asterisk

Asterisk AGI - Originate a call using php agi

谁都会走 提交于 2019-12-05 10:59:54
Is anybody knows , how we can Originate an external number call using PHP AGI script ? MichelV69 You have got two possible options. One is use the "Originate" command. See http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+Originate for an example. The other one, which is what I favor in my solutions, since it does not require AMI, is using spooled call files. See https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files for how to do them. The trick, and I cannot stress it enough, is to create the spool file in /tmp and then "move" the file into the ../spool/asterisk/outgoing

Asterisk - Pre-emption calls

放肆的年华 提交于 2019-12-05 08:08:54
问题 I would like to have pre-emption calls in Asterisk. With this I mean that if user A has priority/access-level 1 and wants to talk to user B, how could it preempt the call that user B is already having with user C which has only priority/access-level 2? Does anyone know if this is supported by Asterisk or how this could be implemented? Any idea would be very welcome. 回答1: No, it not supported by asterisk. But yes, it can be implemented using dialplan+some script magic. Complexity is high and

SIP, asterisk, adhearson and VoIP

扶醉桌前 提交于 2019-12-05 08:07:21
问题 I'm trying to create a VoIP based IVR service that interacts with a web application. From what I understand, adhearson runs on top of asterisk. What else do I need to have on the server to satisfy the equation? I think I need a way for asterisk to connect to a voip account. I'd appreciate any help and/or phrases to google. 回答1: If you want to build your IVR quickly and easily you'd be better off looking at something like tropo.com (tropo's parent, voxeo, own adhearsion) or twilio.com, they've

Asterisk: create user with template via AMI

别来无恙 提交于 2019-12-05 02:06:30
问题 I need to modify sip.conf with AMI, adding a new user to it. Everything works fine, and I can create a user like this without problems: [1000] secret=pass12 But I have to create user with template like [1000](mytemp) secret=pass12 and I don't know how to do this. Neither Google, nor Digium forum can't help me. P.S. I use JavaScript asterisk-manager to interact with Asterisk, and here is my code, which adds extension: var amiAction = { action: 'UpdateConfig', reload: 'yes', srcfilename: 'sip

getting the group name to the according pri port in asterisk

匆匆过客 提交于 2019-12-05 01:39:07
I am using sagoma 8 port card My chan_dahdi.conf to configure the ports are ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit ;autogenrated on 2015-06-12 ;Dahdi Channels Configurations ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no

Open source easy-to-customize call center solution?

假如想象 提交于 2019-12-04 22:06:46
Note: A have asked this question at the Superuser community first , but since it involves a lot of software-building-related topics I decided to move it here. I need to provide a simple call-center solution to a small local business. They have a local 6-digit land line number. They accept calls locally, i.e. nobody calls them from another cities. The problem is that they started to lose customers, because with a certain amount of calls every day it has become impossible to reach the local phone number (it is always busy). So, there is a call center solution needed. I assume that I can somehow

Vicidial SIP Trunk with Twilio

寵の児 提交于 2019-12-04 19:12:51
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. :( 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 use this template: Account Entry: [myname] type=peer secret=mypassword ;if you created a Credentials list in

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

落花浮王杯 提交于 2019-12-04 15:23:01
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_gravity="center_vertical" android:textColor="#ff0000" android:textSize="15sp" /> <android.support.design

Asterisk incoming message gives: 415 unsupported media type

二次信任 提交于 2019-12-04 14:47:07
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: Request-Line: MESSAGE sip:701@xxx.xxx.xxx.109 SIP/2.0 Method: MESSAGE Request-URI: sip:701@xxx.xxx.xxx

Adding chat and VOIP calls functionality? [closed]

自作多情 提交于 2019-12-04 14:15:25
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? jeand It is part of the latest Android 2.3 release See http://developer.android.com/sdk/android-2.3.html http://developer.android.com/resources/samples/SipDemo/index.html If you wish to develop for