Connecting Asterisk to Digium VOip Gateway


#1

Hello Experts,

Thanks in Advance.

I have installed Asterisk certified/13.21-cert2. we have Digium Voip gateway, E1 Link terminated on
DIGIUM
both Digium and asterisk able to talk each other.

DIGIUM IP address:-172.16.10.5
Asterisk:-172.16.14.119

I am new in asterisk i did some things internal call are working properly but when i want to make call outside its not going through.

please find below is SIp.Conf:-

[general]
allowguest=no
udpbindaddr=172.16.14.119:5060
tcpenable=no

[friends_internal](!)
type=peer
host=dynamic
context=from-internal
disallow=all
allow=ulaw

[104](friends_internal)
secret=104 

[105](friends_internal)
secret=105  

[106](friends_internal)
secret=106

[107](friends_internal)
secret=107


[voicehos]
host=172.16.10.5
type=peer

Ext.conf

[from-internal]
exten => 104,1,Dial(SIP/104,20)
exten => 105,1,Dail(SIP/105,20)
exten => 106,1,Dial(SIP/106,20)
exten => 107,1,Dial(SIP/107,20) 

exten => _0XXXXXXXX,1,Dial(SIP/${EXTEN:0}@172.16.10.5,,r)
exten => _5XXXXXXXX,1,Dial(SIP/${EXTEN:0}@172.16.10.5,,r)
exten => _6XXXXXXXX,1,Dial(SIP/${EXTEN:0}@172.16.10.5,,r)

kindly let me know where am wrong.


#2

You are wrong in not supplying a log of the failed call.

Also, you haven’t marked the configuration file extract as unformatted text.

Subject to the last point, it looks like extension friends_internal is open (no password) and the other extensions have everything except the password defaulted.

Extension 105 has an invalid application name.

There are missing $ for two of your outgoing extension ranges.


#3

Hello David,

Sorry to give half and some wrong information.

i have attached text file containing sip.conf, ext.conf, call Logs and Sip show peers details.

please I hope have provided enough details and a right way about the issue.
sorry in advance if i could not provide what you asked for.
call logs.txt (1.1 KB)
ext.txt (330 Bytes)
sip show peers.txt (1014 Bytes)
sip.txt (374 Bytes)

And, can you please explain a little bit as said above about invalid and without password.
i would be really grateful to you if you could help me to understand the ext.conf file as i have a lil knowledge about it.


#4
 == Using SIP RTP CoS mark 5
       > 0x7f7a0400ae90 -- Strict RTP learning after remote address set to: 172.16.14.99:8000
[Nov  7 01:50:55] NOTICE[1529][C-00000000]: chan_sip.c:26459 handle_request_invite: Call from '105' (172.16.14.99:36778) to extension '0501481420' rejected because extension not found in context 'from-internal'.
[Nov  7 01:51:28] NOTICE[1529]: chan_sip.c:28463 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 104
    -- Registered SIP '104' at 172.16.14.119:63117
       > Saved useragent "Z 3.3.25608 r25552" for peer 104
[Nov  7 01:51:40] NOTICE[1529]: chan_sip.c:28463 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 104
  == Using SIP RTP CoS mark 5
       > 0x7f7a0400ee70 -- Strict RTP learning after remote address set to: 172.16.14.119:8000
[Nov  7 01:52:00] NOTICE[1529][C-00000001]: chan_sip.c:26459 handle_request_invite: Call from '104' (172.16.14.119:63117) to extension '0501481420' rejected because extension not found in context 'from-internal'.
[from-internal]
exten => 104,1,Dial(SIP/104,20)
exten => 105,1,Dail(SIP/105,20)
exten => 106,1,Dial(SIP/106,20)
exten => 107,1,Dial(SIP/107,20)

exten => _0XXXXXXXX,1,Dial(SIP/${EXTEN:0}@172.16.10.5,,r)
exten => _5XXXXXXXX,1,Dial(SIP/${EXTEN:0}@172.16.10.5,,r)
exten => _6XXXXXXXX,1,Dial(SIP/${EXTEN:0}@172.16.10.5,,r)

The dialplan expects 9 digits. You have provided 10.


#5

Thanks for reply,

after giving 10 digits,

I got this:-
== Using SIP RTP CoS mark 5
> 0x7fe84400a400 – Strict RTP learning after remote address set to: 172.16.14.119:8000
– Executing [0501481420@from-internal:1] Dial(“SIP/104-00000000”, “SIP/0501481420@172.16.10.5,r”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/0501481420@172.16.10.5
[Nov 7 02:55:05] WARNING[4881][C-00000000]: chan_sip.c:24056 handle_response_invite: Received response: “Forbidden” from ‘sip:104@172.16.14.119;tag=as4095c2c9’
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/104-00000000’ status is ‘CHANUNAVAIL’


#6

There is no section in sip.conf with the name 172.16.10.5, so you are making a call with default options and no authentication data. Even the sip.conf section that could be used to access that address has no authentication data.

If you think that the gateway should not be asking for authentication data, you need to consult its vendor. This forum does not provide support for Digium/Sangoma commercial products.

You don’t want that device to operate without authentication!!

[general]
allowguest=no
udpbindaddr=172.16.14.119:5060
tcpenable=no


[friends_internal](!)
type=peer
host=dynamic
context=from-internal
disallow=all
allow=ulaw

[104](friends_internal)
secret=104

[105](friends_internal)
secret=105

[106](friends_internal)
secret=106

[107](friends_internal)
secret=107


[voicehos]
host=172.16.10.5
type=peer

#7

I won’t pretend to provide any support via this community forum, because we have official avenues for Support, but I’ll note that the gateways operate in one of three modes for endpoints:

  1. No authentication
  2. Send outbound authentication
  3. Expect inbound authentication

And that’s configured when setting up the endpoint from within the UI.