asterisk

问题 I've successfully built VoIP server with FreePBX Asterisk. It works fine when I register a user on FreePBX. However, I would like to register a SIP account from mobile device directly. I found out that I can add custom information into FreePBX MySQL database. However, it doesn't work either, and I couldn't find a place to insert SIP password.. Someone said that I need to do something with /var/www/html/admin/functions.inc.php file. Is there better way to create a new SIP extension from

问题 I want to get the duration of the call but confused which variable to use ${CDR(duration)} or ${CDR(billsec)} Here it is not clear from when ${CDR(duration)} records the time So which should i use ${CDR(duration)} or ${CDR(billsec)} ? 回答1: Call come in, after that not answer X second, after that answered, after that Y sec speak/play something and hangup. So duration will be X+Y, while billsec(time to be billed) will be Y. 回答2: BillSec is "how long was the call off hook" ... a common metric

问题 In extconfig.conf they have mentioned that "However, note that using dynamic realtime extensions is not recommended anymore as a best practice; instead, you should consider writing a static dialplan with proper data abstraction via a tool like func_odbc." 1) Why asterisk is not recommending dynamic realtime extensions? 2) How to do static dialplan with data abstraction using tool liek func_odbc? My requirement is having have more extensions (in this case mobile number) coming up, how can I

问题 This is really starting to get annoying. I´m using Asterisk 1.4 since 2007 to operate a flawless PBX, and it creates a SINGLE CDR per call, like any other version of asterisk would. Yesterday I figured an upgrade would be ok and got Asterisk 13.4. This damn thing is creating 2 CDRs per call... one representing the dial attempt .. and another including both the connected call and the initial dial I don't know where to configure the CDR engine to behave normally... that is, to record A SINGLE

问题 I have configured Asterisk 13.13.1 with PJProject 2.5.5 and enable PJSIP as SIP driver (without compiling chan_sip). I have the fully configured system and it's working but I have some problems with incoming calls. I have few numbers connected with my host and when I calling from any public number I noticed this info on asterisk remote console: [Feb 24 14:27:16] NOTICE[5291]: res_pjsip/pjsip_distributor.c:525 log_failed_request: Request 'INVITE' from '"zzzzz" <sip:zzzzz@192.168.34.1>' failed

问题 Unit and integration testing is usually performed as part of a development process, of course. I'm looking for ways to use this methodology in configuration of an existing system, in this case the Asterisk soft PBX. In the case of Asterisk, the configuration file is as much a programming language as anything else, complete with loops, jumps, conditionals, etc., and can get quite complex. Changes to the configuration often suffers from the same problems as changes to a complex software product

问题 I have a problem,if a single user left the confbridge or disconnect his call... I want to hangup calls of all other users who are in that particular conference room...Any idea regarding this??? Basically I want to disconnect all channels if any of the channel hangup the call.Any guidance? Many thanks. 回答1: There are no simple way do that. Reason is simple. Anyway at some moment in conference will be single user(at start) You can use marked user(and close on marked user exist) or you can use

问题 I'm using FreePBX and have this configuration in extensions_custom.conf so that I can receive a notification via Pushover . [macro-dialout-trunk-predial-hook] exten => s,1,System(/usr/bin/sendpush.php "Call from ${CALLERID(num)} to ${OUTNUM}") I also need to receive notifications on incoming calls, but can't figure it out on what context should I apply it. (If it makes any difference, I'm using 4 trunks and want notifications from all of them) 回答1: Solved by just adding: [ext-did-custom]

问题 Can Any please explain why this is happening. what are the possibilities of errors that are been counted as I have set maxerrorcount = 3 EROR OPEN_VXI luke---- callid=[68] |1098905920|68|CRITICAL|com.vocalocity.vxi|216|VXIinterpreterRun: Maximum loop count exceeded. There is probably an infinite loop of in your VXML document.|URL Please let me know if any further details are required. 回答1: Perhaps, "infinite loop" means to call same form again and again, And it was not inserted caller input

问题 It's difficult to tell what is being asked here. This question is ambiguous, vague, incomplete, overly broad, or rhetorical and cannot be reasonably answered in its current form. For help clarifying this question so that it can be reopened, visit the help center. Closed 7 years ago . How can I create a chat-text/VOIP calls application using Android sdk? What are the available apis and sources? 回答1: It is part of the latest Android 2.3 release See http://developer.android.com/sdk/android-2.3