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主页 / computer / 问题

问题[asterisk](computer)

Martin Hope
elbarna
Asked: 2021-10-26 11:43:16 +0800 CST

星号,为什么我的isdn适配器TA可以接听但不能拨打电话?

  • 6

我有一个 b410p 卡 ISDN BRI,配置有 4 个 NT 端口。在一个 NT 端口 (2) 我连接一个 ISDN 电话(正在工作),在另一个 (4) 我连接一个连接到运行 Windows 2000 的 VM 的 TA(usb hamlet)(没有网络,所以很安全)运行软电话 isdn 称为 rvs-com。在 isdn 电话上,我可以拨打电话和接听电话。isdn TA 可以接听电话(但无法接听),但是当我从 TA 拨打电话时..星号控制台完全静音,我还使用命令设置了调试强度

pri set debug on span 4

当我用软电话拨打电话时,给出错误 isdn 3302,但似乎它没有连接到星号,因为正如我所说的控制台是静音的。可以是什么?我看到 isdn ta 在通话前放了一个国际前缀(+39),这是问题吗?无法从 Windows 中删除此规则(据我所知)。这些是我的 conf 文件

/etc/dahdi/system.conf

; Span 1: B4/0/1 "B4XXP (PCI) Card 0 Span 1" AMI/CCS YELLOW 
group=0,11
context=from-dahdi
switchtype = euroisdn
signalling = bri_net_ptmp
channel => 1-2
context = default
group = 63

; Span 2: B4/0/2 "B4XXP (PCI) Card 0 Span 2" AMI/CCS YELLOW 
group=0,12
context=from-dahdi
switchtype = euroisdn
signalling = bri_net_ptmp
channel => 4-5
context = default
group = 63

; Span 3: B4/0/3 "B4XXP (PCI) Card 0 Span 3" AMI/CCS YELLOW 
group=0,13
context=from-dahdi
switchtype = euroisdn
signalling = bri_net_ptmp
channel => 7-8
context = default
group = 63

; Span 4: B4/0/4 "B4XXP (PCI) Card 0 Span 4" (MASTER) AMI/CCS 
group=0,14
context=from-dahdi
switchtype = euroisdn
signalling = bri_net_ptmp
channel => 10-11
context = default
group = 63

/etc/asterisk/chan-dahdi.conf

[trunkgroups]
[channels]
language=it
context=local
switchtype=euroisdn
signalling=bri_net_ptmp
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
context=local
echocancel=yes
channel => 1,2,4,5,7,8,10,11
echocancelwhenbridged=yes
group=1
callgroup=1
pickupgroup=1
overlapdial=yes
immediate=no

/etc/asterisk/extensions.conf

[from-dahdi]
include => local
exten => 1796522,1,Dial(DAHDI/g12/${EXTEN})
exten => 1796522,2,Hangup()

exten => 1796521,1,Dial(DAHDI/g14/${EXTEN})
exten => 1796521,2,Hangup()

exten => s,1,Answer()
exten => s,2,Dial(Local/${EXTEN},10,t,m)
exten => s,4,Hangup()

[local]
include => from-dahdi

exten => 1001,1,Dial(SIP/telefono1,20,Ttm)
same  => n,Hangup

exten => 7500,1,VoicemailMain(@mycontext)

exten => 600,1,Answer()
exten => 600,2,Playback(demo-echotest) ; Let them know what
exten => 600,3,Echo()                  ; Do the echo test
exten => 600,4,Playback(demo-echodone) ; Let them know it
exten => 600,5,Hangup()
asterisk
  • 1 个回答
  • 27 Views
Martin Hope
elbarna
Asked: 2021-10-26 11:31:57 +0800 CST

星号:这个拨号方案发生了奇怪的事情

  • 5

我有一个连接到 Digium b410p 卡的 ISDN 电话。工作正常,但拨号方案有些奇怪,这个简单的拨号方案

[from-dahdi]
exten => 1796522,1,Dial(DAHDI/g12/${EXTEN})
exten => 1796522,2,Hangup()

exten => s,1,Wait(10)
exten => s,2,Answer
exten => s,3,Dial(DAHDI/g12/${EXTEN})

