What is needed to call with SIP URI

Most people want to be available through the PSTN and therefore purchase a E164-number with a VoIP-provider.

When I know that my correspondent is available on this telephone-number BUT also has a SIP-account registred, I would prefer calling him via SIP.

What is needed to call the SIP URI of my correspondent ??

I have a twinkle-phone and tried sip:correspondent@SIPdomain.tld but the call fails…


This assumes that your correspondent accepts unauthenticated SIP calls, and either has a SIP proxy,
or his phone running on the IP address associated
with the DNS A record for SIPdomain.tld, or that
domain has suitable DNS records to identify the
actual proxy to use.

You can also configure him as a peer in sip.conf.

See my replies to one of the other current related threads for why trying to do it directly doesn't work.

This assumes that your correspondent accepts unauthenticated SIP calls, and either has a SIP proxy,
or his phone running on the IP address associated
with the DNS A record for SIPdomain.tld, or that
domain has suitable DNS records to identify the
actual proxy to use.

You can also configure him as a peer in sip.conf.

See my replies to one of the other current related threads for why trying to do it directly doesn’t work.

[quote=“david55”]This assumes that your correspondent accepts unauthenticated SIP calls, and either has a SIP proxy, or his phone running on the IP address associated with the DNS A record for SIPdomain.tld, or that
domain has suitable DNS records to identify the actual proxy to use.[/quote]

What’s the use of a SIP URI ?
1 contact-address where several other contact-possibilities (tel, gsm, mail) are linked in a certain sequence (user preference).

sip:contact@domain.tld
means that there should be a SIP-server (proxy) within the domain domain.tld that knows a client ‘contact’.
A DNS-query for a SIP-server in the domain domain.tld means that there must be a nameserver that has a NAPTR-record for this SIP-proxy.
The IP of this SIP-proxy will be obtained and communication will go through this SIP-proxo to his registred client ‘contact’.

Am I so wrong ?

Now, when my correspondent has a SIP-account with a VoIP-provider (who has SIP-proxies installed) then I should be able to call SIP:correspondent@VoIPprovider.tld which will redirect the call to my correspondent.
The SIP-proxy knows the IP-address of my correspondent.

So I should be able to contact my correspondent via this way, rather then call the PSTN-number.

It will work if the provider cooperates. As has been said on other threads, the provider may well not accept such connections.

Also, there doesn’t have to be a proxy. If you correspondent has a DNS A record for their phone you could connect direct. Actually, if you know the instantaneous public IP address of the phone, you should be able to do a direct connect using that.

Note that MSN Messenger calls ideally end up with a direct SIP connection. The only catch is that it is actually the called party that issues the SIP INVITE; the Messenger protocol is used to request the called party to make that call.

[quote=“david55”]Also, there doesn’t have to be a proxy. If you correspondent has a DNS A record for their phone you could connect direct.[/quote]I have an Asterisk-server and couple of grandstreams and softphones…

In theory, when someone calls from his softphone “SIP:grandstream_account@my_public_ip” then they should end up at my grandstream ??
Considered that port 5060 is forwarded to my Asterisk-server…

An invite will be send to my_public_ip:5060 and end up @ my Asterisk. My Asterisk will then know, based upon the accountname, where to send the call to.
Considered that authentication is not required…