Question

我在服务器上设置了Asterisk。我设置了基本的“Hello，World”示例（直接来自文档here），其中包含我的extensions.conf文件：

[from-internal]
exten = 100,1,Answer()
same = n,Wait(1)
same = n,Playback(hello-world)
same = n,Hangup()

和我的pjsip.conf文件：

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0

[6001]
type=endpoint
context=from-internal
disallow=all
allow=ulaw
auth=6001
aors=6001

[6001]
type=auth
auth_type=userpass
password=unsecurepassword
username=6001

[6001]
type=aor
max_contacts=1

我有一个简单的Python脚本来调用扩展程序100.这是直接来自文档here，我在这里调用此脚本传入sip地址sip:100@<my-sip-server-ip>：

import sys
import pjsua as pj

# Logging callback
def log_cb(level, str, len):
    print str,

# Callback to receive events from Call
class MyCallCallback(pj.CallCallback):
    def __init__(self, call=None):
        pj.CallCallback.__init__(self, call)

    # Notification when call state has changed
    def on_state(self):
        print "Call is ", self.call.info().state_text,
        print "last code =", self.call.info().last_code, 
        print "(" + self.call.info().last_reason + ")"

    # Notification when call's media state has changed.
    def on_media_state(self):
        global lib
        if self.call.info().media_state == pj.MediaState.ACTIVE:
            # Connect the call to sound device
            call_slot = self.call.info().conf_slot
            lib.conf_connect(call_slot, 0)
            lib.conf_connect(0, call_slot)
            print "Hello world, I can talk!"


# Check command line argument
if len(sys.argv) != 2:
    print "Usage: simplecall.py <dst-URI>"
    sys.exit(1)

try:
    # Create library instance
    lib = pj.Lib()

    # Init library with default config
    lib.init(log_cfg = pj.LogConfig(level=3, callback=log_cb))

    # Create UDP transport which listens to any available port
    transport = lib.create_transport(pj.TransportType.UDP)

    # Start the library
    lib.start()

    # Create local/user-less account
    acc = lib.create_account_for_transport(transport)

    # Make call
    call = acc.make_call(sys.argv[1], MyCallCallback())

    # Wait for ENTER before quitting
    print "Press <ENTER> to quit"
    input = sys.stdin.readline().rstrip("\r\n")

    # We're done, shutdown the library
    lib.destroy()
    lib = None

except pj.Error, e:
    print "Exception: " + str(e)
    lib.destroy()
    lib = None
    sys.exit(1)

一切正常，我打电话给SIP服务器，听到“Hello，World”录音。

我需要更改哪些内容才能通过TCP进行此通信？ 到目前为止，我已经尝试通过在以下部分中将协议更改为tcp来更改我的pjsip.conf文件：

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0

，将;transport=tcp添加到我传入Python脚本的服务器的URI中，并更改Python脚本中创建传输的行

# Create UDP transport which listens to any available port
transport = lib.create_transport(pj.TransportType.UDP)

我得到的Python脚本会发生什么

pjsua_acc.c !...SIP registration failed, status=408 (Request Timeout)

在服务器上我根本看不到请求。

Answer 1

你确定你的Asterisk正在使用pjsip吗？旧版本（低于v.11）使用的是sip.conf而不是pjsip.conf。

您需要在 sip.conf ：

中全局启用tcp

[general]
tcpenable=yes
tcpbindaddr=0.0.0.0

如果您使用的是新的Asterisk，请将其添加到 pjsip.conf ：

[transport-tcp]
type=transport
protocol=tcp
bind=0.0.0.0:5060

[transport-tcp-out]
type=transport
protocol=tcp
bind=0.0.0.0:5060
#local_net=192.168.X.X/XX   //you might also set this

如何更改Hello World Asterisk示例以使用TCP而不是UDP？

1 个答案: