Do anybody have any advise how to setup DIDWW on Trixbox? I am using their TRY account and are not able to make it happen.
Q1: What are the settings on DIDWW? Are you using SIP or IAX?
(example: IAX 5000:5000@my_trix_ip (5000 is extension and password)
Q2: What are the trunk settings in FreePBX? Their support page is quite cryptic.
Q3: Anything else I am missing? I am getting only "Your call did not go through - message"
Thanks!
- Klaus
Did you ever get this working?
Mine is working great,
I created a trunk with Incoming settings:
allow=g729&ulaw
canreinvite=no
context=ext-did
disallow=all
dtmfmode=rfc2833
host=66.54.140.46
insecure=very
nat=no
type=peer
Then I created an Inbound Route, and put the phone number as the DID and any CID
and pointed to an IVR
I did not create an inbound trunk for this. I did enable the anonymous sip trunk and I change the context in sip.conf to "from-trunk". Then I went to DIDWW and entered the URI for my Trixbox, which in my case is;
SIP/qth.cohutta.org/16782794442
Make sure you create an inbound route for the DID (in my case 16782794442) and route it where you like.
JMullinix,
Thanks for the pointers. I now get the trunk to ring into the server, however, after 2 rings it returns busy signal to callers.
When I look in the logs it shows the call coming in but not knowing what to do with it. I have created an inbound route for it to ring on IVR but it doesn't get to that point.
Any suggestions anyone?
Thanks.
I didn't have to allow anonymous or change the context. I just created an inbound route with the DID,
One thing you can do is a sip debug and you are going to see some important information regarding the way didww is sending you the information for your did.
Create an any/any inbound route and see if you can answer it then.
Well that seems to have solved the problem of being answered by TRIXBOX. The calls are now answered by the IVR. However, I want to be able to direct a specific DID to a specific extension not to the IVR. Any suggestions on how to fix that issue?
I did have an inbound route created with 1416342XXXX and I had it routed to a specific extension on the TRIXBOX however, it would ring twice and then return busy.
Any suggestions on where to go next with this? DIDWW don't seem to be too helpful in troubleshooting with the user.
Thanks.
Well, a couple of thing to check would be:
1. Make sure the SIP URI at DIDWW is in the form "sip/
2. Make sure you allow anonymous SIP and change the context in the SIP.conf for the anonymouse sip to "from-trunk".
3. Set up an inbound route matching
4. Check you general settings and make sure you have the default ring time set to a reasonable amount, such as 30 seconds.
5. I would put a voicemail box on the extension for testing.
Let me know if this helps.
hi there,
My DIDww account with US not working.
SIP.conf:[didww]
host=66.54.140.46
dtmfmode=rfc2833
type=peer
context=from-didww
insecure=very
nat=no
canreinvite=no
qualify=yes
allow=g729
allow=ulaw
disallow=all
extensions.conf:
[from-didww]
exten=>416342XXXX,1,Goto(company,209,1)
When I dial the number, it rings continuously at caller side but no ring at the called extension.
Please help me on this.
Thanks,
Amit
Amit,
What do you see when you watch the command line as a call comes in.
I set the mapping at DIDWW to:
SIP/"my domain name"/"My DID"
I enabled anonymous SIP. I edited the sip.conf and changed the context for anonymous SIP calls from
context=from-sip-external
to
context=from-trunk
Then I set up an inbound route for "My DID".
Make sure you have the correct ports forwarded and the proper externip and localnet directives in your SIP.conf file.
Hope this helps.
John
http://www.cohutta.com
1-706-632-3343 - E164 friendly (Use your friendly ENUM trunk today.)
Dial Plan helper http://www.cohutta.com/npanxx.php
Would anybody please elaborate on the security concerns with setting YES to receiving anonymous SIP calls? I too use DIDWW and was wondering if there was a way NOT to do so... for "security reasons"?
This is old and very bad advice from an operations stand point.
First - anonymous SIP is not a security concern. Some folks don't like the random calls however it does not allow anonymous peers to access your system, only call you.
Second - You can setup a very simple trunk with minimal setting that will qualify the peer. Capture sip debug's from inbound calls. You can use the IP of the UA or perhaps the domain to match on in the peer settings.
A complete list of Asterisk peer settings is at this link: http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
Scott
aka "Skyking"
anonymous SIP is not a security concern
Not according to this: http://www.trixbox.org/forums/trixbox-forums/open-discussion/aste...
Member Since:
2006-12-16