asterisk

Vicidial SIP Trunk with Twilio

て烟熏妆下的殇ゞ 提交于 2019-12-06 14:46:22
问题 I need a step by step guide on configuring Twilio Elastic SIP Trunk on my Vicidial Server. I've been working it out for days now. Still can't make an outbound call. My account on twilio is still a trial account. thank you guys. :( 回答1: From vicidial admin panel, go to Admin >> Carriers Add a new carrier named "myname" **Replace "myname" with whatever you like but keep it consistent throughout the config. Anywhere you see "myname" replace it with the same value. In the account entry section

Having the mandatory symbol to the edit text (red color asterisk) which is inside the textinputlayout

半城伤御伤魂 提交于 2019-12-06 09:53:36
问题 I have an edit text inside a textinputlayout. I am trying to set the mandatory field symbol (red color asterisk) to the edittext. I have set the hint to the edit text. My code is as below. <LinearLayout android:layout_width="match_parent" android:layout_height="match_parent" android:layout_marginLeft="16dp" android:layout_marginTop="16dp" android:orientation="horizontal"> <TextView android:layout_width="wrap_content" android:layout_height="wrap_content" android:text=" * " android:layout

Why asterisk not properly working with android sip client?

为君一笑 提交于 2019-12-06 07:18:27
问题 Asterisk= 1.8.11.0 Android= 2.3/4.0.3 Android Sip client=Native Android sip client/sipdemo When i call from my pc using zoiper/xlite to android (native android sip client) now i can hear audio from both sides but when i make call from android to pc (zoiper/xlite) i cannot hear anything on android. On the other hand i have tested this scenario on elastix (which also uses asterisk 1.8.11.0) with no problem in audio. pc(zoiper) ip 192.168.15.27 android ip 192.168.15.71 asterisk server ip 192.168

Asterisk incoming message gives: 415 unsupported media type

亡梦爱人 提交于 2019-12-06 07:13:55
问题 Me and my project group are trying to set up a PBX with asterisk. We've managed to let it work with just SIP calls and that works perfect. But once we want to try add an XML message to it Asterisk doesn't recognize it and gives "415 Unsupported Media Type". It seems like the call isn't even making it through it is getting rejected immediately. We have tried to find the piece of code where this gets handled but didn't found anything. The SIP message that is send to Asterisk looks like this:

Asterisk AGI: How to get or set the value of a global variable?

不羁的心 提交于 2019-12-06 06:16:01
I'm using Asterisk 1.8 with PHP for AGI scripting. EDIT: I'm struggling with setting and obtaining the values of global variables from within an AGI PHP script. I can set channel variables but not global variables. Using PHPAGI lib. Tried: Set({$varname}={$value},g) Set({$varname}=\"{$value}\",g) Set(GLOBAL({$varname})={$value}) That does not seem to work at all, when getting the value from within the dial plan, it is empty. Does anyone have a working example of setting and getting global variables in an AGI script? I found a workaround to make it work. First, the global variable must not be

Asterisk playback sound file into ConfBridge?

荒凉一梦 提交于 2019-12-06 02:48:39
I'm trying to make a conference and play a sound file in the background of the conference. How can I make this possible? this obviously wouldn't work because the sound file will be played before entering the conference. exten => s,1,playback(some/soundfile) same => n,confbridge(1) Thanks in advance! Wanted to add my solution here in case anyone ever needs it. first make a context for the conf bridge in extensions.conf: [conf-msg] exten => s,1,ConfBridge(01) where 01 is the bridge number Then via the command line you can do: asterisk -x 'channel originate local/s@conf-msg application Playback

Asterisk,SIP Retransmission timeout

左心房为你撑大大i 提交于 2019-12-05 19:43:24
问题 I have created a sip trunk from One Asterisk(version 11.2.1) say 'A' server to another Asterisk server(11.7.0) say 'B', and I am getting sip response 200 ok. But when I start calling on a DID on Asterisk A then the call is being routed to Asterisk 'B' and After 38 seconds call has been disconnected showing following warnings : Retransmission timeout reached on transmission 11bc71e029119e5877806ed40fcde691@111.xxx.xxx.xxx:5060 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org

Make phone ring through webserver using Asterisk

天涯浪子 提交于 2019-12-05 18:36:14
What is the easiest way? And how can it be implemented? I thought to something like: I request a page to the webserver Php sends to asterisk a number to dial (php and asterisk are running in the same machine) Asterisk calls the phone or A php sip client registers to remote asterisk server Php sip client sends to asterik the number to dial Phone rings i don't care if i can hear something, it's enough that it rings Call File The easiest way is to generate a Call File. Asterisk has the feature to initiate Calls through a spool directory. https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call

saving data into custom cdr field

自古美人都是妖i 提交于 2019-12-05 16:13:06
I created custom field "rec_name" id table "cdr", database "asteriskcdrdb". In this field I want to store recording name. I know I should do it by adding this line in one of .conf files, but where? exten => s,1,set(CDR(rec_name)=${CALLFILENAME}) I can do it by sql statement, too, but I don't know where is the file that saves details of call into database. For mysql you need add into /etc/asterisk/cdr_mysql.conf [aliases] rec_name=rec_name If you HAVE aliases section, just add to it rec_name=rec_name 来源: https://stackoverflow.com/questions/10499261/saving-data-into-custom-cdr-field

How to get all dialer events from Asterisk REST API (ARI)?

烂漫一生 提交于 2019-12-05 12:28:39
I'm making a web application which should be able to monitor calls on my Asterisk server. I can connect to ARI with Javascript WebSocket on URL ws://(host):8088/ari/events?app=dialer and it works. The problem is that I only get events from calls that are made over ARI. Calls made from other clients like Zoiper are not registered. On the other hand, Asterisk has AJAM which uses long polling on http://(host):8088/rawman?action=waitevent and it registers calls from all the clients, (ARI, Zoiper and others) but there's only information who is calling (caller), not whom (callee). So, my question is