I've just corrected some configurations above on /etc/asterisk/extensions.conf about dialout rules that were put on wrong section.
@pilovis I'm trying to setup small sip server on OpenWrt (linksys EA4500) with asterisk package installed. Need help on how to configure voip server setting for bundled with fiber optic Broad band service
What do you mean for voip server bundled with fiber optic Broad band service?
Are you talking about the Voip service offered by your Internet provider or what?
Let me know so I can answer your question.
Thanks.
should this read 0123456789:username:5060/0123456789? In my case, username is an email address. 'user' is the telephone number.
another stumbling point: you always write "voip.provider.address.com", which does not fit to the SIP data from my provider, i.e. SIP-proxy=Registrar=Realm=tel.t-online.de and STUN server=stun.t-online.de. The stun-server must be given in the settings, otherwise there will be no connection. Where would I put it?
Side note: this is by far the best instruction I have seen after a full day search. Just down to the point. Thank you for that.
Cheers,
Oscar
I can see from your provider voip instruction:
Allgemeine Einstellungen:
| SIP-ID/Benutzer: | Ihre Telefonnummer |
|---|---|
| Bildschirmname (falls vorhanden): | Ihre Telefonnummer |
| Authentifizierungsname/Benutzername: | Ihre E-Mail-Adresse, z. B. ihr-name@t-online.de |
| Passwort: | Ihr Passwort |
| SIP-Proxy: | tel.t-online.de |
| Registrar: | tel.t-online.de |
| Realm: | tel.t-online.de |
| STUN-Server: | stun.t-online.de |
| Outbound-Proxy: | leer lassen oder ebenfalls tel.t-online.de |
Assume your telephone number is +49 30 8151234, your username is your telephone number without ld prefix: 0308151234
[general]
register => username:password:username@tel.t-online.de@tel.t-online.de/username
Known issues
Telekom doesn’t support session timers yet. Without “session-timers=refuse” in sip.conf calls getting dropped every 30 minutes. If this Message occurs append “~480” to the register-line at sip.conf:
register => username:password:username@tel.t-online.de@tel.t-online.de/username~480
/etc/asterisk/sip.conf :
[general]
register => username:password:username@tel.t-online.de@tel.t-online.de/username~480
;; Deutsche Telekom AG
[t-online.de]
defaultuser=username@tel.t-online.de
fromuser=username
secret=password
context=incoming
extension=username
type=peer
host=tel.t-online.de
fromdomain=tel.t-online.de
realm=tel.t-online.de
nat=no
directmedia=no
insecure=port,invite
canreinvite=yes
dtmfmode=inband
qualify=yes
session-timers=refuse ; Important!
allow=!all,alaw,g722
/etc/asterisk/extensions.conf :
[incoming]
exten => username,1,Verbose(Incoming call via DTAG)
same => n,Dial(SIP/john)
same => n,Hangup
[trunk-dtag]
exten => _X.,1,Verbose(Outgoing call via DTAG)
same => n,Set(CALLERID(all)=username)
same => n,Dial(SIP/t-online.de/${EXTEN},180,rg)
same => n,Hangup
Configure STUN
edit /etc/asterisk/rtp.conf
add at the end of the file this line:
stunaddr=stun.t-online.de
Hello Pilovis,
thanks a million! I will look into this over the weekend. The sip.conf file for the my SIP provider looks almost completely different to the generalised one above.
Just to reassure: the line register would read as:
register => 0308151234:my-password:0308151234@tel.t-online.de@tel.t-online.de/0308151234~480
AND
defaultuser=0308151234@tel.t-online.de
not my-email-name@t-online.de (which is "the user" for the Telekom).
Looking forward to get this working...
Cheers
Oscar
1 Like
By checking everything again, I have now a doubt: the two "@" in the register string do not seem correct to me, I would try instead:
register => 0308151234:*my-password*@tel.t-online.de/0308151234~480
hello pivolis is it possible to install asterisk 16 or 13 on nor flash 64 mb .. ??
Yes, it is
@pilovis I have a dongle configured I can make and receive calls from it, but I don't hear the audio (voice), but when I click the dial-pad on the SIP client the clicked number is hearable in SIP and calling PSTN (mobile) device.
I have a Huawei E156G modem with fw 11.609.10.00.00.B418.
[Feb 18 16:43:07] WARNING[17748][C-00000004]: channel.c:1079 channel_indicate: [Dongle/dongle0-0100000001] Don't know how to indicate condition 32
This is a chan dongle issue, you should try to update the firmware on your dongle, try E160Update_11.609.10.02.432.B418.exe
On this software, the modem was not working in call manager on Windows and it won't hear anyrthing.
I tried firmware E156G Update_11.608.08.81.00.B409.exe call now works and also enabled voice feature by Huawei Modem Unlocker v5.8.1 by bojs tool. Also the port of audio was incorrectly set.
asterisk-chan-dongle project was abandoned 6 years ago, it was originally developped for asterisk 1.6 and ported to 11, actually it is quite buggy and does not work well with all huawei dongles.
another thing you can try, it is to connect the dongle to an external powered USB HUB, it could be that your router cannot give all the power required by the dongle (500 mA)
Obviously I always used a general "voip.provider.address.com" in this how-to, just to let people understand that they have to change this with their voip provider settings
the chan_dongle development was abandoned a long ago, this asterisk channel has never worked well, it is quite buggy.
Inviato dal mio Galaxy
-------- Messaggio originale --------
Yes, I know that chan_dongle is buggy 
But I installed a lot of firmware for this Polish Play E156G branded modem. This is firmware related issue for this brand or modem. I saw recommendations to use the Update_11.608.08.81.00.B409.exe for Play E156G and everything works well on Windows and Asterisk with OpenWRT.