asterisk 1.6.0.1, so if I put yes o no I cant set CDR value with that value. It seems to finish the CDR record before h is executed. I'm using cdr_mysql. Any idea?? Thanks!! -------------- next part -------------- An HTML

Hi, I am using Asterisk: 1.6.1.20 LibPRI: 1.4.11.4 DAHDI: 2.3.0.1 Echo Canceller: MG2 Wanpipe-Driver: 3.5.15 Sangoma-Firmware: 43 (A104d) I handle some calls with my own PHP-AGI-Script. After a dial-command I use "GET FULL VARIABLE ${answeredtime}" or "GET FULL VARIABLE ${dialstatus}" and get valid information. Sometimes "dialstatus" has the value

I am trying to include the ${HANGUPCAUSE} in my mySQL cdr tables. I have a field called cause_code but it won't write. I belive it is because the record has already been written by the time I hit the h section of the code. How might I get this info into the CDR. I need this info for Quality of Service as well as route checking. Any ideas would be apperciated. Here is my dial line and my h

Today I get the remark that a call got disconnected after 10 minutes. This what my VERBOSE-logfile tells me : [May 18 15:36:30] VERBOSE[3940] logger.c: -- Executing [00493516426 at intern:1] NoOp("SIP/51-b76023b8", "Gesprek naar GSM-nummer via Telenet") in new stack [May 18 15:36:30] VERBOSE[3940] logger.c: -- Executing [00493516426 at intern:2]

Hopefully I'm not just missing some little detail here. We're trying to set the HANGUPCAUSE on SIP channels to have our softswitch play the proper recording instead of answering the call on Asterisk to play the message. It appears that no matter what the HANGUPCAUSE is set to, Asterisk always just sends "603 Declined". I looked through the source code briefly and it appears

Hello, On your BRI or PRI's what do you guys get as HANGUPCAUSE when dialing an unalocated number? I always get 3 (no route) which is less than helpful.

** THIS IS NOT WHERE YOUR REPLY BELONGS ** On Wednesday 25 Mar 2015, Salaheddine Elharit wrote: > tnaks for your response but the number dialed exist and i can call this > number when i configure the trunk directly in x-lite and i call call also > this number from my cell phone . > any help > thanks and regards Make sure you are sending the number in the correct format, when you