asterisk

Join two conferences in asterisk

由 天大地大妈咪最大 提交于 2020-01-01 19:48:25
问题 I want to join two conferences (confbridge). Suppose there is a conference room c1 and then there is another conf room c2, now i want that participant of c1 can also join participants of c2 but the condition is that c1 participant don't have to leave their room. How can I implement this using confbridge and asterisk? 回答1: you have an implementation here: Connecting two conferences on initial join 回答2: considering you have two conferences rooms named 1 and 2 then add to your extensions.conf:

Replace characters in Asterisk Dialplan

由 柔情痞子 提交于 2020-01-01 19:10:09
问题 I want to change a couple off characters * # for A and P to have the monitor filename with characters more friendly. The only solution I could find was to do it my self within the dialplan but it generates a lot of verbosity output and is not as efficient(fast) as I would like to. I'll post it here just in case someone wants to use it. But I'm looking for an asterisk function that I can compile something that I can call withing the dialplan like ${REPLACE(${EXTEN},*,a)} and have the exten *

Are there parallels to Asterisk AMI and AGI in FreeSWITCH?

由 旧街凉风 提交于 2020-01-01 09:43:13
问题 Asterisk has Asterisk Manager Interface (AMI) and Asterisk Gateway Interface (AGI), using which one can trigger PHP scripts at certain events from Asterisk. Using the same PHP scripts can also instruct Asterisk what to do next to a call/conference. Are there parallels to AMI and AGI in FreeSWITCH? 回答1: These are good questions. I just wanted to add a few things to what @dkwiebe said (which is correct, BTW). The AMI equivalent in FreeSWITCH is the event socket. (Technically it's not the "Event

ConfBridge: end up conference when admin user exit

由 这一生的挚爱 提交于 2019-12-31 04:14:14
问题 I am doing something about ConfBridge these days. I used "confbridge kick all" to end the conferences when admin user exit, but the others in the conference will hear "you have been kick out from this conference.". it is not a good way to end a conference. Is there any configure or command which can set that conference will end up when the admin user exit the conference? 回答1: Just set endmarked=yes on the user profile you're using (found in confbridge.conf). The docs say: ;end_marked=yes ;

SIP trunk config文件

由 杀马特。学长 韩版系。学妹 提交于 2019-12-25 16:24:43
【推荐】2019 Java 开发者跳槽指南.pdf(吐血整理) >>> [General Settings]中 Maximum Channels 设置为CTS提供SIP Trunk的并发数量 [Outgoing Settings]中设置 PEER Details 为如下内容: username=12345678 (CTS提供的SIP Trunk配置号码) type=friend secret=xxxxxxxxxxxx (CTS提供的SIP Trunk配置密码) qualify=yes port=5060 nat=yes insecure=very host=voip.cts.sh.cn fromuser=12345678 (CTS提供的SIP Trunk配置号码) fromdomain=voip.cts.sh.cn dtmfmode=rfc2833 disallow=all context=from-trunk canreinvite=yes allow=gsm&ulaw&alaw&g729&ilbc [Incoming Settings]中设置 USER Details 为如下内容:(如果仅申请呼出号码无需填写) username=12345678 (CTS提供的SIP Trunk配置号码) type=friend secret=xxxxxxxxxxxx (CTS提供的SIP

如何将语音文件转换为Asterisk可以播放的音频格式文件(.wav)

由 江枫思渺然 提交于 2019-12-25 16:04:39
【推荐】2019 Java 开发者跳槽指南.pdf(吐血整理) >>> 1) 总所周知，Asterisk 支持播放的语音文件格式.wav 为： PCM非压缩编码,8Khz,16位,单声道，下面介绍几款转换工具 2）语音编码转换工具 1. GoldWave 【文件】——>【批量处理】，剩下的操作如下图了： 2，英文界面的WavePad Sound Editor 如下图 3. Audacity简体中文版,如下图： 修改为 单声道 ，16 bit ，如下图： 【文件】——>【导出】 来源： oschina 链接： https://my.oschina.net/u/939089/blog/145757

