Unitymedia Voip + Asterisk

mamep

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Hi there,

First of all sorry for my English but my German is not sufficient enough.

Let's go for the plan now..

I have a unitymedia cable connection with static i[ and 3 phone numbers.

As a read through the forum i can see that unitymedia voip numbers can be registered as trunks in asterisk.

Therefore..
I have a rpi 2 model b which should run asterisk and i and use one of the three numbers as trunk for my calls when i'm away..

First question is how to recover voip password from config file?

And secondly is it possible to register one of the numbers with asterisk?

Thank you in advance..
 
The login data with passwords are written down in the welcome letter, you got from unitymedia!

And yes, once you got the password you can register one or all of the three accounts with an asterisk :)
 
The login data with passwords are written down in the welcome letter, you got from unitymedia!

And yes, once you got the password you can register one or all of the three accounts with an asterisk :)

Snuff i can see only telephone numbers in the welcome letter and static ip information but no login info!
 
Then you can add ip-phones ("IP-Telefon") in the fritzbox to register your asterisk or your mobile
 
Snuff is right i've managed to find registration info among a mess of documents..
it is in nXXXXXXXXX_1 , nXXXXXXXXX_2 AND nXXXXXXXXX_3 where is XX refers to your customer number and a common password for all of them..

let me give a shot..

Is there any limitations for the connection to unitymedia servers? Can i use droplets from digitalocean or linode?
 
You can only connect them from your unitymedia cable connection :(
 
So let the magic start with rpi 2 model b :D
 
Here is my sip.conf for Asterisk (also running on a new rpi 2 model b)

i use port 5070 instead of 5060, because the fritzbox has its own sip server on port 5060

[general]
pendantic=no
bindport = 5070
context=incoming
qualify=no
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear
srvlookup=yes
language=de
session-timers=refuse
allowsubscribe=yes
notifyringing=yes
notifyhold=yes
limitonpeers=yes
externrefresh = 10
externhost = MY.DYNAMIC.HOSTNAME
externip = MY.DYNAMIC.HOSTNAME
localnet=192.168.111.0/255.255.255.0
vmexten=VoiceMail
nat = yes

register => h279146802_1:p[email protected]/496087736xxxx

[496087736xxxx]
type=friend
insecure=invite
username=h279146802_1
fromuser=h279146802_1
fromdomain=telefon.unitymedia.de
secret=PASSWORD
host=ssl62.telefon.unitymedia.de
qualify=yes
context=incoming
callbackextension=h279146802_1
nat=yes
allow=all
 
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snuff RasPBX or other release?

Trying to run it with usb from Mikrotik RB951G-2HnD
 
Default Raspian with Asterisk 1.8, but i switch to Asterisk 13 this week
 
Here is the extensions.conf

[incoming]
exten => 496087736XXXX,1,Set(CALLTIME=${STRFTIME(${EPOCH},CET,%d.%m.%y %H:%M)})
exten => 496087736XXXX,n,Set(GLOBAL(ORIGIN)=49${CALLERID(num):1})
exten => 496087736XXXX,n,Macro(revdblookup,${CALLERID(num)},736XXXX,incoming,${CDR(uniqueid)})
exten => 496087736XXXX,n,Set(CALLERID(name)=${RLNAME})
exten => 496087736XXXX,n,GotoIf($["${WHITELIST}" = "1"]?tagschaltung-736XXXX,s,1)
exten => 496087736XXXX,n,GotoIf($["${BLACKLIST}" = "1"]?spam,s,1)
exten => 496087736XXXX,n,Goto(tagschaltung-736XXXX,s,1)

[tagschaltung-736XXXX]
exten => s,1,Noop(Tagschaltung 736XXXX)
exten => s,n,SipAddHeader(P-Preferred-Identity: <sip:${ORIGIN}@ssl62.telefon.unitymedia.de>)
exten => s,n,Dial(SIP/21&SIP/22,30,trg)
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-BUSY,1,Goto(ab-736XXXX,s,1)
exten => s-NOANSWER,1,Goto(ab-736XXXX,s,1)
exten => s-CHANUNAVAIL,1,Goto(ab-736XXXX,s,1)
exten => s-CONGESTION,1,Goto(ab-736XXXX,s,1)

[outgoing]
exten => _0X.,1,Set(CALLTIME=${STRFTIME(${EPOCH},GMT-2,%d.%m.%y %H:%M)})
exten => _0X.,n,Dial(SIP/${EXTEN}@496087736XXXX,60,trg)
exten => _0X.,n,Hangup
 
snuff : I see too many reconnecations with the server..
Do you have the same problem?

2015-04-06 14:08:26 UTC] NOTICE[2734] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #125)
 
No, i have no problems. Is the account also registered in the fritzbox?
 
No it is disabled from fritz box


[2015-04-06 15:03:27 UTC] NOTICE[2734] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #290)
[2015-04-06 15:03:30 UTC] NOTICE[2734] chan_sip.c: Peer '49xxxxxxxxxx' is now Reachable. (1025ms / 2000ms)
[2015-04-06 15:03:47 UTC] NOTICE[2734] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #291)
[2015-04-06 15:04:34 UTC] NOTICE[2734] chan_sip.c: Peer '49xxxxxxxxxx' is now UNREACHABLE! Last qualify: 1025
[2015-04-06 15:05:23 UTC] NOTICE[2734] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #2)
[2015-04-06 15:05:37 UTC] WARNING[2734] chan_sip.c: Probably a DNS error for registration to [email protected], trying REGISTER again (after 20 seconds)
 
it seems that was two instances of asterisk running..
And reconnects stopped
 
To register a VoIP-Device other than the UM-cablebox is difficult and not supported via UM.
Nobody knows how long your configuration works, everytime its possible that unitymedia changes something.
The easiest way is really is to crete for every number one LAN-Phone at the Fritzbox and to connect asterisk to this fritzbox.

I use the same configuration without any problems.
 
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