Send a SIP Invite to

In the Data API tab a SIP request is explained as:

Requires IP Authentication to be setup through the portal and associated with LRN under Telephone Data Tab

Send a SIP Invite to with the number you wish to dip on port 5060

this is done in sip.conf?

I was unable to open the api link, it is asking for a password. So I’m not sure about what do they request you to do. Any way you can send an INVITE dialing to SIP URI, this is done through /etc/asterisk/extensions.conf . If you want to set as a SIP peer you will need to configure it on /etc/asterisk/sip.conf , or you will need to enable guest calls (usally enable by default) and configure the context for guest calls

Is this describing p2p SIP? as:

If so, how does sending an invite relate to “trunking”? As I recall from the O’Reilly book, there’s really no such thing as trunking in SIP, it’s just a short-hand. The wikipedia entry on sip trunking lists this resource:

Alternately, customers connecting individual handsets to our cloud hosted voice platform will use Registration. They simply enter a username, password and SIPcity proxy address ( on each handset.

I have a few different SIP accounts registered on a Cisco IP phone. Yes, I’m interested in Asterisk, but, for the moment, only want to “use Registration” as in the above sipcity usage:

`INVITE sip: SIP/2.0
Via: SIP/2.0/UDP;branch=z9hG4bK-d8754z- 0d7929dc4684a058-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip: 61881220301@>
To: <sip:>
From: "61399815701"<>;tag=56566ed5
Call-ID: MWM2Nzk3Nzk2MzgzOWY1NDlmNTgxYTE0N2FhZDc2ODg

Effectively, can this invite only be sent through Asterisk? Or, can a softphone, or hardphone, send this invite? Perhaps sipsak from the CLI?

An INVITE is a request to set up a session, commonly referred to as a call. Anything supporting SIP to establish calls uses INVITE to do so. It’s equivalent to picking up the phone and dialing a number.

In the case of chan_sip the sip.conf file is used to configure the settings for making the call, and extensions.conf is the logic which describes when to place an outgoing call when we get an incoming call.

Whether you must use Asterisk is really up to your experience, what you are trying to do, and whether other things can do what you need.

Can you elaborate on how it’s done through extensions.conf?


exten    =>    _1NXXNXXXXXX,1,NoOp()
exten    =>    _1NXXNXXXXXX,n,Set(CALLERID(name)=opm)
exten    =>    _1NXXNXXXXXX,n,Log(NOTICE,Dialing out from ${CALLERID(all)} to ${EXTEN:1} through ${TOLL})
exten    =>    _1NXXNXXXXXX,n,Dial(${TOLL}/${EXTEN:1},60)
exten    =>    _1NXXNXXXXXX,n,Playtones(congestion)
exten    =>    _1NXXNXXXXXX,n,Hangup()

so that peer TELNYX from sip.conf is used in the outbound context. Sending an INVITE would be dialing through the TELNYX peer? This peer doesn’t use but has instead.