한국어

소프트스위치

온누리070 플레이스토어 다운로드
    acrobits softphone
     온누리 070 카카오 프러스 친구추가온누리 070 카카오 프러스 친구추가친추
     카카오톡 채팅 상담 카카오톡 채팅 상담카톡
    
     라인상담
     라인으로 공유

     페북공유

   ◎위챗 : speedseoul


  
     PAYPAL
     
     PRICE
     

pixel.gif

    before pay call 0088 from app


http://staskobzar.blogspot.kr/2017/01/opensips-command-line-tricks.html


OpenSIPS provides powerful and flexible tool called "opensipsctl". Here I put some useful commands I am using while working with OpenSIPS to: restart phone, change MWI, change presence, remote dial with REFER.


Restart phone


Some SIP phones (Polycoms, Avaya, Aastra and more) support remote configuration reload or reboot when they receive SIP NOTIFY request with special "Event" header. Value of "Event" header depends on phone vendor implementation. Usually it is "check-sync" but some other vendors can have different values. Like Sipura will use "resync", Linksys - "reboot_now".

With "opensipsctl", we can do it as following (it's all in one line):

opensipsctl fifo t_uac_dlg NOTIFY sip:3864@10.0.20.52 . . '"From: <sip:3864@voip.proxy.com>;tag=8755a8d01a12f7e\r\nTo: <sip:3864@voip.proxy.com>\r\nEvent: check-sync\r\n"'

Details on the MI command "t_uac_dlg" can be found on transaction module (TM) OpenSIPS documentation page.

Here "sip:3864@10.0.20.52" is subscriber contact of the phone registered with OpenSIPS and "sip:3864@voip.proxy.com" is subscriber AOR. You can get these data with command:

opensipsctl ul show 3864@voip.proxy.com


MWI light on/off


When SIP user receive new voicemail, then SIP PBX sends NOTIFY request with Event "message-summary" and the body text with details about voicemail messages.

Here opensipsctl command to activate MWI light (it's all in one line):

opensipsctl fifo t_uac_dlg NOTIFY sip:299@voip.proxy.ca . . '"To: <sip:299@voip.proxy.ca>\r\nFrom: <sip:299@voip.proxy.ca>;tag=e17f5bec0b4bfb0bd\r\nEvent: message-summary\r\nContent-Type: application/simple-message-summary\r\nExpires: 3600\r\n"' '"Messages-Waiting: yes\r\nMessage-Account: sip:*97@voip.proxy.ca\r\nVoice-Message: 2/0 (0/0)\r\n"'

This command assumes that we subscriber "299@voip.proxy.ca" is registered with OpenSIPS server and (!) has subscribed to message-summary event. This can be checked with command: "opensipsctl fifo subs_phtable_list". That's why in this command, the request URI (sip:299@voip.proxy.ca) looks like a AOR and not a Contact with IP address. The point is OpenSIPS will check presence subscribers table in memory for subscriber "sip:299@voip.proxy.ca" and event "message-summary", recover the contact IP from the found record, and, finally, will create and send SIP packet to the good destination. The TM module will also take care that all the headers conform subscription session: call-id, from-tag, to-tag, content-length etc.
The header "Message-Waiting" must have value "yes". Another important header is "Voice-Message: 2/0 (0/0)". It means there 2 new messages and 0 old and in parentheses,   (0/0), it's number of urgent new/old messages.

Same is for deactivating MWI light. Just set "Messages-Waiting" to "no" and for "Voice-Message" new messages to 0. For example:

opensipsctl fifo t_uac_dlg NOTIFY sip:235@voip.proxy.ca . . '"To: <sip:299@voip.proxy.ca>\r\nFrom: <sip:299@voip.proxy.ca>;tag=e17f5bec0b4bfb0bac22d\r\nEvent: message-summary\r\nContent-Type: application/simple-message-summary\r\nExpires: 3600\r\n"' '"Messages-Waiting: no\r\nMessage-Account: sip:*97@voip.proxy.ca\r\nVoice-Message: 0/0 (0/0)\r\n"'


Actually, the same can be done with another MI function: "pua_publish" from module "pua_mi". While previous example can be changed to use IP address in R-URI and send "message-summary" notification even to users without voicemail subscriptions, pua_publish will send "notify" message only if there is a subscription exists.

Following command will activate MWI light:

opensipsctl fifo pua_publish sip:7402@voip.proxy.ca 3600 message-summary application/simple-message-summary . . '"Messages-Waiting: yes\r\nMessage-Account: sip:*97@voip.proxy.ca\r\nVoice-Message: 1/0 (0/0)\r\n\r\n"'

And this one to deactivate MWI light:

opensipsctl fifo pua_publish sip:7402@voip.proxy.ca 3600 message-summary application/simple-message-summary . . '"Messages-Waiting: no\r\nMessage-Account: sip:*97@voip.proxy.ca\r\nVoice-Message: 0/0 (0/0)\r\n\r\n"'

The difference is only "Message-Waiting" header, which is set to "no" and "Voice-Message" header.

