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chriserasmo

7 posts

Wannabe Geek


#28946 16-Dec-2008 21:59
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Hi guys,

I have problem with my inbound calls when I dialed my 2talk number which is 028XXXXXX using my cellphone. I heared a 2 rings on my cellphone and it goes to busy tone. The incoming calls did not get through to my asterisknow server.
My outgoing works fine now. only my inbound .. Its more than a week now since I get my 2talk until now i never used it. please help..

I am using Asterisknow. All of the configuration below is based on GUI.
What Did i missed on my configuration below.


Edit Service Provider
=======================
    Comment:  2talk
    Protocol: SIP
    Register: Yes
    Host: 2talk.co.nz
    Username: 028XXXXXX
    Password: XXXXXXXXX

    Codec Preferences
    ------------------
    Allowed: u-law & G.726
    Disallowed: GSM & a-law
    Disallow All: No (i leave it untick)

    Advanced Settings
    ------------------
    trunkname: trunk_2
    insecure: very
    port: 5060
    CallerID:
    fomdomain:
    fromuser:649974XXXX
    contact: friend
    can reinvite: No (I leave it untick)


Calling Rules
=============================
    Rule Name: 2talk
    Place this call through: Custom-2talk
    Dialing Rules: if the number begins with [02889] and
    followed by [4] digit [*] or more

    Strip [ ] digit from the front and prepend [ ] before dialing


Incoming Calls Rules
===============================
    Route [ incoming call that match ] pattern [ _028XXXXXX ]
    from provider [ Custom - 2talk ]
    to extension [ 6000-chris ]

    Route [ All Unmatched incoming calls ]
    from provider [ Custom-2talk ]
    to extension [6000-chris]

Regards,
Chris

[Moderator (ND): Removed phone numbers to prevent spamming]

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meesham
972 posts

Ultimate Geek


  #184718 17-Dec-2008 09:53
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Have you double checked your firewall/port forwarding settings? UDP Port 5060 for SIP and possibly UDP ports 10000 - 11000 for RDP.

tknz
171 posts

Master Geek


  #187093 2-Jan-2009 22:48
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Remove the first inbound route rule - ignore the firewall ports the other user specified below - your using an outbound sip register which works through NAT - but verify that you don't have "Trunking" turned on in your 2talk options - login via SSH to the server -

type

sudo asterisk -r

this should get you to the cli -

try and make the call and anything asterisk recieves will come up on the GUI -

if it doesn't show up type - at the cli
 sip show registry

make sure it is registered -

Then verify inbound trunk

sip show peers


 
 
 
 


tknz
171 posts

Master Geek


  #187107 3-Jan-2009 01:15
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I think I have figured it out - I was having the same issue this evening - I now have it working please see below -
There are two problems.

1. Trunk mode is Insecure "no" this needs to be insecure "very" this will solve the first problem which is below the X is just disguising my number -

[Jan  3 01:07:30] WARNING[4483]:
chan_sip.c:8507 check_auth: username mismatch, have , digest has <04XXXXXXX>
[Jan  3 01:07:30] NOTICE[4483]: chan_sip.c:13971 handle_request_invite: Failed to authenticate user "Tui Kapo" ;tag=as3e2325ff
david*CLI>

2. Problem -

[Jan  3 01:05:43] NOTICE[11770]: pbx.c:1892 pbx_extension_helper: No such label 'directory/5001' in extension 's' in context 'DID_trunk_1'

So it is trying extension S - which is a catch all phrase - so to resolve this you put the following in extensions.conf
 exten=s,1,Goto(default,6000,1)

6000 was my test extension number -

I believe that in AsteriskNOW that would be to go to Inbound Calling Rules - under pattern matching stick 's' without the quotes - and then the extension you want to go to.

Hopefully these resolve y our issues -

BTW - Asterisk 1.4 with Asterisk GUI 2.0 is great - if your Linux knowledge is ok then you should consider the upgrade project. It is incredibly stable on Debian, with Asterisk 1.4 Stable and Asterisk Gui 2.0

Best of Luck

Tui

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