We have a asterisk VOIP system. We are maintaing 4000minuits per day.
Some times we are facing an issue with Duration. I am usind Dial command to make an outgoing call.
If we tested one after the another all calls gets disconnected properly. When we hava Huge traffic sometimes the call may not hangup properly or asterisk not receving Hangup request or asterisk not disconnecting properly.
Because of this reason we are getting craze calldurations i mean ANSWEREDTIME.
Hung channels can be the result of any one of a number of contributing factors. My suggestion to you would be to try an Asterisk version change/upgrade as the simplest means to of addressing this issue. Without log samples and detailed information about the environment, this issue cannot be addressed otherwise.
if i use rtptimeout then will it terminate the channels or session if detects silence on the channel
We are usin our application for Making calls on credit basis.
we are facing an issue when the channel is not hung up or hung up channels are not detcted by asterisk then the call duration between A and B parties keep on adding. even they are not on call…
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
; on the audio channel
; when we’re not on hold. This is to be able to hangup
; a call in the case of a phone disappearing from the net,
; like a powerloss or grandma tripping over a cable.
What is the Best configuration to avoid these type of issues…
what could be the reason for " The asteisk restarts unexpectedly…" If needed i can provide you the log of last asterisk restarted…We are facing the issues more when asterisk disconnect unexpectedly…
you seem to have two issues…
1/ your calls ending while you dont detect. Add rtptimeout and see if it improves…
2/ security… you’ve now anwsered my question:
[quote]But we will get lot of unwanted requests some times at that time asterisk behaves abnomally…
if it is spam / scanning requests (ie from ips that have nothing to do your box), then you need to secure your box, iptables + fail2ban is a good start.
DID calls (as generally understood by Asterisk SIP users**) are SIP calls; your last posting does not make sense.
allowguest controls the handling of calls for which there is no matching entry in sip.conf. The way that people normally use SIP means that there should be no such calls on a properly configured system.
(**In most cases, so called DID callers are only DID to the service provider, with only one number being passed to the PABX. Traditional DID is direct in dialling to the customer, with a range of numbers being routed to the same line, with routing digits to allow the PABX to distinguish between them.)
We have lots of DID’s which are called by users to provider, Provider inturn route to asterisk to reach our PBX system then IVR then they select contact from IVR. PBX will dial out to destination…
Here when i add allowguest =no in sip.conf file asterisk not allowing the DID calls routed from provider which is an incoming call to provider from customer…
Who pays for these calls? With allowguest, anyone can call in and make a call through your IVR.
allowguest = yes is only really suitable for use behind a firewall that completely blocks direct external access, or when the default context is totally secure and cannot incur any outgoing call costs. The latter case is only really applicable if you are operating in a full SIP peer to peer world, rather than one that uses PSTN numbers as the user part.