similar to: AEL NoOp not working

Hi, Do you have any example showing how to use SendFAX ? I can see several examples of ReceiveFAX but not a single one showing SendFAX. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081127/b41ca08b/attachment.htm

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

How do I include a file (not a context) in AEL? #include "filename" returns an error. Thanks, Jay

Hi, I'm using asterisk 1.6.1 and AEL2. I'm trying to find the best way to write my own custom functions ? At the moment, I'm using this pattern (extensions.ael) : context foo { 123 => { &myfunc(123456); NoOp(${GOSUB_RETVAL}); }; macro myfunc (arg) { Return (${arg}); } 1. First, I keep getting warnings like Warning: file /etc/asterisk/extensions.ael, line

I am building a new Asterisk server here at the office, and I'm wondering if there are any downsides to creating my dialplan with AEL. It seems more intuitive (to me), but I'm not sure if there are any pitfalls I need to be aware of first. We use this for internal extensions, 8 pots lines, and our answering service which gets about 500 incoming calls a day down our T1. Also, one more

When the asterisk a queue reset their counters? I 'm talking about this kind of info in asterisk console. >show queue 600 600 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime), W:0, C:14, A:8, SL:0.0% within 0s I just say that because I have a queue with strategy "Fewest Calls" working for a couple of mouths, and a new agent has been added this

Hi i have problem with AddQueueMember logic. I need login Agent(Member) in asterisk. use this option: for example: AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13) and now i want to call to this Agent: exten => _1XX,1,Dial(Agent/${EXTEN:1}) call to 113 and asterisk should call to Agent => 13 on interface SIP/ekiga. This doesn't work, How can i do this on Asterisk 1.4(not

when i wnat to working with realtime and mysql for any context i have to insert (switch => Realtiem/context at extensions) statment into extensions.conf for example if i want to have 10 context, i have to insert these lines into extension.conf : [context1] switch => Realtiem/context1 at extensions [context2] switch => Realtiem/context2 at extensions [context3] switch =>

Hi, VERBOSE[6120] logger.c: -- Got SIP response 500 "Server Internal Error" I just noticed that i sometimes get those back from provider. They are currently general SIP message log entries with verbose level 3. I wonder if such SIP fails could generate at least WARNING in log? Currently i'm checking logs for warnings and errors, so i probably have missed those.. It would be

Hi guys I've just read this about the upcoming release of * 1.6: ?Better reporting through a new call event logging capability in Asterisk 1.6 will allow complete tracking of events that take place during a call. The goal, according to Fleming, is to provide more detail than traditional CDR (Call Detail Recording) features offer and to allow for more granular tracking and auditing.? That

Hi, Here you can find a list of MS Exchange 2007 compliant systems: http://www.microsoft.com/technet/prodtechnol/exchange/telephony-advisor.mspx You cannot see mention of Asterisk ;-)) Has anyone tried to "integrate" Asterisk and Exchange 2007 ? A prospective customer is using MS Exchange 2007 and is asking for such integration. Passion aside, what do you think of such integration ?

Hi, I have an simple queue and agents defines with memeber => SIP/123. If for example Agent "SIP/123" has an call, the queue didnt care and tries to send additional calls to this agents. So Iam loosing time. SIP/123 (In use) has taken no calls yet How to stop this, especially when the device is not able to send an BUSY back. Use LOCAL channels and parse 'show queues' or

Hi All; I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile: Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server? Regards Bilal

Hello, I am trying to alter the outbound callerID for extensions within a context I have created. I wrote the following: exten => _9.,2,ExecIf($[$["${REALCALLERIDNUM}" = "360"] | $["$ {REALCALLERIDNUM}" = "670"]]|Set|CALLERID(num)=581560) exten => _9.,3,ExecIf($[$["${REALCALLERIDNUM}" = "361"] | $["$

Having a weird issue with some agents getting stuck busy on my system. Call will come into the queue and the agent will hit DND, or be DND when the call comes in (DND being the button on eyeBeam softphone, not a star code). After the agent comes back from DND they will be "stuck" as busy in the queue and I have to reload chan_agent.so in order to get them available. I'm running

I create my sip users using a common macro in 1.4: [internal] exten => 200,1,Macro(phones|200|SIP/200) [macro-phones] exten => s,1,Dial(${ARG2}|45|Tt) etc... But now in 1.6 this fails: -- Executing [200 at handsets:1] Macro("SIP/201-0942b530", "phones|200|SIP/200") in new stack [Nov 20 08:55:55] WARNING[5958]: app_macro.c:201 _macro_exec: No such context

Hi all! I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers and tools but my calls aren't logged. I'd enabled mysql log and noticed that asterisk send a 'DESC cdr' so connection is working between asterisk and mysql and I am able to call other phones so Asterisk is working as well. No error messages on startup though. Any idea why is it happen? As I realized

Hi, i'm new in asterisk programming. Maybe my question was posted thousand times but i found nothing using google. I'm looking for a method to limit the total simultaneous calls (inbound and outbound) that pass from internal phones to 2 SIP providers. I found the calllimit option but it works only on a per-channel basis. Instead i want limit the total amount of calls, abstracting from

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 all, I received a report of a client which stated that two of its agents are ?logging in? to the queues when they actually aren?t there working. They appeared to be logged on all night. They thought they weren?t logging off correctly, but they checked one of them and he was following the procedure. Any ideas of what can be happening? Is there a way to prevent logins to queues afterhours?