sip

问题 I am wondering how you can get the call id when you make an outgoing call using the android.net.sip API. I am currently just making an outgoing call as they do in the android sip demo. call = manager.makeAudioCall(me.getUriString(), sipAddress, listener, 30); I also saw in the documentation that you can create a sip session when making a call in order to get the call id, but I just can't figure it out. See http://developer.android.com/reference/android/net/sip/SipManager.html#createSipSession

问题 I want to implement the SIP protocol in java and would want to be able to create different clients (5 or more) and make them connect to a proxy server. This is all for testing purposes so I would like to be able to see well what's happening on a rather low level. The clients should first be able to communicate trough text and later on maybe also by audio. (If I ever get that far) I already read a bit about the JAIN libraries and what I understood from that is that they are not really well

问题 I'm trying to compile this lib: https://github.com/BelledonneCommunications/linphone-iphone During generation I get this: /linphone-iphone/submodules/build/..//externals/speex/libspeex/cb_search.c libtool: Version mismatch error. This is libtool 2.4.6, but the libtool: definition of this LT_INIT comes from libtool 2.4.2. libtool: You should recreate aclocal.m4 with macros from libtool 2.4.6 libtool: and run autoconf again. make[4]: *** [cb_search.lo] Error 63 make[3]: *** [all-recursive]

问题 Are there any code example for this? I want to know what should be the flow in processRequest when an INVITE request is received. I'm particularly interested in how you forward a request from an UA to another previously registered UA. 回答1: Check out https://svn.java.net/svn/jsip~svn/trunk/src/test/unit/gov/nist/javax/sip/stack/WebsocketSelfTest.java the WebsocketServer class should be enough to get you started to write a simple client-server. Forwarding just means you act as client repeating

问题 We have a server "Elastix". Is there a way to make something like this? If someone calls on fax machine, write this behavior in database and hangup immediately. I heard about AMD application and read a lot of information, but still don't quite understand, how to make it do what I need. Can you help me with solution, please? Thank you in advance! 来源： https://stackoverflow.com/questions/38570123/how-can-we-handle-outgoing-fax-calls

问题 Let's say I need to implement a callback service on a website. I'm curious if it's possible to develop it using Twilio. For example, if a website visitor provides his mobile phone number from one side, from another side I have a manager using his web CRM application. Is it possible to: Make a call (using WebRTC?) to a manager When the manager replies the incoming call, call to the website visitor (on his mobile phone) Connect both sides together I know there are lots of ready-to-use callback

问题 What I'm actually looking for are the termination and origination SIP URI's for a given trunk. The closest I found so far was: getCredential public Credential getCredential(String credentialSid) Gets the credentials from the credential list Returns: the credentials https://twilio.github.io/twilio-java/com/twilio/sdk/resource/instance/sip/CredentialListInstance.html#getCredential-java.lang.String- How do I get a credentialSid , and, what is a credentialSid ? SID is Security IDentifier? see

问题 I want to implement Notify event CallBack in CsipSimple here is the C code i have written My C/C++ files The pjsip_event_notification.h file #include <pjsip-simple/evsub.h> class EventCallBack { public: virtual ~EventCallBack() {} virtual void on_evsub_state(pjsip_evsub *sub, pjsip_event *event){} virtual void on_tsx_state(pjsip_evsub *sub, pjsip_transaction *tsx, pjsip_event *event){} virtual void on_rx_refresh(pjsip_evsub *sub, pjsip_rx_data *rdata, int p_st_code, pj_str_t **p_st_text,

问题 I'm sending DTMF to Asterisk using python and pyst. The code is: def playDTMF(self, channel, digit): print "DTMF: Sending %s to %s" % (digit, channel) response = self.manager.send_action({ "Action" : "PlayDTMF", "Channel" : channel, "Digit" : digit }) print "DTMF: %s - %s" % (response['Response'], response['Message']) Output of the script is good: DTMF: Sending 1 to SIP/S3bc9c3c-00000081 DTMF: Success - DTMF successfully queued But Asterisk reacts to it like that: [Jul 2 17:48:53] DEBUG[6967]

问题 I can register from my webclient to my freeswitch. But, when I try to make call the call gets rejected with 488 not acceptable here. From freeswitch console log im getting. 2014-07-22 22:03:59.673585 [DEBUG] switch_core_state_machine.c:53 sofia/internal/alice@192.168.146.133 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION I added < action application="export" data="rtp_secure_media=true" /> with my extension; but no luck. below is the SDP of my INVITE v=0 o=Mozilla-SIPUA-31.0 26508 1 IN