[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
include => default
include => from-dahdi

exten => 600,1,Playback(demo-echotest) ; Let them know what’s going on
exten => 600,2,Echo ; Do the echo test
exten => 600,3,Playback(demo-echodone) ; Let them know it’s over

当我拨打 600 时,我必须用自己的电话接听(我拨打 600 但我的电话说有来自我自己的号码 1796522 的电话!)如果我接听,我会听到音乐。

如果我删除行 exten => s,3,Dial(DAHDI/g12/${EXTEN}) 呼叫返回我一个超时我错过了什么?

asterisk
  • 1 个回答
  • 24 Views
Martin Hope
elbarna
Asked: 2021-10-25 11:58:09 +0800 CST

Asterisk 和 isdn 电话:可以“接听”电话,但去电电话看不到 Asterisk

  • 5

我已经配置了一个带有四个 NT 端口的 b410p isdn 卡。在第 4 个 NT 端口上,我放置了一个在 Windows 操作系统上的 TA,并运行一个旧的 isdn 电话软件(rvs 电话)。我已经像这样配置了 extensions.conf 文件

[from-dahdi]
exten => 084766508,1,Dial(DAHDI/g14/${EXTEN})
exten => 084766508,2,Hangup()
exten => _039.,1,Dial(DAHDI/4/${EXTEN})

如果从控制台我拨打 084766508 号码电话响铃(但如果我接听电话失败并出现 3302 isdn 错误),但如果我从 isdn 电话拨打测试号码(例如 200),则软件电话输入 039 前缀,所以我写了扩展_039。问题是星号看不到拨出电话,我也写了

pri set debug on span 4

但什么也不报告。

asterisk
  • 1 个回答
  • 169 Views
Martin Hope
elbarna
Asked: 2021-10-24 16:21:49 +0800 CST

Asterisk 和 isdn bri 卡给我错误 3302 isdn

  • 5

我正在尝试用星号配置 b410p 卡。我已将电话 isdn 连接到卡的 NT 端口。这是我的配置

/etc/dahdi/system.conf

# Span 1: B4/0/1 "B4XXP (PCI) Card 0 Span 1" AMI/CCS RED 
span=1,0,0,ccs,ami
# termtype: nt
bchan=1-2
hardhdlc=3
echocanceller=mg2,1-2
# Span 2: B4/0/2 "B4XXP (PCI) Card 0 Span 2" AMI/CCS RED 
span=2,0,0,ccs,ami
# termtype: nt
bchan=4-5
hardhdlc=6
echocanceller=mg2,4-5
# Span 3: B4/0/3 "B4XXP (PCI) Card 0 Span 3" AMI/CCS RED 
span=3,0,0,ccs,ami
# termtype: nt
bchan=7-8
hardhdlc=9
echocanceller=mg2,7-8
# Span 4: B4/0/4 "B4XXP (PCI) Card 0 Span 4" (MASTER) AMI/CCS 
span=4,0,0,ccs,ami
# termtype: nt
bchan=10-11
hardhdlc=12
echocanceller=mg2,10-11

/etc/asterisk/chan_dahdi.conf

[trunkgroups]
[channels]
language=it
context=local
switchtype=euroisdn
signalling=bri_net_ptmp
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
context= incoming
echocancel=yes
channel => 1,2,4,5,7,8,10,11
echocancelwhenbridged=yes
group=1
callgroup=1
pickupgroup=1

/etc/asterisk/dahdi-channels.conf

; Span 1: B4/0/1 "B4XXP (PCI) Card 0 Span 1" AMI/CCS RED 
group=0,11
context=from-dahdi
switchtype = euroisdn
signalling = bri_net_ptmp
channel => 1-2
context = default
group = 63
; Span 2: B4/0/2 "B4XXP (PCI) Card 0 Span 2" AMI/CCS RED 
group=0,12
context=from-dahdi
switchtype = euroisdn
signalling = bri_net_ptmp
channel => 4-5
context = default
group = 63
; Span 3: B4/0/3 "B4XXP (PCI) Card 0 Span 3" AMI/CCS RED 
group=0,13
context=from-dahdi
switchtype = euroisdn
signalling = bri_net_ptmp
channel => 7-8
context = default
group = 63
; Span 4: B4/0/4 "B4XXP (PCI) Card 0 Span 4" (MASTER) AMI/CCS 
group=0,14
context=from-dahdi
switchtype = euroisdn
signalling = bri_net_ptmp
channel => 10-11
context = default
group = 63

