Its my first post. Kindly guide me with your expertise.
I have installed asterisk 1.4 successfully. I started asterisk and as described in linux cookbook by carla, i entered the following command
console dial 1000
It played the greeting. I can hear the voice but its very very jittery. Not of good quality. Can someone tell me what might be the issue ?
thanks alot in advance
Trying to run it on a virtual machine.
An overloaded network.
Running X applications at the same time.
I didnt ran any x applications. So you mean to say, its because i ran it on virtual machine ? will it work fine if i install it on a p4 system ?
It should work better, but I don’t think the console device has jitter buffering by default, so it may still be subject to network jitter. I think you may be able to turn on jitter buffering, but, in general, the console device is not intended for serious use, except possible to driver a paging speaker. For a fair test, you should use two SIP phones.
It was installed on my system. How can network play a role when i was listening to it from directly the machine it was installed on ?
Sorry. I had forgotten the details of the sample configuration. If you are just listening to the internal recordings, rather than making the test call to Digium, or to a local SIP phone, the problem is either due to running on avirtual machine on a loaded hsot, or because you haven’t got a properly configured timing source (I’m not sure, but the console device may provide a good enough timing source).
Please search on timing sources, as the options are changing at the moment. for 1.6, you would generally want dahdi_dummy and to have internal timing enabled in asterisk.conf, but as noted, the console device might not need this.
Note 1.4 is not reccommended for new installations. 1.8.x is the stable series with the longest support.
Actually i installed 1.4 since it was mentioned in several books.
Can you kindly guide me to installation procedures of asterisk 1.8 ?
Basicly yes, But first of all test it with a sip softphone and a trunk of some type.
Asterisk can run just fine on virtual machine , But how good the console to sound card driver will be under vm is another question.
Basicly test in a way that is realistic IE use a handset or softphone.
I think the ability to run Asterisk on a VM depends on what else is on the VM. If the host is configured to give enough resources to the Asterisk VM, you may be OK, especially with real SIP phones, which will have good jitter buffering.
If it is heavily loaded, you will get broken up speech, even with real SIP phones. We get this problem when we try to development test on VMs. Hopefully a production VM would be on a system that was carefully dimensioned.
The other risk with development testing on a VM is that you are much more likely to only gain access to only one processor core, so you are much less likely to find one of the many race conditions in Asterisk.
Thanks alot guys.
I am installing asterisk 1.6 for now. I will install it on my p4 system in this week