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Allow Anonymous Inbound SIP Calls?

remo
Posts: 8
 Member Since:
2006-07-09
Submitted by remo on Mon, 08/07/2006 - 2:42pm.

Hi

I am trying to install a new trunk with guest-voip.ch. The only way to receive incoming calls is to set yes in general the option Allow Anonymous Inbound SIP Calls?

Is there any risk to change this to yes and if yes do you know another way to accept this calls without setting this parameter to yes ?

It seems that the trixbox doesn't recognise this trunk. asterisk -rvvv gives this :

-- Executing NoOp("SIP/41xxxxx99-c25d", "Received incoming SIP connection from unknown peer to 41xxxxx99") in new stack
-- Executing Set("SIP/41xxxxx99-c25d", "DID=41xxxxx99") in new stack
-- Executing Goto("SIP/41xxxxx99-c25d", "s|1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing GotoIf("SIP/41xxxxx99-c25d", "0?from-trunk|41xxxxx99|1") in new stack
-- Executing Set("SIP/41xxxxx99-c25d", "TIMEOUT(absolute)=15") in new stack
-- Channel will hangup at 2006-07-19 06:05:37 UTC.

sip_additional.conf

register=41xxxxx99:yyyyy@voipgateway.org/41xxxxx99

[41xxxxx99]
username=41xxxxx99
type=user
secret=yyyyy
qualify=yes
nat=yes
insecure=very
host=voipgateway.org
fromuser=41xxxxx99
fromdomain=voipgateway.org

[SIPxx99]
username=41xxxxx99
type=peer
secret=yyyyy
insecure=very
host=voipgateway.org
fromdomain=voipgateway.org

Thank you for your answer.

Remo

‹ Experience with DIDWW ??? Anonymous SIP incoming calls.? ›

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philaiv
Posts: 12
 Member Since:
2006-07-02
Re: Allow Anonymous Inbound SIP Calls?
Thu, 07/20/2006 - 10:43pm

My Trixbox is set to accept anon sip calls and I've never had any problems though that doesn't mean that I won't in the future. The two problems that I can think of occuring would be telemarkters calling for free. Also, if you use an IVR you need to make sure it is secure, so that someone can't call in and then use your box to call Internationally. Hopefully this helps a little. If not the info, but maybe to attract someone who knows more, lol. Good luck

--

My Blog


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remo
Posts: 8
 Member Since:
2006-07-09
Re: Allow Anonymous Inbound SIP Calls?
Fri, 07/21/2006 - 1:40am

Thank for your answer philaiv !

So I put now this anon sip call to yes, as I don't have IVR on this machine I don't care.

Thanks to people working on this project it is a really nice software !

Remo :lol:


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n3glv
Posts: 20
 Member Since:
2006-07-31
Re: Allow Anonymous Inbound SIP Calls?
Mon, 07/31/2006 - 1:53am

Ok, Anon-sip a subject near and dear to my heart.
Risks (as I see it)
Flood calls, no "legs" limit unless you impose one on the whole box.
War dialers? Is there such a thing as calling all numbers @ an ip?

Reality, there are far more neat things that you can do that I think
outweigh the so called "risks"

ENUM is way cool (www.e164.org)
You can register ANY number you can recieve an inbound call on.
My cell number is registered on ENUM, if you dial 10 digits you
get my cell, if you dial (on a box with enum trunk) 11 digits you
get my * box!
Another neat trick is that you can then do pattern matching
on inbound anon calls by DID. IE I have a pattern for 888
that when someone dials sip/888@my.ip.or.dyndns.addy
It will plug them into my MEET_ME conf room.
Also, ENUM gives out 100 "numbers" starting with 88299
to each registrant. These numbers can be used between
servers and even via the sip-broker free pstn indial numbers
to call all kinds of services on your boxes. (one of mine go's
to my conf room for example, another to my pc)
Go to add trunk, add an ENUM and then put it as your first trunk
for all calls, if you want to dial the enum specific ones, add an
outbound route to dialplan to match 88299. and send to enum trunk. (must have the period)
The lookups fail from time to time, but as it grows it's got to get
better.
Look me up on the white pages and ring me up guys!


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hoolahoous
Posts: 59
 Member Since:
2006-06-02
Re: Allow Anonymous Inbound SIP Calls?
Tue, 08/01/2006 - 3:26pm

can you please detail, how you do the pattern matching on inbound anon calls by did ?
I was searching but couldn't find relevent info.


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n3glv
Posts: 20
 Member Since:
2006-07-31
Re: Allow Anonymous Inbound SIP Calls?
Wed, 08/02/2006 - 9:59am

Yeah, doccumentation is a little thin on this and also on the whole
ENUM subject. I think they may feel it's so simple it should be
obvious but it's not to everyone.
Once you have registered for your block of numbers or if you
just want to filter (route) by DID, just go to Inbound Routes.
Fill in top box (DID) with any pattern you want to match such
as 8829900373943 wich will go to my echotest (hence the 43 on
the end, just the one I picked for this job). Then direct the destination at the bottom to whatever extension or service.
I have a series of them set.
8829900373900 IVR
8829900373901 Chat Room
8829900373943 Echo Test
8829900373999 ME

Look for me on IRC if you have any more questions or want to test.
(Can always just call me from above list)


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hoolahoous
Posts: 59
 Member Since:
2006-06-02
Re: Allow Anonymous Inbound SIP Calls?
Thu, 08/03/2006 - 1:38pm

i was wondering if we can give wildcards etc. in pattern matching. the patterns you have put are exact numbers. Problem with my setup is that number i get from provider is of 1234567890-xxxx where xxxx keeps changing everytime.. and these xxxx are alphanumberic (not just numeric)


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dajo
Posts: 3
 Member Since:
2006-08-07
Re: Allow Anonymous Inbound SIP Calls?
Mon, 08/07/2006 - 2:42pm

@Remo

This may help you:
http://www.guest-voip.ch/VoIP_Client_Setup/trixbox_asterisk_at_ho...