连接电话的端口是绿色的,所以没问题

dahdi_tool 说:

RED             B4XXP (PCI) Card 0 Span 1                       ↑  │ 
                          │     RED             B4XXP (PCI) Card 0 Span 2                       ▒  │ 
                          │     RED             B4XXP (PCI) Card 0 Span 3                       ▮  │ 
                          │     OK              B4XXP (PCI) Card 0 Span 4 

从 isdn 电话我尝试调用 200 测试扩展并退出 isdn 错误代码 3302

这是我的 extensions.conf 部分

exten => 200,1,Answer()
same => n,Playback(hello-world)
same => n,Hangup()

exten => 1000,1,Dial(dahdi/4,20)

当我尝试从控制台调用“1000”时说

 Span 2: Channel 0/1 got hangup, cause 18
    -- Hungup 'DAHDI/i2/-2'
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'ALSA/default' status is 'CHANUNAVAIL'
  << Hangup on console >> 

一些忠告?谢谢

Pri 调试报告此(我已将扩展名从 1000 更改为 039991122 和 extensions.conf

039991122
    -- Executing [039991122@local:1] Dial("ALSA/default", "DAHDI/g14/039991122") in new stack
PRI Span: 4 -- Making new call for cref 32773
    -- Requested transfer capability: 0x00 - SPEECH
PRI Span: 4 Sending message for call 0x7f0d84008b20 on call->link: 0x782140 with TEI/SAPI 127/63
PRI Span: 4 
PRI Span: 4 > Protocol Discriminator: Q.931 (8)  len=26
PRI Span: 4 > TEI=127 Call Ref: len= 1 (reference 5/0x5) (Sent from originator)
PRI Span: 4 > Message Type: SETUP (5)
PRI Span: 4 > [04 03 80 90 a3]
PRI Span: 4 > Bearer Capability (len= 5) [ Ext: 1  Coding-Std: 0  Info transfer capability: Speech (0)
PRI Span: 4 >                              Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
PRI Span: 4 >                                User information layer 1: A-Law (35)
PRI Span: 4 > [18 01 89]
PRI Span: 4 > Channel ID (len= 3) [ Ext: 1  IntID: Implicit  BRI  Spare: 0  Exclusive  Dchan: 0
PRI Span: 4 >                       ChanSel: B1 channel
PRI Span: 4 >                     ]
PRI Span: 4 > [70 0b 80 30 37 38 34 33 36 36 35 30 38]
PRI Span: 4 > Called Party Number (len=13) [ Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)  '039991122' ]
PRI Span: 4 > [a1]
PRI Span: 4 > Sending Complete (len= 1)
PRI Span: 4 q931.c:6531 q931_setup: Call 32773 enters state 1 (Call Initiated).  Hold state: Idle
    -- Called DAHDI/g14/039991122
PRI Span: 4 T303 timed out.  cref:32773
PRI Span: 4 Sending message for call 0x7f0d84008b20 on call->link: 0x782140 with TEI/SAPI 127/63
PRI Span: 4 
PRI Span: 4 > Protocol Discriminator: Q.931 (8)  len=26
PRI Span: 4 > TEI=127 Call Ref: len= 1 (reference 5/0x5) (Sent from originator)
PRI Span: 4 > Message Type: SETUP (5)
PRI Span: 4 > [04 03 80 90 a3]
PRI Span: 4 > Bearer Capability (len= 5) [ Ext: 1  Coding-Std: 0  Info transfer capability: Speech (0)
PRI Span: 4 >                              Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
PRI Span: 4 >                                User information layer 1: A-Law (35)
PRI Span: 4 > [18 01 89]
PRI Span: 4 > Channel ID (len= 3) [ Ext: 1  IntID: Implicit  BRI  Spare: 0  Exclusive  Dchan: 0
PRI Span: 4 >                       ChanSel: B1 channel
PRI Span: 4 >                     ]
PRI Span: 4 > [70 0b 80 30 37 38 34 33 36 36 35 30 38]
PRI Span: 4 > Called Party Number (len=13) [ Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)  '039991122' ]
PRI Span: 4 > [a1]
PRI Span: 4 > Sending Complete (len= 1)
PRI Span: 4 T303 timed out.  cref:32773
PRI Span: 4 q931.c:6415 t303_expiry: Call 32773 enters state 22 (Call Abort).  Hold state: Idle
PRI Span: 4 Fake clearing.  cref:32773
PRI Span: 4 q931.c:9910 pri_internal_clear: alive 1, hangupack 1
Span 4: Processing event PRI_EVENT_HANGUP(6)
    -- Span 4: Channel 0/1 got hangup, cause 18
PRI Span: 4 q931.c:7270 q931_hangup: Hangup master cref:32773
PRI Span: 4 q931.c:7312 q931_hangup: Remaining slaves 0
    -- Hungup 'DAHDI/i4/039991122-5'
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [039991122@local:2] Hangup("ALSA/default", "") in new stack
  == Spawn extension (local, 039991122, 2) exited non-zero on 'ALSA/default'
  << Hangup on console >> 
PRI Span: 4 T312 timed out.  cref:32773
PRI Span: 4 Destroying call 0x7f0d84008b20, ourstate Call Abort, peerstate Call Present, hold-state Idle