SIP Trunk / SIP 中继服务

由 风流意气都作罢 提交于 2019-12-25 16:01:35
【推荐】2019 Java 开发者跳槽指南.pdf(吐血整理) >>> CTS SIP Trunk/SIP中继可以让你现有的IPPBX连接到CTS的平台从而使用VOIP外线落地。享受拨打全国统一资费0.09元，多路中继号码，可接听电话。如果你已经有了IPPBX系统，使用CTS SIP Trunk企业级VOIP中继/SIP中继话务服务，可以让你在外呼电话时节省更多的钱 ，让您的VOIP系统对内对外都实现真正的 Over IP。使用CTS SIP Trunk/SIP中继无需硬件语音板卡，无需电话线路，只需一条网线，架设从内部分机到外呼电话的完全 VOIP 系统。不限企业地域，全国均可使用。如需办理请电话咨询：4008290998. 备注：使用CTS SIP Trunk服务，几乎任何的现有IPPBX均可以直接配置使用，无需任何硬件设备， 仅需支持SIP协议即可。例如：FreePBX，Asterisk，FreeSwitch，Trixbox，PBX IN A Flash，Elastix，3CX，AVAYA，Lync等等。 来源： oschina 链接： https://my.oschina.net/u/1452491/blog/200550

Asterisk拨号计划之匹配规则和优先级详解

由 妖精的绣舞 提交于 2019-12-25 15:46:19
【推荐】2019 Java 开发者跳槽指南.pdf(吐血整理) >>> 1. Asterisk拨号计划简介 自己查资料 2. Asterisk配置 先添加SIP分机 801，用软电话注册分机后，修改801分机的context=test-inc ，因为我们下面要探究Asterisk 基于 类似正则表达式的匹配 以及 include=>包含指令 的 优先级 。在asterisk拨号计划配置文件extensions.conf 中加入如下拨号规则 [test-inc] include => inc1 include => inc2 include => inc3 exten => _.,1,NooP(==00==) [inc1] exten => _33XX,1,NooP(==11==) [inc2] exten => _22.,1,NooP(==22==) [inc3] exten => 3333,1,NooP(==3333==) 3. 先直接给出结论 1）在 同一个context 中，对于正则匹配的规则， 分机越详细的规则 对应的APP就会被优先匹配执行。例如： exten=> _X5X.,n,APP1 和 exten=> _15X.,n,APP2 ,后面的正则匹配表达式“_15X.” 明显就比前面的要 详细和明确 ，所以对于 exten为 1581000000的手机号

Making A Call Using .Call file from Asterisk

由 ﹥＞﹥吖頭↗ 提交于 2019-12-25 04:53:30
问题 Hello thank you in advance for your help. I just installed chan_dongle with asterisk using one of the tutorials viewed this site. At the moment I can make and receive calls using a softphone, but you, I would like to make a call automatically asterisk. by the time I have. Extensions.conf ; Asterisk basic conf files ; By Giovanni Bosa - giovbs@gmail.com - ; Build 01/01/2013 [general] static=yes writeprotect=no clearglobalvars=no [globals] LINE1=SIP/101 [FromDongle] exten => +584147390386,1

Execute dialplan context from command line

由 ＊爱你＆永不变心＊ 提交于 2019-12-25 04:26:15
问题 I'm trying to execute an extension from the command line (via asterisk -rx "command") on a context that makes a AGI based query to determine which extension needs to be dialed (these extensions are updated on the DB). It's something like this: [autodialer] exten => 2,1,Answer() exten => 2,n,AGI(database_query.php); Makes a database query and generates vars exten => 2,n,Set(CALLERID(name)=${db_customer_name}); Sets callerid from DB data exten => 2,n,Dial(SIP/${db_customer_extension}); Also,
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