Daniel


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dean.collins
Posts: 139
 Member Since:
2006-06-01
german?
Tue, 05/26/2009 - 11:10am

no that link wont help you as it's in german - does anyone have an answer to this? i'd like to only allow sip calls from 2 defined SIp trunk providers and am having troubles trying to find the answer to this.

--

Cheers,
Dean Collins
www.Cognation.net
Delivering Your Solutions Now.


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b14ck
Posts: 773
 Member Since:
2009-03-03
deanc, You can simply
Tue, 05/26/2009 - 1:39pm

deanc,

You can simply register both of your SIP trunks, then turn allow anonymous calls to no, and route by DID accordingly (default trixbox behavior). Once your trunk is registered, place a call to one of the DIDs you wish to route to a certain destination, and watch the asterisk console, you will see the incoming number (as sent to your PBX by your carrier), and can then create an inbound route for that DID. Also--you may want to create a new thread as this thread is not related to your post. Thanks!

--

Randall Degges
Lead Developer, RCI Telecommunications
projectb14ck - http://projectb14ck.org/ - Weblog


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colinjack
Posts: 334
 Member Since:
2006-06-01
That doesn't work (unless
Thu, 07/09/2009 - 8:55am

That doesn't work (unless I'm missing something) - which I might be!

Our provider - Entacall in the UK - sends DIDs and I have set up inbound routes that route the DIDs correctly. Turn off 'Allow anonymous inbound SIP' and I get 'this number is not in service' even though it is hitting the Trixbox okay and I can see the incoming DID on the CLI. I have got around this by using my firewall to limit inbound SIP to only the IP addresses used by Entacall.

What am I missing do you reckon? I know my way around TB/FPBX pretty well after 4 years.

Colin

--

Colin


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SkykingOH
Posts: 9680
 Member Since:
2007-12-17
If you need to turn on
Thu, 07/09/2009 - 9:19am

If you need to turn on anonymous SIP then you don't have your trunk configured properly.

In the log you will see 'no matching peer found'

Anonymous SIP simply accepts calls from any peer.

--

Scott

aka "Skyking"


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colinjack
Posts: 334
 Member Since:
2006-06-01
Always a good chance I've
Thu, 07/09/2009 - 9:42am

Always a good chance I've got it wrong :)

username=44xxxxxxxxx
type=friend
secret=xxxxxx
qualify=no
insecure=very
host=proxy.entacall.com
fromdomain=proxy.entacall.com
disallow=all
context=custom-get-did-from-sip
canreinvite=yes
allow=ulaw&alaw

What do you reckon?

Colin

--

Colin


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SkykingOH
Posts: 9680
 Member Since:
2007-12-17
Quote: What do you
Thu, 07/09/2009 - 10:14am
Quote:
What do you reckon?

I "reckon" that something doesn't match what the provider is sending you. You have way to many entries to start with. Kill the fromdomain and check the SIP debug and make sure the call is really arriving from proxy.entacall.com.

Also you have it pointed to a non-existent context. Context should be "from-trunk"

--

Scott

aka "Skyking"


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colinjack
Posts: 334
 Member Since:
2006-06-01
You assume I have it pointed
Thu, 07/09/2009 - 11:06am

You assume I have it pointed to a none existent context. You are wrong. :)
That is a custom context that I wrote to correct the CID so that it matches the records in the MySQL database I created for our Cisco directory.
The 'fromdomain' is required by entacall to avoid spoofing.

Thanks for your response anyway.

regards

Colin

--

Colin


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SkykingOH
Posts: 9680
 Member Since:
2007-12-17
Quote: You are wrong.
Thu, 07/09/2009 - 12:19pm
Quote:
You are wrong. :)

It's certainly not the first time!

I still stand that the peer is not matching. Are the calls hitting your custom context? That would be a good inidicator that the trunk is working and would not require anonymous SIP.

--

Scott

aka "Skyking"


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colinjack
Posts: 334
 Member Since:
2006-06-01
You are correct ... after
Thu, 07/23/2009 - 11:56am

You are correct ... after discussing it with my provider it worked out that they use 6 media servers to deliver calls.
I needed to create a trunk for each media server with a matching 'host=' entry for each media server IP.

Colin

--

Colin


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SkykingOH
Posts: 9680
 Member Since:
2007-12-17
The example configs for 1.6
Thu, 07/23/2009 - 12:45pm

The example configs for 1.6 show concatenated host names for SIP peers IE: host=1.1.1.1&1.1.1.2&1.1.1.3

I hope this works as expected it would be slick.

--

Scott

aka "Skyking"


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colinjack
Posts: 334
 Member Since:
2006-06-01
Absolutely - can't
Mon, 07/27/2009 - 3:45am

Absolutely - can't understand why it wasn't available before (or to be able to use subnets like host=84.44.88.0/28).

--

Colin


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conradical
Posts: 13
 Member Since:
2007-12-13
Has anyone tested the
Tue, 10/13/2009 - 6:50pm

Has anyone tested the concatenated hostname for sip peers yet?


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obeliks
Posts: 878
 Member Since:
2010-03-14
"concatenated hostnames"
Tue, 06/29/2010 - 7:21am

"concatenated hostnames" sounds like an urban legend at this point. I have not seen anything in chan_sip.c that could allow this.


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