另一个问题是:来电正在“工作”,因为电话响了(当应答失败时),来电没有被 pri 调试报告:我错过了什么?这是新的 extensions.conf

[from-dahdi]
exten => _XXXXX.,1,Dial(DAHDI/g14/${EXTEN})
exten => _XXXXX.,2,Hangup()

如何定义拨出电话的分机?

asterisk
  • 1 个回答
  • 74 Views
Martin Hope
Roman
Asked: 2021-07-04 02:35:21 +0800 CST

星号:如何获取当前的 sip.conf 设置?

  • 6

我不确定是否rtptimeout设置了 sip 通用参数(例如 )。中没有提到它/etc/asterisk/sip.conf。但可能是在其他地方设置(例如在包含文件中)。如何在运行时检查此类参数的当前活动值(可能使用 CLI)?

asterisk voip
  • 1 个回答
  • 224 Views
Martin Hope
Roman
Asked: 2021-07-03 11:44:05 +0800 CST

星号:如何结束冻结的呼叫

  • 5

有时我的 voip 提供商会关闭服务而不结束正在进行的呼叫(这些呼叫传入我的服务器并且它们正在排队)。所以通话永远不会优雅地结束,我永远不会得到 BYE,而且我的队列中的同伴永远处于“通话中”状态。我只能通过重新启动星号服务来重置它们的状态。

如何设置星号以自动结束此类呼叫?这种情况是否有某种超时?

asterisk voip
  • 1 个回答
  • 79 Views
Martin Hope
Eugene
Asked: 2021-04-06 11:08:22 +0800 CST

Asterisk 无法接听来电

  • 5

我已经配置了一个本地 Asterisk 服务器。有本地用户和一个用于外部呼叫的中继线。

用户可以成功拨打本地和拨出电话。但是,当外部呼叫到达服务器时,它会响应 401 错误(1.1.1.1 代表提供商 IP,2.2.2.2 代表本地服务器)。

邀请:

<--- Received SIP request (1079 bytes) from UDP:1.1.1.1:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK2b074f13;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as5b4c0767
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 105 INVITE
User-Agent: VoIPVTK
Authorization: Digest username="s", realm="asterisk", algorithm=MD5, uri="sip:[email protected]", nonce="1617646804/4c072b7aa1d5799d84d11f1e857387a0", response="7e12bdf0955235437f8d49fee0fc517a", opaque="1d4bee854523cdda", qop=auth, cnonce="3c7da01c", nc=00000003
Date: Mon, 05 Apr 2021 18:20:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 226

v=0
o=VoIPVTK 1371915978 1371915981 IN IP4 1.1.1.1
s=VoIPVTK
c=IN IP4 1.1.1.1
t=0 0
m=audio 11250 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

回复:

[Apr  5 21:20:04] NOTICE[72898]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'INVITE' from '<sip:[email protected]>' failed for '1.1.1.1:5060' (callid: [email protected]:5060) - Failed to authenticate
<--- Transmitting SIP response (502 bytes) to UDP:1.1.1.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 1.1.1.1:5060;rport=5060;received=1.1.1.1;branch=z9hG4bK2b074f13
Call-ID: [email protected]:5060
From: <sip:[email protected]>;tag=as5b4c0767
To: <sip:[email protected]>;tag=z9hG4bK2b074f13
CSeq: 105 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1617646804/4c072b7aa1d5799d84d11f1e857387a0",opaque="52b0860022fc8faa",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.2.0
Content-Length:  0

