asterisk

Asterisk= 1.8.11.0 Android= 2.3/4.0.3 Android Sip client=Native Android sip client/sipdemo When i call from my pc using zoiper/xlite to android (native android sip client) now i can hear audio from both sides but when i make call from android to pc (zoiper/xlite) i cannot hear anything on android. On the other hand i have tested this scenario on elastix (which also uses asterisk 1.8.11.0) with no problem in audio. pc(zoiper) ip 192.168.15.27 android ip 192.168.15.71 asterisk server ip 192.168.15.118 Sip debug when calling from android to zoiper . <--- SIP read from UDP:192.168.15.71:45616 --->

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? Michael Collins 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 Socket Library" or ESL, which is an abstraction layer for writing programs that use the

I am using Asterisk 1.8 on Ubuntu. the problem is however I change the amd.conf configurations AMD() call detects AMDSTATUS as MACHINE. Please guide me on the right settings for AMD detection if a person picks up phone and says 'Hi this is so and so' vs. machine picks up with a 10-19 minute message followed by a beep. I am willing to pay if some one can solve this for me. Here is my amd.conf settings: [general] initial_silence = 2250 ; Maximum silence duration before the greeting. ; If exceeded then MACHINE. greeting = 15000 ; Maximum length of a greeting. If exceeded then MACHINE. after

I'm trying to create a VoIP based IVR service that interacts with a web application. From what I understand, adhearson runs on top of asterisk. What else do I need to have on the server to satisfy the equation? I think I need a way for asterisk to connect to a voip account. I'd appreciate any help and/or phrases to google. If you want to build your IVR quickly and easily you'd be better off looking at something like tropo.com (tropo's parent, voxeo, own adhearsion) or twilio.com, they've done a lot of the grunt work for you and setting up Asterisk is not for the faint hearted. If you want

I would like to have pre-emption calls in Asterisk. With this I mean that if user A has priority/access-level 1 and wants to talk to user B, how could it preempt the call that user B is already having with user C which has only priority/access-level 2? Does anyone know if this is supported by Asterisk or how this could be implemented? Any idea would be very welcome. No, it not supported by asterisk. But yes, it can be implemented using dialplan+some script magic. Complexity is high and require expert or guru skill. Short plan is following: check if B is in call (need use DEVICE_STATE or GROUP

I need to modify sip.conf with AMI, adding a new user to it. Everything works fine, and I can create a user like this without problems: [1000] secret=pass12 But I have to create user with template like [1000](mytemp) secret=pass12 and I don't know how to do this. Neither Google, nor Digium forum can't help me. P.S. I use JavaScript asterisk-manager to interact with Asterisk, and here is my code, which adds extension: var amiAction = { action: 'UpdateConfig', reload: 'yes', srcfilename: 'sip.conf', dstfilename: 'sip.conf', 'action-000000': 'newcat', 'cat-000000': '1000', 'action-000001':

Asterisk 英文名是’星号‘的意思，设计的初衷是，任何和电话系统有关的东西，它都能做，当然并不是所有和电话有关的功能靠 asterisk 实现都是最好的。下面我们简单谈几点点学习 Asterisk 必须知道的！ Asterisk 最擅长的就是做一个 ippbx ，当然有人用它来做 voicemail 服务器、 ivr 服务器、会议服务器、中继网关、 sip server 、发卡系统等等，在融合通信方面， ippbx 起到重要作用有的和呼叫中心结合，有的要和 crm 结合，有的呢，和 erp 结合，还有的和 im 结合。 Asterisk 能够支持传统的线路有： tdm （ time division multiplexing ） t1/ e1 pri/ pra & rbs （ robbed bit signal ） modes analog phone lines/ phones （ pots ） isdn （ integrated services digital network ） both bri （ basic rate ） and pri （ primary rate ） Asterisk 需要的带宽，一般为： 32kb/ 线路。也就是说每支持一条线路，只需要增 32kb 的带宽，但是需要网络质量良好。 Asterisk 支持的协议包括： session