similar to: Issue with transfer (sip)

Hi, I'm playing around with asterisk 1.6.2.0 and the first try was to replace my now non-functionning 'app-realtime' macro which emulated RealTime with REALTIME_HASH() There is very few documentation on the subject except for this bug report: https://issues.asterisk.org/view.php?id=13651#c94998 However when i try this syntax:

Hi, I'm having a question with asterisk queue system, is it a fifo or a lifo or random ? Sometimes when we have people waiting in the queue and new agents are connected to handle the load the first call that is handled is not the one which is already waiting for 4min, but the new one which has just arrived. However this doesn't happens everytimes Is it normal ? regards, benoit

Hi, To force user to behave correctly, i want to make a process of disconnecting every member but one (special alarm phone) every day at a special time. I'm thinking of a cron job that will create a "call file" that will dial an appropriate extension to do the job, is this the correct way ? I'm facing two problems however, the call file system is here to connect a channel to

Hi, For some reason (outbound call tracking) I've got a few different outbound call process (using a macro for queuemetrics logging, or direct call) i wanted to factorise the routing process so i came up with something like the following. All in one it's working like expected, however every "ael reload" command trigger a lot of warning like that "application call

Hi all, i have the following setup PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk 1.6.2.9 -> SIP -> agent Does work quit fine - then agent does have the abibility to transfer a call to a third party - the agent can initiate the transfer over a web interface - it does generate a asterisk manager atxfer request... So agent does initiate transfer - call

Hi, I'm a little surprised, up until 1.4.22 my agents where using an IAX channel to ZoIPer Softphone, however since after the upgrade to .22 we experienced a problem with hangup failure between zoiper and asterisk (look like bug http://bugs.digium.com/view.php?id=13184) i made them switch to SIP instead. Weird thing is that the 'Not In Use' warning message keep showing

Hello, I'm having difficulty with registering a SIP account in a Snom 320 IP-phone. This is what sip debug tells me : [Oct 7 13:28:42] VERBOSE[20314] chan_sip.c: [Oct 7 13:28:42] <--- SIP read from UDP:public_ip:58697 ---> REGISTER sip:sip.domain.tld SIP/2.0 Via: SIP/2.0/UDP 192.168.114.200:2048;branch=z9hG4bK-vj1xvbdnp4dw;rport From: <sip:test3 at

Hello all - This is basically an updated re-posting of one I've posted a few days ago. Thanks to the kind help provided but I still can't make it work. But I'm moving a little further down the line (thanks to you folks). Basically, I've got an Asterisk server in a LAB ENVIRONMENT on my home LAN. The server has a Wildcard TDM400 installed but has no POTs lines/phones connected.

> The Buildbot has detected a new failure of Ubuntu-maverick-x86 on > Network UPS Tools. > Full details are available at: > http://buildbot.networkupstools.org/public/nut/builders/Ubuntu-maverick-x86/builds/39 Looks like the nutscan-snmp.h file has not been checked into the branch: make[3]: Entering directory `/var/lib/buildbot/buildbot-slave-cayenne/

I'm new to ubuntu but i'm starting to understand it. However, Ubuntu is completely useless for me if I can't run the latest version of wine (1.3). Wine 1.2 was working great on 10.04 however photoshop cs4 wouldn't install on it. So I upgraded to maverick, deleted wine 1.2, and am now trying to install wine 1.3. Should be simple, right? I've added the correct repositories to

How can I set the prefered codec between 2 calling parties ?? My Grandstream supports G729, alaw and gsm... in this order. The Zoiper softphone has alaw and gsm as codecs... in that order. Although there should be a matching codec found, my Grandstream can not call the Zoiper softphone. CLI shows : [Mar 11 17:47:21] WARNING[22367]: channel.c:3340 ast_channel_make_compatible: No path to

From: Philip Reames <listmail at philipreames.com<mailto:listmail at philipreames.com>> Date: mardi 8 septembre 2015 12:50 To: Benoit Belley <benoit.belley at autodesk.com<mailto:benoit.belley at autodesk.com>>, "llvm-dev at lists.llvm.org<mailto:llvm-dev at lists.llvm.org>" <llvm-dev at lists.llvm.org<mailto:llvm-dev at lists.llvm.org>> Subject:

Hi all Apparently there is a patch for those 1.2.4/5 MeetMe Freezes: http://bugs.digium.com/view.php?id=5884 Haven't tried it out yet. Benoit Panizzon -- I m p r o W a r e A G - System Services ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz

I am trying to install the maverick version. The lucid version works flawlessly, but this one promts this messages. The following packages have unmet dependencies: r-base: Depends: r-base-core (>= 2.12.0-1maverick0) but it is not going to be installed Depends: r-recommended (= 2.12.0-1maverick0) but it is not going to be installed Recommends: r-base-html but it is not going

Thanks for the hindsight. I am currently working on a patch/potential fix which introduces a new Linker::ImportIntrinsicGlobalVariables flag. The patch includes a unit test reproducing the problem. Hopefully, that will help getting more feedback. Note that it might take a while before I am allowed to upload the patch since I need approval from Autodesk Legal department. Cheers, Benoit Benoit

Hi Richard > That could be possible and would be a bug in chan_sip. Ok, so I switched to PJSIP to see if this behaves differently So ip do a Transfer(PJSIP/${DESTNUMBER}@trunk) And this results in: Failed to parse destination URI '[destnumber scrubber]' for channel PJSIP/trunk-00000011 Do I have to specify the destination number differently when using Transfer with pjsip that I

Thanks Peter, this is very useful feedback. I did manage to change the behavior of LinkOnlyNeeded to correctly import all variables with AppendingLinkage. In fact, I discovered that there was already something fishy. A variable with AppendingLinkage would get imported correctly from the source module if the destination module already contained a definition for that variable and wouldn't be

Hello I want to use a variable in the global section of smb.conf If the variable is %u (or %g), it concerns the current user (root at this moment) and not the veritable user. while ? does somebody can help me ? thanks ------------------------------------------------------------------------------ | Beno?t Marchal | Tel : (33) 03.83.59.55.75 | | E.N.S.E.M.

> > The Buildbot has detected a new failure of Ubuntu-maverick-x86 on > Network UPS Tools. > Full details are available at: > http://buildbot.networkupstools.org/public/nut/builders/Ubuntu-maverick-x86/builds/32 > > Buildbot URL: http://buildbot.networkupstools.org/public/nut/ > > Buildslave for this Build: cayenne Fred, I had a few spare cycles and tried building

In my hunt to track down my echo issues, I tried disabling all echo cancellation, suppression, adaption, on my SPA-2000 (Advanced section of the config, under Line 1/2). Then calling from one local extension to another. (SPA-2000 Line1, to Line2 on the same device) I was pretty shocked with the results, the echo was HORRIBLE! I even tried 3 different analog phones. Now, once I turned the echo