提供者也发送失败并出现相同错误的 OPTIONS 请求。来自我们服务器的 OPTIONS 请求成功。

似乎提供者未经授权发送数据包,而服务器需要它。

这怎么能解决?

asterisk
  • 1 个回答
  • 376 Views
Martin Hope
nd_
Asked: 2019-10-18 02:02:06 +0800 CST

Asterisk、IAXModem 和 HylaFax:未检测到运营商

  • 5

我尝试在运行的 Debian 10 服务器上安装 HylaFax + IAXModem + Asterisk 设置。服务器直接连接到 Internet,防火墙被禁用(没有设置规则)。服务器唯一要做的就是发送传真。我使用 sipgate 中继。在 HylaFax 尝试发送传真期间,我在 Asterisk 控制台中看到了此输出(SIP 调试和详细程度为 on/10):

    -- Accepting AUTHENTICATED call from 127.0.0.1:4570:
    --        > requested format = alaw,
    --        > requested prefs = (),
    --        > actual format = alaw,
    --        > host prefs = (alaw),
    --        > priority = mine
    -- Executing [RECIPIENT@fax_out:1] Set("IAX2/iaxmodem-7708", "CALLERID(num)=CALLER") in new stack
    -- Executing [RECIPIENT@fax_out:2] Set("IAX2/iaxmodem-7708", "CALLERID(name)=CALLER") in new stack
    -- Executing [RECIPIENT@fax_out:3] SIPAddHeader("IAX2/iaxmodem-7708", "P-Preferred-Identity:<sip:CALLER>") in new stack
    -- Executing [RECIPIENT@fax_out:4] Dial("IAX2/iaxmodem-7708", "SIP/sipconnect.sipgate.de/RECIPIENT,,r") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 17702
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 217.10.68.150:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5060;branch=z9hG4bK12349644
Max-Forwards: 70
From: "CALLER" <sip:[email protected]>;tag=as65bc91b0
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Date: Mon, 21 Oct 2019 12:22:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Preferred-Identity: <sip:CALLER>
P-Asserted-Identity: "CALLER" <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 270

v=0
o=root 698738770 698738770 IN IP4 127.0.1.1
s=Asterisk PBX 16.2.1~dfsg-1+deb10u1
c=IN IP4 127.0.1.1
t=0 0
m=audio 17702 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
    -- Called SIP/sipconnect.sipgate.de/RECIPIENT

<--- SIP read from UDP:217.10.68.150:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 127.0.1.1:5060;rport=5060;received=X.X.X.X;branch=z9hG4bK12349644
From: "CALLER" <sip:[email protected]>;tag=as65bc91b0
To: <sip:[email protected]:5060>;tag=c63713a666d5644779294882ed89253a.0c69
Call-ID: [email protected]
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="sipconnect.sipgate.de", nonce="Xa2kKF2tovxRNF/AcCiPaUlB/z/ev7jl"
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 217.10.68.150:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5060;branch=z9hG4bK12349644
Max-Forwards: 70
From: "CALLER" <sip:[email protected]>;tag=as65bc91b0
To: <sip:[email protected]:5060>;tag=c63713a666d5644779294882ed89253a.0c69
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Content-Length: 0


---
Audio is at 17702
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 217.10.68.150:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5060;branch=z9hG4bK248f1ffc
Max-Forwards: 70
From: "CALLER" <sip:[email protected]>;tag=as65bc91b0
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Proxy-Authorization: Digest username="USER", realm="sipconnect.sipgate.de", algorithm=MD5, uri="sip:[email protected]:5060", nonce="Xa2kKF2tovxRNF/AcCiPaUlB/z/ev7jl", response="42016e991f588a252062bb86b35a3f6c"
Date: Mon, 21 Oct 2019 12:22:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Preferred-Identity: <sip:CALLER>
P-Asserted-Identity: "CALLER" <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 270

