SIP Trunk for Keku.com

Trying to configure keku.com as a SIP trunk, and not getting much luck.

The trunk I’ve setup is:

disallow=all
username=317XXXXXXX
type=friend
secret=xxxxxxx
qualify=no
insecure=port,invite
host=sip.keku.com
dtmfmode=rfc2833
context=from-pstn
canreinvite=no
allow=ulaw&alaw
echocancel=yes

Register String: 317XXXXXX:[email protected]

The trunk registers fine… CLI shows Registered status. However, dialing any number, I get a recording saying “all circuits are busy now”… CLI shows the following excerpt:

Called SIP/Keku/0033123123123
[2014-08-31 15:20:09] WARNING[32542][C-00000001]: chan_sip.c:23031 handle_response_invite: Received response: “Forbidden” from ‘“701” sip:[email protected];tag=as15c3a45d’
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [[email protected]:21] NoOp(“SIP/701-00000000”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21”) in new stack

If I configure a softphone SIP client, the trunk works fine. Any ideas what I’m doing wrong?.

Is 701 an extension on your FreePBX? Maybe try outbound CID Override on the SIP trunk, if 701 IS your test extension number. Just a thought.

701 is a working extension in my system, and it works fine with a variety of other trunks (Google Voice, CallWithUs among others) - thanks for the suggestion.

In any case, I kept trying and this trunk works

username=317XXXX
type=peer&friend
secret=xxxxx
qualify=yes
nat=yes
insecure=very
host=sip.keku.com
fromuser=317XXXXXX
sendrpid=yes
fromdomain=sip.keku.com
disallow=all
authuser=317XXXX
allow=ulaw&alaw&g729

Do you have a g729 license. Try removing it from your trunk to test.

You don’t have a context specified. Also you have authentication parameters then insecure.

Sounds like you just put everything you could find together, You would never need fromdomain and fromuser together. One or the other.

Alan, yes, I have a g729 license - so that not a problem.

Sky King: you are absolutely correct… I have taken a couple of my working trunks (CallWithUs, VoipTalk etc), and first just changed parameters like username, domain and password… That didnt work, then I tried merging … Trial n error… Eventually, the trunk details I posted in my second post actually started working!!.

Riz

I have a feeling that the “fromuser=317XXXXXX” did your trick, as an override on the outbound route or trunk would have also done. Your SIP carrier is not authorizing you to send calls from an unknown outbound CID.