sip

How to notify SIP client when there is an incoming call on another phone

淺唱寂寞╮ 提交于 2019-12-11 06:26:24
问题 is it possible to notify a SIP client when there is an incoming call on another phone? I know that there are the SUBSCRIBE and NOTIFY commands but I have found no event package for signaling incoming calls. Background: for a SIP-capable telephony system, I would like to provide an application that displays information about the caller (e.g. name, address, contracts, etc.) when the phone rings. The phones are external to the PC; they are not soft-phones. -Frank 回答1: The dialog event package

Astrisk+SIP+Database

强颜欢笑 提交于 2019-12-11 06:24:17
问题 Can I use SIP account from database? As I understand. I can config SIP account in sip.conf. But I want to support use many account. Can I get SIP account from database that I created for dial out instead of sip.conf? Or Other way I can do it. 回答1: What you are looking for is called 'RealTime' in Asterisk (yes, it doesn't make much sense ;) ). http://www.voip-info.org/wiki/view/Asterisk+RealTime http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip 来源: https://stackoverflow.com/questions

SipManager cannot create SipService. Cannot Bind Context.SIP_SERVICE in Nougat

别来无恙 提交于 2019-12-11 05:11:51
问题 I have an application that uses SipManager to open a SIP profile. I have no difficulties opening the SIP profile on my older(KitKat) device, but my new(Nougat) device throws the a SipException "SipService is dead and is restarting..." Stepping through the SipManager reveals that 'createSipService' is not binding 'Context.SIP_SERVICE'. I found this post, which seems to be the same issue. Does anyone know what changed between KitKat and Nougat that would cause such an error, and what I can do

Freeswitch India Progress tones

安稳与你 提交于 2019-12-11 04:57:18
问题 I'm using Freeswitch 1.6 to detect Progress tones. I need to configure India but I'm unable to find correct test numbers to validate Ringback/Busy/Reorder tones, looking for existing working settings. Using spandsp.conf.xml module. I have US, MX, BR working just fine. I looked into ITU-T standard and Asterisk sample configuration: [in] description = India ringcadence = 400,200,400,2000 dial = 400*25 busy = 400/750,0/750 ring = 400*25/400,0/200,400*25/400,0/2000 congestion = 400/250,0/250

Android SIP Client

时光总嘲笑我的痴心妄想 提交于 2019-12-11 04:45:32
问题 Which is the better way to start the SIP client for android using any external libraries.Since the native inbuilt SIP libraries work only with versions greater than 2.3.1.Looking for the better and easier solution to build the SIP Client for lower versions of android.Could any one help with this... 回答1: To start implementing software SIP Client, you would need 3rd party SIP implementation stack. Check out this thread for a list of sip implementation stacks. What to look for: evaluate the

Binder cannot bind Context.SIP_SERVICE - SIP Client not working on Nougat

狂风中的少年 提交于 2019-12-11 04:24:49
问题 I've implemented SIP feature in my app. I am currently working on Galaxy Tab SM-T580 with Nougat. Problem started at the moment when I've decided to upgrade Android (from 6.0 to mentioned 7.0) - then my SIP client stopped working. When I try to open my sipManager implementation with open(sipProfile, pi, null) method android.net.sip.SipException: SipService is dead and is restarting... at android.net.sip.SipManager.checkSipServiceConnection(SipManager.java:183) is thrown. Debugging code I can

Asterisk how check if a number is busy before dialing it

廉价感情. 提交于 2019-12-11 04:21:49
问题 As a beginner that just installed his first asterix server I came across a small issue. i want to check if a number that i dial is busy or not before actually dialing it. So for example I would call 0904 => number busy => redirect to 0905 However I don't want it to dial 0904 first if its busy but just instantly redirect it to 0905 instead. I have a very basic setup with just 3 users. 2 of them are in a queue "support" one of them is in the queue "admin" I have seen and read a little about

Any good SIP library for C#?

孤人 提交于 2019-12-11 02:14:08
问题 Does anyone know if there is any good library around that can be used to easily build a SIP softphone? Thanks in advance, Cheers, Gianluca. 回答1: Konnetic provide fully managed SIP (and MSRP if you want messaging) components for .NET development. Their .NET SIP SDK is probably the most comprehensive and is aimed at clients. Otherwise, Microsoft's Lync server comes with a very good managed SIP library, available here: http://www.microsoft.com/en-us/lync/default.aspx Although you are tied to

File transfer using SIP

守給你的承諾、 提交于 2019-12-11 02:04:46
问题 The question is - is there any way to transfer files using my sip provider - I'd like to make an android application - sip client with the only function to send files. You enter your sip account information, number to deal and choose the file to send. You deal your friend, he answers and file transfer begins. The files can be any format. I read a lot of information but didn't find the way to do the project. Any ideas? 回答1: You probably should have went with jabber instead of SIP. Anyway,

Internet explorer and google chrome frame can support webRTC?

微笑、不失礼 提交于 2019-12-11 01:01:40
问题 I have tried and tested various approaches to make webRTC work on internet explorer using Google chrome plugin webRTC4all Sipml5 is not responding to either of these approaches I also read the "Customizable, Ubiquitous Real Time Communication over the Web (CU-RTC-Web) Real-Time Media and Peer-to-Peer Transport API " document which is still in development stage and not version is released . Could someone tell me a solution to the problem , or correct me if i am wrong . 回答1: Okay i am going to