Refer to dial


Some SIP phones support dial on REFER request behavior. Polycoms, for example, do support this. It is possible to simply send REFER SIP message to phone and  phone would initiate a call to the destination specified in the REFER message.
Here is an example:

opensipsctl fifo t_uac_dlg REFER sip:7329@10.10.0.141 . . '"From: <7329@voip.proxy.ca>;tag=123456789\r\nTo: <sip:7329@voip.proxy.ca>\r\nRefer-To: <sip:5553312244@voip.proxy.ca>\r\nRefer-Sub: false\r\n"'

MI function "t_uac_dlg" will send REFER to the user "7329" with contact IP "10.10.0.141". Notice, that this is initial dialog (no to-tag). When phone receives this message, it will initiat call to "5553312244" (see Refer-To header).

Presence state


Update presence state of the SIP phone is also quite easy with opensipsctl command and pua_publish function. As with an MWI example, phone must be subscribed to watch presence of particular user.

Following example will change presence state of "agent01@bar.voip.ca" to "open":

opensipsctl fifo pua_publish sip:agent01@bar.voip.ca 3600 presence application/pidf+xml . . "<?xml version='1.0'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity='sip:agent01@bar.voip.ca'><tuple id='0x81475a1'><status><basic>open</basic></status></tuple></presence>"

After that message, it is expected that on all the phones that watch user "agent01@bar.voip.ca" the corresponding state indication would be changed. For example, it can be green light on the button or icon change, or some message on soft-phones. The implementation also depends on the phone vendors but usually follows PIDF specifications (for example RFC3863).

조회 수 :
44449
등록일 :
2017.09.13
05:06:37 (*.160.88.18)
엮인글 :
http://webs.co.kr/index.php?document_srl=3311877&act=trackback&key=d87
게시글 주소 :
http://webs.co.kr/index.php?document_srl=3311877
List of Articles
번호 제목 글쓴이 날짜 조회 수sort
168 OpenSIPS Control Panel (OCP) Installation Guide admin 2014-03-06 278093
167 Opensips Gateway between SIP and SMPP messages admin 2019-02-19 265326
166 What is new in 1.8.0 opensip admin 2012-05-21 250819
165 What is new in 2.3.0 opensips admin 2017-09-04 243538
164 Using SIP Devices behind NAT OPensip Asterisk IPPhone SIP Telephony file admin 2013-03-31 223739
163 OpenSIPS vs Asterisk admin 2013-04-06 218210
162 PUSH SERVER 푸시서버 안드로이드 애플 admin 2017-09-11 207193
161 MediaProxy Installation Guide admin 2014-03-06 179413
160 telepresence: Open Source SIP Telepresence/MCU admin 2014-03-12 178008
159 RTPproxy Frequentry Asked Questions (FAQ) ¶ admin 2014-03-07 174453
158 SIP PBX - OpenSIPS and Asterisk configuration admin 2014-03-12 160321
157 Asterisk v1.4x built on FreeBSD v7.1 UNIX admin 2012-01-06 148222
156 사설 망 환경에서 SIP 의 NAT Traversal 문제 admin 2011-12-23 142831
155 How to install Mediaproxy 2.5.2 on CentOS 6 64 bit admin 2017-09-04 142447
154 SIP 트래픽 생성 테스트 툴 admin 2011-12-23 135598
153 opensips command /sbin/opensipsctl detail admin 2017-09-04 123970
152 How to setup a Jabber / XMPP server on Debian 8 (jessie) using ejabberd admin 2017-09-13 123204
151 Opensips_1.9 install guide this is great I like this admin 2014-03-04 106681
150 OpenSIPS basic configuration script 기본 컨피그 admin 2017-09-05 104278
149 Welcome to the Smartvox Knowledgebase admin 2013-04-06 103906
148 Installing RTPproxy Start RTPproxy in Bridged mode very good admin 2014-03-07 101157
147 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅 admin 2014-08-11 100706
146 Flooding Asterisk, Freeswitch and Kamailio with Metasploit admin 2013-04-06 98853
145 OpenSIPS Control Panel install guide admin 2014-03-06 95233
144 dictionary.opensips radius admin 2017-12-09 94044
143 Installation and configuration process record opensips 1.9.1 admin 2014-08-09 93770
142 My new toy: Bluebox-ng admin 2013-04-06 90826
141 in opensips what is lookup(domain [, flags [, aor]]) admin 2017-12-09 90339
140 Conference Support in Kamailio (OpenSER) admin 2014-03-12 84151
139 Kamailo OpenSIPs installation on Debian admin 2014-03-09 81069