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.
Related
I found an interesting documeent about realisation of eCall (Emergency Call) in EU: http://www.heero-pilot.eu/ressource/static/files/heero_wp3_d3-3_final-operational-results_v2.3_final.pdf
Germany somehow did it on Asterisk. Whatever, I don't understand how they process MSD (minimal set of data) using Asterisk. In the call session for the first step caller sends DTMF signals to send MSD packet. As I understood, Asterisk must redirect this call to In-band modem on COM port or to another machine with such modem. After PSAP successfully received MSD for the second step caller switches to voice channel that must be redirected to some sip-client of PSAP operator. How they do it? Is there a way to receive DTMF signals w/o modem by using internal capabilities of Asterisk? How the same call to redirect to another SIP on the same time?
I suspect that you are referring to emergency services, rendered by emergency dialers - eg. for senior citizens. These are fairly common where I live, and I've created in the past a solution to handle the calls from these, based on Asterisk. The solution involved a way to intercept the various DTMF signals that the device generates, then making Asterisk do stuff with it. Back then, I used Asterisk 1.6 and it is pain staking, because I had to do everything from within a MeetMe bridge, and interact with Manager alot. Today, doing the same with Asterisk 12/13 and ARI is a breeze. Just remember one thing, most of these dialers will utilize the A,B,C,D DTMF signals, which are somewhat unknown to most people - they exist and Asterisk is very much capable of handling those.
The only snag is - make sure you are connected via a PRI, as most SIP carriers aren't aware of these signals, and their SIP trunks won't support this type of signalling.
Asterisk can send dtmf(natively,via command SendDTMF in dialplan or D option in Dial command) or any other sound(custom c/c++ app needed)
No, you not need special modem.
However there are no realisation acordinly to that document, you need do that yourself or hire someone
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.
Can a cell phone transmit SMS-CB (Short Messaging Service-Cell Broadcast) ?
If not, Can I get a device that can transmit SMS-CB messages ?
Else, Is there a good simulator that can simulate SMS-CB transmission and receiving mobile phones ?
Thank You
NOTE: Cell Broadcast (SMS-CB) is designed for simultaneous delivery of messages to multiple users in a specified area. For example, information such as Location, Tower name, Ads or Emergency messages can be transmitted.
Technically, the SMS-CB messages originate at a device called "Cell Broadcast Centre (CBC)", which is part of the network operators equipment. It sends the SMS-CB through the Base Station Controller (BSC). This cannot be done over the air, it is something which happens inside the mobile operators network. It would probably be too much to explain all GSM/3G/UMTS network components here, you might want to read up on mobile network architecture.
So the simple answer is no, a handset (mobile phone) cannot directly send SMS-CB messages.
Now the question is, how to tell the CBC to send an SMS-CB to some network cells. There exist some standardized interfaces for that, which are used for emergency alerting, e.g. the Commercial Mobile Alert System (CMAS) in the US. If these interfaces are designed sensibly, they cannot be abused by just about anyone using a mobile handset. But I would not be surprised if there were security gaps in some operator's networks which would allow unauthorized parties to send SMS-CB, e.g. via insecure Internet/SS7 gateways. But that is wild speculation. Normally, it should not be possible to send unauthorized SMS-CB from outside of the operator's network.
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.