v=0
o=root 698738770 698738771 IN IP4 127.0.1.1
s=Asterisk PBX 16.2.1~dfsg-1+deb10u1
c=IN IP4 127.0.1.1
t=0 0
m=audio 17702 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:217.10.68.150:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 127.0.1.1:5060;rport=5060;received=X.X.X.X;branch=z9hG4bK248f1ffc
From: "CALLER" <sip:[email protected]>;tag=as65bc91b0
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:217.10.68.150:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 127.0.1.1:5060;rport=5060;received=X.X.X.X;branch=z9hG4bK248f1ffc
Record-Route: <sip:172.20.40.8;lr>
Record-Route: <sip:217.10.68.150;lr;ftag=as65bc91b0>
From: "CALLER" <sip:[email protected]>;tag=as65bc91b0
To: <sip:[email protected]:5060>;tag=as21618100
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:[email protected]:5060>
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:217.10.68.150;lr;ftag=as65bc91b0>
sip_route_dump: route/path hop: <sip:172.20.40.8;lr>
    -- SIP/sipconnect.sipgate.de-00000003 is ringing

<--- SIP read from UDP:217.10.68.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.1.1:5060;rport=5060;received=X.X.X.X;branch=z9hG4bK248f1ffc
Record-Route: <sip:172.20.40.8;lr>
Record-Route: <sip:217.10.68.150;lr;ftag=as65bc91b0>
From: "CALLER" <sip:[email protected]>;tag=as65bc91b0
To: <sip:[email protected]:5060>;tag=as21618100
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 298

v=0
o=root 1290589385 1290589385 IN IP4 217.116.117.70
s=sipgate VoIP GW
c=IN IP4 212.9.44.253
t=0 0
m=audio 15550 RTP/AVP 8 0 101
a=silenceSupp:off - - - -
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:15551
a=ptime:20
<------------->
--- (13 headers 14 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
       > 0x7f4634015640 -- Strict RTP learning after remote address set to: 212.9.44.253:15550
Peer audio RTP is at port 212.9.44.253:15550
sip_route_dump: route/path hop: <sip:217.10.68.150;lr;ftag=as65bc91b0>
sip_route_dump: route/path hop: <sip:172.20.40.8;lr>
set_destination: Parsing <sip:217.10.68.150;lr;ftag=as65bc91b0> for address/port to send to
set_destination: set destination to 217.10.68.150:5060
Transmitting (no NAT) to 217.10.68.150:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5060;branch=z9hG4bK12463de5
Route: <sip:217.10.68.150;lr;ftag=as65bc91b0>,<sip:172.20.40.8;lr>
Max-Forwards: 70
From: "CALLER" <sip:[email protected]>;tag=as65bc91b0
To: <sip:[email protected]:5060>;tag=as21618100
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Content-Length: 0


---
    -- SIP/sipconnect.sipgate.de-00000003 answered IAX2/iaxmodem-7708
    -- Channel SIP/sipconnect.sipgate.de-00000003 joined 'simple_bridge' basic-bridge <262f871a-8fc7-4bb9-a64b-981378a84acd>
    -- Channel IAX2/iaxmodem-7708 joined 'simple_bridge' basic-bridge <262f871a-8fc7-4bb9-a64b-981378a84acd>
       > 0x7f4634015640 -- Strict RTP switching to RTP target address 212.9.44.253:15550 as source

<--- SIP read from UDP:217.10.68.150:5060 --->

<------------->

<--- SIP read from UDP:217.10.68.150:5060 --->
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 217.10.68.150;branch=z9hG4bKd711.674d11475dd9179e68c4eb52c2088642.0
Via: SIP/2.0/UDP 172.20.40.8;branch=z9hG4bKd711.39ae43c7fc3d82443eba26f3d75b5f39.0
Via: SIP/2.0/UDP 217.116.117.70:5060;branch=z9hG4bK5d2cbd05
Max-Forwards: 68
From: <sip:[email protected]:5060>;tag=as21618100
To: "CALLER" <sip:[email protected]>;tag=as65bc91b0
Call-ID: [email protected]
CSeq: 102 BYE
Reason: Q.850;cause=16
Content-Length: 0
X-hint: rr-enforced

<------------->
--- (12 headers 0 lines) ---
Sending to 217.10.68.150:5060 (no NAT)
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: BYE)

