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Asterisk 的出站音频问题

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Asterisk 的出站音频问题

我在客户所在地的 CentOS 服务器上运行 Asterisk 16.9.0。该服务器通过电信公司提供的 SIP 中继进行连接。

我在那里的几台电脑上配置了 Zoiper，使用 IAX 连接可以接通来电和拨出电话。出于某种原因，通过 SIP 连接会出现音频问题。现在，分机与分机之间的通话、分机与外部之间的通话以及来电都运行正常，没有问题。

现在我正尝试从办公室的 PC 连接到该服务器。没有 VPN；端口 5060 已从客户端的防火墙转发。

从 Zoiper，我使用 IAX 进行连接，控制台显示客户端已注册。我可以拨打外线号码，并能听清被叫方的声音。被叫方听到的是我的声音失真严重。

使用 Grandstream GXP1625，控制台显示 SIP 客户端已连接。拨出后，我可以听到被叫方的声音，但他听不到声音，通话在 60 秒后断开。

我应该检查/改变什么？

Connected to Asterisk 16.9.0 currently running on server2 (pid = 2733)
    -- Registered SIP '1001' at xx.xx.xx.xx:61548
    -- Unregistered SIP '1001'
    -- Registered IAX2 '1001' (UNAUTHENTICATED) at xx.xx.xx.xx:4569
    -- Accepting UNAUTHENTICATED call from xx.xx.xx.xx:4569:
    --        > requested format = opus,
    --        > requested prefs = (),
    --        > actual format = gsm,
    --        > host prefs = (gsm|ilbc|speex|g729|g723|ulaw|alaw|g726|g726aal2|adpcm|slin|slin16|lpc10|speex16|g722|siren7|siren14|testlaw|g719|opus...),
    --        > priority = mine
    -- Executing [x-Called num-x@test:1] Set("IAX2/1001-3187", "CALLERID(num)=99222333") in new stack
    -- Executing [x-Called num-x@test:2] Dial("IAX2/1001-3187", "SIP/x-Called num-x@ntc-out") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/x-Called num-x@ntc-out
    -- SIP/ntc-out-0000001b is ringing
    -- SIP/ntc-out-0000001b is making progress passing it to IAX2/1001-3187
    -- SIP/ntc-out-0000001b answered IAX2/1001-3187
    -- Channel SIP/ntc-out-0000001b joined 'simple_bridge' basic-bridge <f874b82d-74f3-430f-8475-2ea62dec3ff6>
    -- Channel IAX2/1001-3187 joined 'simple_bridge' basic-bridge <f874b82d-74f3-430f-8475-2ea62dec3ff6>
    -- Channel SIP/ntc-out-0000001b left 'simple_bridge' basic-bridge <f874b82d-74f3-430f-8475-2ea62dec3ff6>
    -- Channel IAX2/1001-3187 left 'simple_bridge' basic-bridge <f874b82d-74f3-430f-8475-2ea62dec3ff6>
  == Spawn extension (test, x-Called num-x, 2) exited non-zero on 'IAX2/1001-3187'
    -- Hungup 'IAX2/1001-3187'
    -- Unregistered IAX2 '1001' (UNAUTHENTICATED)

    -- Registered SIP '1001' at xx.xx.xx.xx:53486
  == Using SIP RTP CoS mark 5
    -- Executing [x-Called num-x@test:1] Set("SIP/1001-0000001d", "CALLERID(num)=99222333") in new stack
    -- Executing [x-Called num-x@test:2] Dial("SIP/1001-0000001d", "SIP/x-Called num-x@ntc-out") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/x-Called num-x@ntc-out
    -- SIP/ntc-out-0000001e is ringing
    -- SIP/ntc-out-0000001e is making progress passing it to SIP/1001-0000001d
    -- SIP/ntc-out-0000001e answered SIP/1001-0000001d
    -- Channel SIP/ntc-out-0000001e joined 'simple_bridge' basic-bridge <57e8fc55-7721-4465-840c-3d37bc76329c>
    -- Channel SIP/1001-0000001d joined 'simple_bridge' basic-bridge <57e8fc55-7721-4465-840c-3d37bc76329c>
[Mar 17 22:13:26] WARNING[2978]: chan_sip.c:4126 retrans_pkt: Retransmission timeout reached on transmission 187fd4970375442eafe988f72e7ce772 for seqno 413 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
[Mar 17 22:13:26] WARNING[2978]: chan_sip.c:4150 retrans_pkt: Hanging up call 187fd4970375442eafe988f72e7ce772 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
    -- Channel SIP/1001-0000001d left 'simple_bridge' basic-bridge <57e8fc55-7721-4465-840c-3d37bc76329c>
    -- Channel SIP/ntc-out-0000001e left 'simple_bridge' basic-bridge <57e8fc55-7721-4465-840c-3d37bc76329c>
  == Spawn extension (test, x-Called num-x, 2) exited non-zero on 'SIP/1001-0000001d'
    -- Unregistered SIP '1001'

编辑 查看日志，我发现 SIP 连接存在网络问题，因为服务器无法将数据包发送回我的客户端 PC。因此，我通过 VPN 连接到客户端的 LAN，并将客户端上的连接地址从防火墙的外部 IP 更改为 Asterisk 服务器的 LAN IP。结果相同。

编辑 sip.conf 用于电话线路。该线路连接到服务器的第二个网络端口。

[telco]
type=friend
description=Telco
context=from-telco
externip = 10.12.15.222 <-- My server's 2nd port IP
localnet = 10.12.15.224/29
host=10.12.15.225  <-- telco's IP
media_address =10.12.15.226
permit=10.12.15.224/29
username=
fromuser=
secret=
insecure=invite
canreinvite=yes
qualify=yes
dtmfmode=rfc2833
allow=ulaw,alaw,gsm
nat=no

用户配置文件：

[101]
type=peer
username=clientpc
callerid="Client PC" <101>
secret=123123123
context=test
host=dynamic
qualify=yes
allow=all
nat=no

[1001]
type=peer
callerid="My office" <1001>
username=hussain
password=111222333
context=test
host=dynamic
allow=all
qualify=yes
nat=force_rport,comedia

答案1

您应该阅读更多有关网络和 NAT 遍历的内容

https://www.voip-info.org/asterisk-sip-nat/

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