Make Outgoing Calls Through BroadWorks/BroadSoft SIP Gateway From Asterisk

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I’ve been given a SIP hard phone pre-configured to work with another party’s BroadWorks system. I want to use my Asterisk system to connect to this SIP service rather than the handset I’ve been given. I have extracted the authentication details from the phone and have successfully registered Asterisk with the gateway (incoming calls work fine) using a line like this in sip.conf:

register => 441000123123@auth.realm:password:authuser@nnn.nnn.nnn.nnn/441000123123

Outgoing calls are proving to be more challenging. I have this so far:

callerid=”My Name” <441000123123>
type=peer host=nnn.nnn.nnn.nnn auth=authuser realm=auth.realm fromuser= 441000123123
secret=password insecure=invite context=from-sip nat=yes qualify=no canreinvite=no allow=all

The main part I’m confused about is that in most examples I’ve seen, the username and authname are the same value, whereas in this case we seem to have:

A username (the phone number)
An auth name (a different value)
An auth realm A SIP realm (different value to auth realm)
A password A gateway host

When I place a call with Dial(SIP/441000123123/somenumber), I get a 403 response from the gateway. Looking at a packet dump, I can see that Asterisk is not attempting to authenticate. On the other hand, REGISTER requests do authenticate successfully – I can see the digest authentication taking place in tcpdump.

I have observed successful outgoing calls from the hard phone using tcpdump and I can see the phone using digest like so:

Authorization: DIGEST username=”authuser”, realm=”BroadWorks”, nonce=”BroadWorksASHORTHASH”, qop=auth, cnonce=”ASHORTHASH”, nc000001, uri=”sip:number@auth.realm:5060;user=phone”, response=”ALONGERHASH”, algorithm=MD5

What is the correct configuration to use – how do I get Asterisk to successfully authenticate outgoing calls?

Many thanks, James.

One thought on - Make Outgoing Calls Through BroadWorks/BroadSoft SIP Gateway From Asterisk

  • I have answered my own question.

    The remote host is reachable only by IP address. Setting host=nnn.nnn.nnn.nnn causes Asterisk to send INVITEs to somenumber@nnn.nnn.nnn.nnn instead of some number@sip.domain. This is what was causing the 403 response, it’s not necessary to authenticate.

    I haven’t found a way to set the sip domain to be used for outgoing calls, so as a workaround I have inserted ‘nnn.nnn.nnn.nnn sip.domain’ into my /etc/hosts file. Not elegant, but it works.