<--- Transmitting (no NAT) to 217.10.68.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.68.150;branch=z9hG4bKd711.674d11475dd9179e68c4eb52c2088642.0;received=217.10.68.150
Via: SIP/2.0/UDP 172.20.40.8;branch=z9hG4bKd711.39ae43c7fc3d82443eba26f3d75b5f39.0
Via: SIP/2.0/UDP 217.116.117.70:5060;branch=z9hG4bK5d2cbd05
From: <sip:[email protected]:5060>;tag=as21618100
To: "CALLER" <sip:[email protected]>;tag=as65bc91b0
Call-ID: [email protected]
CSeq: 102 BYE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Channel SIP/sipconnect.sipgate.de-00000003 left 'simple_bridge' basic-bridge <262f871a-8fc7-4bb9-a64b-981378a84acd>
    -- Channel IAX2/iaxmodem-7708 left 'simple_bridge' basic-bridge <262f871a-8fc7-4bb9-a64b-981378a84acd>
  == Spawn extension (fax_out, RECIPIENT, 4) exited non-zero on 'IAX2/iaxmodem-7708'
    -- Hungup 'IAX2/iaxmodem-7708'
Really destroying SIP dialog '[email protected]' Method: BYE

在/var/spool/hylafax/log/xferfaxlog:

10/21/19 15:38  SEND    000000096       ttyIAX0 41      ""      [email protected]        "RECIPIENT"   ""      2220072 0       0:00:03 0:00:00 "No carrier detected"   ""      ""      ""      "root"  "00 00 00"

(我在这里替换了发件人/收件人号码和用户名)

防火墙当前未运行:

root@asterisk:/etc/asterisk# iptables -nL
Chain INPUT (policy ACCEPT)
target     prot opt source               destination

Chain FORWARD (policy ACCEPT)
target     prot opt source               destination

Chain OUTPUT (policy ACCEPT)
target     prot opt source               destination

这是我的星号配置:

在/etc/asterisk/sip.conf:

[general]
context=unauthenticated
bindport=5060
bindaddr=0.0.0.0
realm=domain.tld
externhost=domain.tld:5060
localnet=127.0.0.0/255.255.255.0
nat=no
srvlookup=yes
allowguest=no
alwaysauthreject=yes

register => USER:[email protected]/USER
[sipconnect.sipgate.de]
type=peer
host=sipconnect.sipgate.de
outboundproxy=sipconnect.sipgate.de
port=5060
username=USER
defaultuser=USER
fromuser=USER
fromdomain=sipconnect.sipgate.de
secret=PASS
dtmfmode=rfc2833
insecure=port,invite
canreinvite=no
directmedia=no
registertimeout=600
sendrpid=pai
usereqphone=no
t38pt_udptl=no
disallow=all
allow=alaw
allow=ulaw
qualify=yes
context=unauthenticated

(我在这里替换了凭据和域名)

在/etc/asterisk/extensions.conf:

[general]

[sipgate_in]
exten => sipgate_in,1,Goto(siptrunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)

[siptrunk]
exten => 1234567,1,Dial(IAX2/iaxmodem)
exten => 1234567,n,Hangup

[fax_out]
exten => _X.,1,Set(CALLERID(num)=00491231234567)
exten => _X.,n,Set(CALLERID(name)=${CALLERID(num)})
exten => _X.,n,SipAddHeader(P-Preferred-Identity:<sip:${CALLERID(num)}>)
exten => _X.,n,Dial(SIP/sipconnect.sipgate.de/${EXTEN},,r)

[unauthenticated]

(我这里换了发件人号码)

在/etc/asterisk/iax.conf:

[general]
bindport=4569
bindaddr=127.0.0.1
calltokenoptional=127.0.0.1/255.255.255.0

[iaxmodem]
port=4570
type=friend
host=dynamic
qualify=yes
secret=pwd
requirecalltoken=no
disallow=all
allow=alaw
jitterbuffer=no
trunk=no
context=fax_out

(我在这里替换了凭据)

在/etc/iaxmodem/ttyIAX0:

device     /dev/ttyIAX0
owner      uucp:uucp
mode       660
port       4570
refresh    60
server     127.0.0.1
peername   iaxmodem
secret     pwd
codec      alaw
nojitterbuffer

(我在这里替换了凭据)

IAX调制解调器ttyIAX0注册成功,Asterisk 在 sipgate 中继线上。从已知的工作设置向收件人发送传真成功。以前我收到了一些网络协议错误,但由于我在测试期间没有激活防火墙,这些来自任何尝试注册为设备等...

