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问题 I am running a B2C outbound Campaign on VicidialNow C.E 1.1 as Asterisk Server / SIP Server . The call is made from server to customer and connected to agents waiting for calls. The agents transfers the call to third party (not a blind transfer). The 3rd party sees the Caller ID of agent. Now, what I want is to display the caller id or the phone number of the customer to the 3rd party. I Googled and searched over SO, found this sendrpid=pai to add on sip.conf file. but this functionality only

问题 My company is currently using Grasshopper as our voice system for receiving and routing inbound customer support calls. It's working pretty well. Except that it's 2014 and people expect to be able to text issues to our toll free customer support number and get responses. Grasshopper doesn't support receiving/forwarding SMS. So I want to use Twilio just for receiving inbound text and Grasshopper for voice on the same number . It seems like a number has to be registered either with Twilio or

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 I want to pickup call in Asterisk using AMI. I can originate call, but totally don't know, how to answer the phone... Script for calling: #login sock = socket.socket(af, socktype, proto) sock.connect(sockaddr) sock.send('Action: login\r\n') sock.send('Events: off\r\n') sock.send('Username: '+str(ast_server.login)+'\r\n') sock.send('Secret: '+str(ast_server.password)+'\r\n\r\n') #originate call sock.send('Action: originate\r\n') sock.send('Channel: ' + str(user.asterisk_chan_type) + '/' + str

问题 I’m using FreePBX with Asterisk’s Java API. For the moment, I’m able to display all my SIP peers with their respective states: public void onManagerEvent(ManagerEvent event) { // Look if the event is a IP phone (Peer entry) if(event instanceof PeerEntryEvent) { PeerEntryEvent ev = (PeerEntryEvent)event; // Get the user extension peer = ev.getObjectName(); // Add to the array peersName.add(peer); } } I’m able to display the phone number and name of both callers when a channel is open: private

问题 Me and my team were tasked to integrate our application with Alcatel Genesys call center, but we don't have access to a proper instalation nor equipment (like, for instance, phones). Is there some kind of software I can use to simulate such environment to test our application? And where should I begin researching how to do this integration? (PS: I posted this same question on https://serverfault.com/questions/308381 - I didn't exactly know which of the sites this really belongs to). 回答1:

问题 I m planning to create a centrex system app in android. In which there is a voice mail will set in the receiver side. By the instructions in the voice mail, caller has to press the number in the dial pad, receiver side should identify the number and do the corresponding actions. Does anyone have any idea about how to detect the number pressed by the caller in receiver side?, What is the technology behind it?, Is it possible in android? Any help will be appreciable.. 回答1: Decoding DTMF is

问题 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

问题 I want to get Call Details from Genesys Platform SIP Server. And Genesys Platform has Platform SDK for .NET . Anybod has a SIMPLE sample code which shows how to get call details using Platform SDK for .NET [ C# ] from SIP Server? Extra Notes: Call Details : especially i wanted to get AgentId for a given call and From Sip Server : I am not sure if Sip Server is the best candiate to take call details. So open to other suggestions/ alternatives 回答1: You can build a class that monitor DN actions.

问题 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. 回答1: 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