I have got asterisk 1.12.1 installed on my server, which until recently was working perfectly. Then for some reason I can’t understand all calls made over my VoIP supplier are all choppy, almost inaudible. I have been told by the receiver that it is fine on their end.
I’m not sure how to go about debugging this issue. My internal network can handle the load as I have six phones going through two ATAs. I have also tested that my internet connection can handle the load by setting an ATA to the VoIP supplier and I get perfect quality again.
I have tried reinvite as yes and no, fiddled with almost every codec setting, only alaw and ulaw work at all.
It’s not that the call isn’t going through, just that it’s all broken. I’ve tried Googling for an answer, but as of yet I’ve found no one with the same problem. Fact is searching for “asterisk choppy sip” isn’t a very good search string.
I would appreciate any help possible. The reason I have not upgraded to 1.13 is that the system is working for the PSTN and there seems to be no fixes related to my issure in the ChangeLog, therefore I did not want to risk an upgrade and messing up everything else.
is the asterisk server on the LAN? is it connected directly to the internet or does it go through a firewall? are you connecting to your ITSP with SIP or IAX?
what does ‘uptime’ tell you the 24 hour and 5 minute load averages are?
more details are needed i think.
[quote=“aie93”]I have got asterisk 1.12.1 installed on my server, which until recently was working perfectly. Then for some reason I can’t understand all calls made over my VoIP supplier are all choppy, almost inaudible. I have been told by the receiver that it is fine on their end.
I’m not sure how to go about debugging this issue. My internal network can handle the load as I have six phones going through two ATAs. I have also tested that my internet connection can handle the load by setting an ATA to the VoIP supplier and I get perfect quality again.
I have tried reinvite as yes and no, fiddled with almost every codec setting, only alaw and ulaw work at all.
It’s not that the call isn’t going through, just that it’s all broken. I’ve tried Googling for an answer, but as of yet I’ve found no one with the same problem. Fact is searching for “asterisk choppy sip” isn’t a very good search string.
I would appreciate any help possible. The reason I have not upgraded to 1.13 is that the system is working for the PSTN and there seems to be no fixes related to my issure in the ChangeLog, therefore I did not want to risk an upgrade and messing up everything else.
The server is on the LAN, i did mean 1.2.12. It is in the DMZ on my network, incidentaly the ATA which I tested the SIP provider with was not, and worked perfectly.
I am connecting through SIP to my supplier and uptime reports ‘load average: 0.00, 0.03, 0.00’
Upgraded to asterisk 1.2.13 from source and the latest FreeBX and all seems to be working fine again. Only problem is that when the SIP call is answered on the other end it takes a while for it to be passed through, any ideas why that might be?