接收方是运行不使用 T.38 的 3CX,所以我也在我的配置中禁用了 T.38,以确保 T.38 不是问题。

据我了解,调试输出显示目标设备在发送传真之前挂断。我看对了吗?有谁知道为什么通信会这样?怎么才能知道早挂机的原因?

更新:我现在可以向另一个目标号码发送传真。也许我做的一切都是正确的,但是 3CX 传真出了问题。但我仍然不确定调试协议 - 在这种情况下它看起来应该吗?

更新 2:目标 3CX 已启用 T.38 回退,现在它从我的 Asterisk PBX 接收传真。我不太了解那个 3CX 配置,那个“后备”是什么意思——但是,它现在可以工作了。我仍然很好奇我怎么能得到早期挂断的原因——也许根本不可能。我希望我的配置现在对现实生活有好处......

asterisk
  • 1 个回答
  • 440 Views
Martin Hope
Jaques
Asked: 2019-07-02 01:32:40 +0800 CST

Asterisk 16.4 pjsip 中继注册

  • 5

我不是星号专家,我现在被困住了。

我正在尝试实时设置一个星号框。大多数工作正常,而且我的端点能够在彼此之间进行调用。但我需要为 VOIP 提供商设置 SIP 中继,但我不确定该怎么做,因为我所做的不起作用。

  1. 第一个问题是我的注册没有加载。据我了解,需要为不同的对象设置 sorcery.conf 文件。我确定类型基本上映射到中的类型pjsip.conf,所以我在那里设置了信息。就像我说的,我的中继的 aors、endpoint 和 auth 反映并加载了,但我似乎无法获得注册。

我已经尝试在 pjsip.conf 以及 mysql 数据库中设置注册(和身份),但是当我运行时pjsip show registrations,没有找到任何对象。我想也许是因为sorcery.conf没有映射表,但是当我添加registration=realtime,ps_registrations到 sorcery.conf 时,pjsip 根本没有启动

这是我当前的 sorcery.conf 设置。我已经包含了 incase 的注释行

[res_pjsip]
endpoint=realtime,ps_endpoints
auth=realtime,ps_auths
aor=realtime,ps_aors
domain_alias=realtime,ps_domain_aliases
;registration=realtime,ps_registrations

[res_pjsip_endpoint_identifier_ip]
identify=realtime,ps_endpoint_id_ips

;[res_pjsip_outbound_publish]
;outbound-publish=realtime,ps_outbound_publishes

;[res_pjsip_pubsub]
;inbound-publication=realtime,ps_inbound_publications

;[res_pjsip_publish_asterisk]
;asterisk-publication=realtime,ps_asterisk_publications

我在 pjsip.conf 中也有相同的设置

[mtntrunk]
type=registration
outbound_auth=mtntrunk_auth
server_uri=sip:<ip address of provider>
client_uri=sip:<number>@<ip address of provider>
retry_interval=60
;forbidden_retry_interval=600
;expiration=3600
;line=yes
endpoint=mtntrunk

当我运行命令时端点mtntrunk确实显示pjsip show endpoints

如果您需要更多信息,请询问。

有什么我想念的吗?我的配置不正确吗?我应该配置配置文件以及实时配置来使它工作吗?

asterisk
  • 1 个回答
  • 2841 Views
Martin Hope
POMATu
Asked: 2019-05-28 15:16:58 +0800 CST

星号如何通过名称而不是数字来调用

  • 5

我还没有在任何地方找到清晰的样本,我该如何为内部号码设置字母数字别名。

例如,我现在打电话给 504@ip。

我怎样才能通过编辑配置让 john@ip 达到 504 号码?

我发现的所有扩展示例都有很多关于屏蔽的信息,但没有简单的“john”别名配置示例。我希望 asterisk 将 john 视为 504@ip,将 kate 视为 505@other_server_ip。我可以在每个 asterisk 服务器上轻松设置吗?

谢谢

alias asterisk
  • 1 个回答
  • 429 Views

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