sip

OpenSIPS 2.4 call forbidden

由 ぃ、小莉子 提交于 2019-12-06 13:35:58
问题 I discovered OpenSIPS and all the possibilities a few days ago. I would just use it as a simple SIP proxy to get started. Proxy between my designated UAC and my UAS (asterisk, not natted). The goal is to use a proxy to prevent bot attacks on my UAS. After installing OpenSIPS, I tried to configure my XLITE (natted) by simply adding the proxy URL in the configuration. It works, I register and I can see in my UAS peers my extensions with proxy IP. But when I make a call, I got a forbidden error.

Difference between JAIN API and JAIN SLEE API

由 ☆樱花仙子☆ 提交于 2019-12-06 12:29:47
问题 I was reading about SIP and found that there's an java API JAIN SIP to develop SIP based applications. Then I also found that there is JAIN SLEE and SIP servlets. I believe JSLEE and SIP Servlets are containers to deploy applications based on JAIN and SIP servlets respectively. Can some please explain. Also, please tell me which one is better to start with for learning. Thanks 回答1: JAIN SIP is a java specification (https://jcp.org/en/jsr/detail?id=32) for writing Java SIP applications in a

Kubenetes：服务

由 狂风中的少年 提交于 2019-12-06 10:57:25
1）创建服务 编辑gwp-svc.yaml apiVersion: v1 kind: Service metadata: name: gwp spec: ports: - port: 80 targetPort: 8080 selector: app: gwp 创建服务 kubectl create -f gwp-svc.yaml 查看服务 kubectl get svc 在运行的容器中远程执行命令，其中两个双斜杠（--）代表着kubectl命令项的结束，之后时pod内部需要执行的命令 kubectl exec gwp-fgjg6 -- curl -s http://10.98.20.174 同一个服务可以暴露多个端口，此时需要指定端口名称 编辑gwp-svc.yaml apiVersion: v1 kind: Service metadata: name: gwp spec: ports: - name: http port: 80 targetPort: 8080 - name: https port: 443 targetPort: 8443 selector: app: gwp 此外，targetPort还可以使用命名端口（就是pod中配置的containerPort对应的名称），最大的好处是即使更换端口号也无须更改服务spec。 2）服务发现 通过环境变量发现服务

SipAudioCall without registering a local SipProfile

由 ☆樱花仙子☆ 提交于 2019-12-06 08:48:40
问题 I'm trying to use the native Android SIP stack to make direct SIP calls within a LAN. It appears that within the native stack, you are required to register a local profile in order to make SIP calls. Here is the code I (try to) use to register a profile. I do not have a SIP server on this network, so I just use localhost for a domain. if (mSipManager == null) { mSipManager = SipManager.newInstance(mContext); try { SipProfile.Builder builder = new SipProfile.Builder("foo", "localhost");

SIP ACK Dialog is null

由 你离开我真会死。 提交于 2019-12-06 08:21:52
I am creating a SIP client in java using JAIN SIP. I have managed to register and send INVITE but when sending the ACK back to the server I am getting error which says: Cannot Create ACK - no remote Target I checked the value of Dialog and it is null In the processResponce(), I get the value it is also null this.dialog = responseEvent.getClientTransaction().getDialog(); I pass it to ack(responseEvent.getResponse(), dialog) request =this.dialog.createAck(((CSeqHeader)response.getHeader("CSeq")).getSeqNumber()); dialog.sendAck(request); Also in the Register() and Call() this.dialog = inviteTid

SMS messaging using Twilio with SIP

由 余生长醉 提交于 2019-12-06 07:59:19
I'm trying to use Twilio to send and receive SMS on my iPhone. I've gotten inbound and outbound calling set up following these instructions . I'd like the equivalent instructions for SMS. I'm presently using Zoiper as a SIP client, but that's flexible. The phone number configuration page (from here click a phone number) has a section called "Messaging" which allows one to configure a TwiML bin. However, I haven't been able to find any documentation about how to configure SMS over SIP, let alone TwiML code. 来源： https://stackoverflow.com/questions/52100722/sms-messaging-using-twilio-with-sip

Android audio calls using android's sip

由 孤街浪徒 提交于 2019-12-06 07:14:08
问题 I developed an application for a customer to internet calling using sip.For that he provided me two valid sip user_id and password. Am Using SIP API for SIP implementation.customer says that the call is not going.he don't get any notification about missed calls when he logged using his account.i cant find any mistakes in the code.please help me .the code is given below. public class CallActivity extends Activity { public String sipAddress = null; public SipManager mSipManager = null; public

Asterisk incoming message gives: 415 unsupported media type

由 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:

How to send SIP message using Python sockets

由 泄露秘密 提交于 2019-12-06 06:10:57
I need to send SIP message using Python sockets, I've made that client sends something to server but I'm not able to make that client sends a SIP message INVITE to server #!/usr/bin/python import socket R_IP = '192.168.2.1' R_PORT = 5060 message = 'INVITE sip:user1110000000350@.com SIP/2.0 To: <sip:user4110000000350@whatever.com>\x0d\x0aFrom: sip:user9990000000000@rider.com;tag=R400_BAD_REQUEST;taag=4488.1908442942.0\x0d\x0aP-Served-User: sip:user4110000000350@whatever.com\x0d\x0aCall-ID: 00000000-00001188-71C0873E-0@10.44.40.47\x0d\x0aCSeq: 1 INVITE\x0d\x0aContact: sip:user9990000000000@rider

SipManager.newInstance returns null

由 好久不见. 提交于 2019-12-06 05:33:48
问题 I implement simple SIP client app for receiving calls. I go through official manual and get code from it. I noticed that SipManager.newInstance(getApplicationContext()); returns null. Documentation says that it happens when SIP API is not supported by device. However I use LG G6 with Android 7.0 and I successfully test third-party SIP clients from Google Play. So I doubt that API is not supported really. How could I check that? My manifest has all permissions ( INTERNET and USE_SIP )
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