similar to: Important security alert: update your dialplans now!

> > > > 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

I've built chromium with " -fprofile-instr-generate -fsanitize=address" -- the performance looks good! The file format from r198638 is indeed rudimentary. Do you already know how the real output format will look like? Just to summarize what I think is important: - minimal size on disk, minimal amount of files - minimal i/o while writing to disk, no lockf or some such -

Hi, I am using Asterisk 13.6.0 and was wondering if I can programmatically add users (to pjsip.conf) and dialplan (to extensions.conf) to the Asterisk server using API of some sort. Please do let me know. Thanks, Sonny. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi. If I use a Cisco as a PSTN termination GW and need to route all incoming isdn calls to my asterisk and all outgoing calls from asterisk via the cisco out to pstn, how do I do that ? in the cisco I have this: dial-peer voice 1 pots destination-pattern [0-9]T no digit-strip direct-inward-dial port 3/0:D ! dial-peer voice 50 voip destination-pattern [0-9] voice-class codec 1 session

Finally I decided to rewrite my dialplans according to the right sintax, that is exten => someexten,priority,application(arg1,arg2,...) should be exten => someexten,priority,application,arg1|arg2... Isn't there anybody skilled enough in regular expressions that could write a quick Search 'n' Replace vi command, please? TIA, Alex

I have an Asterisk system setup for our small business, and its working well. I posted to the list about a week or so ago, regarding having it handle direct extension dialing, and unfortunately I'm not any closer to solving this issue, so I was hoping someone might have a working example of how to set this up they could point me towards. Basically I have everything EXCEPT direct

I have a working dialplan for our phone system with Mon-Fri, business hours identification, etc. But what I'm lacking right now is support for company holiday dates. What I'd like to do is to create a database of these dates and just update them as new years rollover. I suspect others have done this sort of thing with Asterisk before, but I've not found any resources so far.

I have some Cisco 7940's running SIP image 6.3 and a newphone account. Reguarding my dialplan I'm having a small issue. I'd like to dial 9,2,xxx-xxx-xxxx for a LD Nufone calls - however I also need to dial local phone numbers ie 9,2xx-xxxx Currently my dialplan looks like so <TEMPLATE MATCH="9,1.........." Timeout="0" User="Phone"/>

This past holiday weekend has resulted in some real groaners when it comes to bugs in our dialplan, making obvious the need for some changes in our procedures. First, our hours of operation for Christmas Eve, Christmas, Boxing Day and New Year's Eve had changed with little to no notice. Okay, fine, whatever, I fix. Our Christmas Eve hours (made worse by being Monday this year) dialplan

Does anyone have experience with ATA-186 Dialplans? If I was doing this in Asterisk I would use the following patterns: 911 _9NXXXXXX _91XXXXXXXXXX _NXXX _*XX I tried the following in my ATA-186, but it doesn't seem to work: 911S|9.r6S|91.r9S|^1.r2S|*..S Does any one have any ideas/suggestions? Thanks in advance, --Eric

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, When reloading a diaplan (asterisk 1.6.1.X), I can see in console : [Apr 2 09:02:00] WARNING[2217]: ael/pval.c:2522 check_pval_item: Warning: file /etc/asterisk/extensions.ael, line 621-621: application call to Gosub affects flow of control, and needs to be re-written using AEL if, while, goto, etc. keywords instead! What is then the recommended substitution for Gosub() application

Hi, I have read the docs, and now I want to attempt to setup Asterisk 1.6. I am not going to complicate it with load balancing, etc. The setup is just 1 SIP line - no other in-house connections. All inbound traffic. I intend to keep this simple. Imagine that I sell pies in my neighborhood.. I want to make it easy to order them. So I create a dialplan to receive a call, record the caller

Hi, How do you handle the case where a module returns -1 ? eg consider this: exten => 123,1,Answer exten => 123,n,Playback(some-message) exten => 123,n,etc ... exten => 123,n,etc ... exten => 123,n,etc ... exten => 123,n,Command(${SOME_PARAMETER}) Now what if command returns -1 here ? I would like to branch accordingly. Also how do you handle jumping to

On 2007-05-15 Tony Mountifield wrote (wrt using AGI scripts to dial out): > Can't comment on this one, as I never use AGI to dial. > My AGIs just set the context, extension and priority, > and exit to the dialplan to do any dialling. (http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/185537) I would like to do this, but I am having trouble figuring out how. I have

There is a problem that bothered me for a long time: Since one of the 1.6.0.x patch releases up until 1.6.2.5 a "dialplan reload" works only once with a bigger dialplan. If I issue "dialplan reload" again, it won't do anything. After doing so the cli won't show responses to any commands anymore. So if I have to do another change to the dialplan, I have to stop/start

I tried putting this as the ata's dailplan: *St4-|#St4-|9|^9t4>$.- this is sip.conf [ata2001] type=friend username=ata2001 secret=SoMeSeCrEt host=dynamic context=fromata canreinvite=no and this in extensions.conf [fromata] ignorepat => 9 exten => _91700NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel) exten =>

Hey, how can I implement a dial plan like the following: incoming call: 1. ring phones 1,2,3 monday to friday between 9:00 and 20:00; if no answer after 15 sec also ring phones 4 and 5 2. ring phone 1 monday to friday between 0:00-9:00 and 20:00-24:00; if no answer after 20 sec also ring phones 2 and 3 3. ring phone 1 saturday and sunday all day I do not need a in detail answer for each of the

Does anyone know the maximum number of digits for an international phone number? Doing some searching, it looks like 16 numbers including the "011" is the maximum number, because 17 is just not found: OK: 1234567890123456 http://www.google.com/search?q=011XXXXXXXXXXXXX Not OK: 12345678901234567

I am working on a system where I connect to an external API and based on what it gives me I generate the Asterisk dial plan accordingly. I am thinking about my different options and wanted feedback from others on how to best do it. 1) Generate conf files for Asterisk - This seems the easiest but then I will be doing a dial plan reload on all of my dial plan for handful of lines of code. The plus