We have a SIP trunk in our company. This SIP trunk is connected to Panasonic PBX and the PBX routes the calls to the extensions. Now we need a passive call recording server. The only task that this server should do is recording all the incoming and outgoing calls of the sip trunk and SHOULD NOT answer any call. So can we use asterisk as a recording server? If not what are other solutions?
If you can connect it like panasonic->e1 card->asterisk->incoming/outgoing sip or panasonic ->sip(if your panasonic have it)->asterisk->incoming/outgoing sip - then answer is yes.
If you want it use like panasonic recording card - answer is no.
Related
I want to program an esp8266 doorbell to call me when someone presses the bell. I have a STARFACE telephone system (Asterisk) and would like to tell STARFACE to make a broadcast call. I have searched the Internet but I find only FritzBox examples.
I do not want to do this with a call file.
Sorry for my English. I am not a native Englishman.
Call file is simplest way do that.
Some other ways
asterisk AMI protocol Originate command
asterisk ARI
perl,sipp(testing tool) or other script which send sip invite with auth.
https://gist.github.com/maximevalette/802764
http://sipp.sourceforge.net/
click2call script on asterisk (using call file or other)+ curl request on your device.
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)
How can I setup asterisk to dialog with sip devices using
the Multicast transmission protocol ?
Basically I have an asterisk box conected to a VSAT network.
On the other end I have a SIP box in a network receiving
the signal from the other VSAT.
The return from the SIP box to asterisk is unicast.
Is it possible to make it work ?
Asterisk SIP currently not support multicast.
Multicast rtp supported.
http://www.voip-info.org/wiki/view/Asterisk+MulticastRTP+channels
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.