How to Test FXM board.docVIP

下载本文档

1
0
约2.43千字
约 4页
2017-06-08 发布于河南
举报
版权申诉

How to Test FXM board.doc

1、原创力文档（book118）网站文档一经付费（服务费），不意味着购买了该文档的版权，仅供个人/单位学习、研究之用，不得用于商业用途，未经授权，严禁复制、发行、汇编、翻译或者网络传播等，侵权必究。。
2、本站所有内容均由合作方或网友上传，本站不对文档的完整性、权威性及其观点立场正确性做任何保证或承诺！文档内容仅供研究参考，付费前请自行鉴别。如您付费，意味着您自己接受本站规则且自行承担风险，本站不退款、不进行额外附加服务；查看《如何避免下载的几个坑》。如果您已付费下载过本站文档，您可以点击这里二次下载。
3、如文档侵犯商业秘密、侵犯著作权、侵犯人身权等，请点击“版权申诉”（推荐），也可以打举报电话：400-050-0827(电话支持时间：9:00-18:30)。
4、该文档为VIP文档，如果想要下载，成为VIP会员后，下载免费。
5、成为VIP后，下载本文档将扣除1次下载权益。下载后，不支持退款、换文档。如有疑问请联系我们。
6、成为VIP后，您将拥有八大权益，权益包括：VIP文档下载权益、阅读免打扰、文档格式转换、高级专利检索、专属身份标志、高级客服、多端互通、版权登记。
7、VIP文档为合作方或网友上传，每下载1次，网站将根据用户上传文档的质量评分、类型等，对文档贡献者给予高额补贴、流量扶持。如果你也想贡献VIP文档。上传文档

How to Test FXM board

How to test FXM board Test method: using a sip terminal to call analog phone through the FXM board. e.g. asterisk installed on PC 2 IP:201.123.114.41 . SIP terminal take eyebeam for example installed on PC 1 IP:201.123.114.42 . a configuration on IP:201.123.114.41 After successful install the asterisk on IP:201.123.114.41. please following the below operations: Run command “vim /etc/asterisk/sip.conf”, delete the content of this file and input the below information. [general] bindport 5056 ; the asterisk SIP server port. The default value is 5060. srvlookup yes [0123] username 0123 ; type friend context dial ; the dialing rule of sip soft terminal host dynamic Open the file “/etc/asterisk/extensions.conf” , delete the content of this file and input the below information. [default] [incoming] exten s,1,Answer exten s,2,dial SIP/0123,10 exten 0123,1,dial SIP/0123,10 [dial] exten 120,1,dial dahdi/2/120,10 ;the port 2 is station port. exten 110,1,dial dahdi/1/110,50 ; the prot 1 is trunk port. open the file “/etc/asterisk/chan_dahdi.conf” and modify it as below. [trunkgroups] [channels] context text usecallerid yes hidecallerid no callwaiting yes usecallingpres yes callwaitingcallerid yes threewaycalling yes transfer yes canpark yes cancallforward yes callreturn yes echocancel yes echocancelwhenbridged yes relaxdtmf yes rxgain 0.0 txgain 0.0 group 1 callgroup 1 pickupgroup 1 immediate no pridialplan unknown prilocaldialplan unknown group 1 context incoming ; signalling fxs_ks channel 1 group 2 context dial signalling fxo_ks channel 2 start the asterisk server b configuration on IP:201.123.114.42 Open the eyebeam software and configure it According to the “sip.conf” file. BTW: If the sip successful registered , the asterisk interface will show below information. test example use sip terminal to dial 110 to call phoneA. use sip terminal to dial 120 to call phoneB. Use phoneA to dial 9999 to call sip terminal. Use phoneB to dial 012