Hi All:
I’m new to asterisk & this domain. I wanted to generate a SIP call using asterisk to outer world (any phone) without using any sip or pstn service provider. Is this possible ?
I wanted to test a conference solution . To do stress & load test i want to generate a sip call. Please plz help me and let me know at least this is possible or not ???
Regards,
heman
Hi Heman,
You will an asterisk server configure for meetme, then registering both users using expernal ip of your system you can deploy the same.
[quote=“heman4488”]
I wanted to generate a SIP call using asterisk to outer world (any phone) without using any sip or pstn service provider.[/quote]
If by “any” you mean an arbitrary phone, the answer is no.
If by “any” you mean at least one, you find a phone that is open, and put SIP/user@domain-name, or even, if suitably configured, SIP/domain-name, in your Dial application parameters. Although SIP is perfectly capable of supporting this, very few people actually operate open SIP phones or open SIP Pabxes.
You will need Asterisk to have a routable IP address, or suitable NAT configuration to be in place.
You should read up on the allowguest option, as, by default, if you have suitable routing and NAT configured, anyone else can do the same to you.