As far as I know, there are many applications using SIP : ekiga, linphone, even skype, … Are those device all able to work with each other? I mean, if I register with, say, linphone, will someone on skype be able to ring me?
Skype is not using SIP. But if you interconnect any SIP system then there are plugin available like SipTheeSkype, SIP to Skype Gateway, Skype with asterisk & some other.
By using that plugin you can interact with Skype network from your SIP network.
As far as concern to other SIP client they are all interact with each other if they don't have any proprietary header check to register with specific server only.
Related
We seek your help in implementing bluetooth on our smart home controller. Our current scenarios is described below:
We want secure communication between our embedded linux controller(our smart home hub) and user's mobile phones running android/iOS.Our system is in home automation.
The hub in turn communicates with home devices that support a different protocol - we need to make communication between hub and mobile devices using bluetooth.
Currently there is not direct communication between our hub and mobile phone except through the internet.
We will use Bluetooth 4.x, not 5 for hardware support on the controller.
We are using nodejs on our controller and standard bluethooth OS libraries on android/iOS. We are open to suggestions.
Our main concern is to establish secure communication between phone and the hub. With this regard we are looking for implementation guidance on pairing and communicating securely. We have been able to establish unsecure communication - exposing characteristics and services on the hub. Our understanding of the bluetooth protocol is limited and we have only used libraries available for users.
Please provide specific pointers as to what we should do and what is possible.
I am new in Asterisk field. I am facing a situation.
I have 10 sip accounts and 20 clients (softphones), so how to dynamic allocate those sip account to those clients?
Is there any best practices in the case?
Thanks
Best practice is just add 10 more sip phones.
All other variants are adminstrative like not allow user X register fomr 10am to 10pm and have no any relation to asterisk.
Please note, sip device != extension. You can have more sip devices then extensions if your dialplan support that.
I've Built a VOIP Network for my House using Asterisk as server and SIP softphone as client. Everthink is going good and i can call all SIP client of my VOIP Network.
Now I have no idea how to call an extern mobile phone or analog phone .
I've heard something about Gateway to access to another network.
any hehp woulb be appreciated.
You have to buy SIP trunking providers service for that. Find out all the sip trunking providers who have voip termination in your country. You have to create SIP trunk into your asterisk server and call mobile phone and analog phones through their trunk. Alternatively you can also buy digium PRI cards and configure your own T1, E1 PRI. You have to buy T1,E1 PRI service from Telco operators such as in India there are Airtel, Reliance who is providing PRI service.
Some of the SIP trunking providers are such as
Callbox and Rapidvox and Twilio
As far as I know, for this purpose, you need a VoIP GSM gateway, or an ATA device or a VoIP Service provider connection. As you are interested in VoIP GSM gateways, you will need a device like Cisco SPA3102 VoIP phone adapter.
The SPA3102 features the ability to connect standard telephones and
fax machines to IP-based data networks with the additional benefit of
an integrated connection for legacy telephone network hop-on, hop-off
applications. SPA3102 users will be able to leverage their broadband
phone service more than ever by automatically routing local calls from
mobile phones and land lines over to VoIP service providers and vice
versa.
(Source: Analog adapter with FXS and FXO port)
I am using csipsimple as sip client and asterisk server to set up call. Calls are made between 2 sip clients but voice is not getting transferred.
Calls are made between 2 sip clients using AMI.
I can give my asterisk cli log.
Can anybody please give me some idea to solve this issue?
Thanks
More info would be useful. First, make sure both clients are registered, and can use at least one common codec. In most cases, these aren't the problem. It's usually a NAT/Firewall issue. Are the two clients on the same subnet? Is there any firewall rules blocking the communication?
SIP signaling usually goes on udp:5060. But that seems working. Media is tricky. In each call, the ports for RTP audio changes, in the range specified in rtp.conf. This RTP traffic goes over UDP as well. By default it't 10000-20000.
If there is only routing done between the two endpoints, it should still be fine. NAT (Network Address Translation) is your main concern. Take a look at iptables, sip_nat_conntrack. To debug, use asterisk's sip set debug on command and look for the SIP headers and verify the correct IP addresses.
Hope this is the right place to ask - assuming so as it is related to programming.
I am looking for some hardware (say a dongle) that would open up an API for my computer to a SIM card. Does that exist anywhere?
The idea would be that I can then use my existing SIM card on its existing network/contract and with a bit of code, send/receive calls and texts. I figured that if a phone can be programmed to do it why can't a computer? I just need the hardware.
Any suggestions if this kind of thing is possible? Even by maybe integrating a phone?
You can use AT commands with any modem or phone, whose driver exposes a modem port when you connect it to a computer. Plug in the phone or modem, and go to Windows Device Manager, and look under "Modems" and see what you've got. From "properties" of the Modem, you can see which COM port it is on.
AT commands are an older method of communicating with a modem. There is a standard set of commands for GSM/GPRS/3G/4G devices available from 3GPP.org, here. Manufacturers add their own proprietary commands for more obscure functions. Many but not all of these are in the public domain.
Some newer plug-in modems may appear as a network adapter (you can see this in Windows), due to their drivers. In this case, you can use the following interfaces:
on Vista: NDIS, proprietary interfaces
on Windows 7 and 8.x Desktop apps: Windows Mobile Broadband API
on Windows 8.x, in the Metro/app area: Windows.Networking and MobileBroadband APIs.
on Windows 10, UWP, there are UWP APIs here. But they do not give as much functionality as the Desktop APIs.
On Windows 8.1 and Windows 10, if the appropriate drives are present it is possible to use the new MBIM interface via the desktop Mobile Broadband DeviceServices API. Mobile Broadband Interface Model spec available here.
On Linux, use AT commands via the serial port.
I'm a bit puzzled by your requirement to make (voice?) calls via a phone connected to a computer. Do you mean Skype? In this case, of course, you wouldn't need any interface for making calls, you would just open an IP connection over a data session - which can be done via any of the above interfaces.
(added this comment as an answer, as there was more information)
I use this GSM device: http://www.mikroe.com/click/gsm2/ connected to a basic UART like: https://www.sparkfun.com/products/718?gclid=CIj1xOzbur4CFUVbfgodCpQASQ
Then use AT COMMANDS.
I also use a piece of software called QNavigator to inspect the modem/gsm: http://www.soselectronic.com/?str=1329