similar to: AEL Macro are evil :-)

Hello, I have a Google Voice phone number and want to connect it to my asterisk box to have calls handled to my SIP account. When I call the number I receive the correct INCOMING request on Jabber portion of asterisk, but the call is not connected to the gtalk part. JABBER: asterisk INCOMING: <iq from="+ 17174695631 at voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" to="

I am banging my head over a simple asterisk trick I was seeing on one asterisk server. An extension dials an international premium number, the called number answers, then the extension hangups, but the call continue to run on the international number side, generating an high profit for the premium number company and a big loss for the asterisk owner. I think some sort of "transfer"

What the minimum Server Specifications do I need to run 200 concurrent channels at the time with .WAV recording (MixMonitor)? It will be connected via VOIP sip account. Codec will be ulaw. Which UK dedicated server provider do you recommend and how much bandwidth do I need? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, almost any useful CLI command has an analogue on Asterisk Manager Interface, but I cannot find a way to get the list of subscriptions using AMI. Which is the command, if any? The CLI command is "sip show subscriptions" Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have a recording started in the dialplan with the MixMonitor application. I want to be able to stop it during a call and maybe restart it. I tried using the value defined in [featuremap] but it starts another MixMonitor application even if there already one instead of stopping it. Any idea on how I can stop the MixMonitor application while it is running? [featuremap] automixmon =>

I'd like to dial two extensions (or external number) and ask for confirmation to accept the call. Dialing an extension, asking for confirmation and then dialing a second extension if the call has not been accepted is easy by using the dial option U(...), but if I dial two extensions at once, when the first answers, the other stops ringing. Any idea to make the first continue to ring until

I'm trying to clarify my understand of gosub, macros and AEL. My understanding is that macros using the Macro() application, which is defined in extensions.conf by: [macro-foo] ... and called in extensions.conf with exten => _9NXXNXXXXXX.,n,Macro(fastbusy) is deprecated in favour of Gosub(). True so far? But then there are "macro"s defined in extensions.ael: macro foo() {

Hello, I am moving from realtime chan_sip to pjsip and one of the problem I am facing is the lack of "sipregs". With chan_sip, when an extension registers, the server where it has registered to is stored in sipregs. Is there something similar in pjsip? How can I find on which server the pjsip extension has registered to? Leandro -------------- next part -------------- An HTML

Hello, I have an asterisk box with a peer configured with nat=force_rport,comedia, but asterisk keeps sending the audio to the private IP address and ignoring the client peer nat settings. If I check the "sip show peer extension", I see both symmetric RTP and Force Rport are set to yes, but asterisk seems ignoring them. Force rport : Yes Symmetric RTP: Yes Asterisk is behind a

>>> Nobody has any information or opinions on any of this? Personally, I don't think MACROS are going anywhere any time soon, so I have not bothered looking into a substitution. As for ael; I've never used it. Doug

Hi, according to discussion on asterisk IRC, where people said, that macros will be depracated, I tried to migrate from macros to contexts and Gosub but if I try to use gosub in extensions.ael, ael compiler complains, that I shouln't use Gosub app, but I can't find ael keyword, that will be Gosub equivalent, or can I ignore this ael warnings? thanks PJ LOG: lev:3 file:pval.c

Hi list someone is trying to hack my server . Is there any way by whcih I can stop hacking of my server except iptables ? I want to stop on the basis of sip.conf account only. bcoz I can't apply iptables rules on server it's remote server of server provider and we used it for making voip call for customers. for the time been i have close all sip accounts. but can't stop for more then

Almost two years ago, a change between how AEL code is built into Asterisk dialplan between minor versions made clear the need to provide a sane entry point into AEL subroutines and that's how AELSub() born. With Asterisk 11 release, they way [stdexten] at extensions.conf is invoked changed from Macro to Gosub using the 'missing context feature' and this caused that any stdexten

Hello, I was wondering if there is a tool that can create the realtime database structure for latest Asterisk version or a web resource/file containing the sql scripts. Hope I haven't missed obvious things, I had no luck searching on the web, in the wiki I found few pages with bits of sql or table structures, like:

Hi, On 64 bit centos 5.6 I have virtualbox 4 and 64 bit elastix latest. Since yesterday cpu utilization has been constantly peaking 65-75%. Hardly 1-2 concurrent calls. No other activity on server. Top shows asterisk on top. Its quad xeon server with 4 gb ram. Any suggestion where should I start and how should I go about with my investigation. Thank you and have a great weekend. Sans

Hi, all. I'm making my first foray into AEL. I'm starting with a simple macro, but I've already encountered an odd behaviour. I'm wondering if someone can shed some insight: Asterisk 1.2.9.1 macro newPlaceCallPSTN { s => { TIMEOUT(absolute)=7200; NoOp(Analog Out List: ${ANALOGOUT}); ChanIsAvail(${ANALOGOUT}); NoOp(Available Out List:

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 I came across couple of pointers on the Internet regarding solutions available for providing hosted PBX service. 1. Multiple PBXs: Using separate hardware to host each PBX. Pretty straightforward, but no hosting company wants to use it. 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. partitioning a single instance of Asterisk into multiple PBXs by way

Hello, I was trying to use a queue in linear order and to provide the exact order of members to dial by adjusting the uniqueid value. Obviously it doesn't work and it seems an old problem: https://issues.asterisk.org/jira/browse/ASTERISK-18480 Realtime configuration can't identify "orders" in the list of results, so the members for the queue are returned in random order.

Hello friends, when a call arrives in the queue, a CDR record is created, but there is no info about which agent has picked up the call. I can find that info only in queue_log. Is there a way to have that info in the CDR or maybe in a variable in the "h" context, when the call is ended? Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL: