VoipBuster Hang Up problem when using hardware

I have signed up to VoipBuster.com and have made calls using the software client sucessfully.

However I really want to use a proper phone connected to my Draytek 2600VG router.

I have configured the router as follows:

SIP Port: 5060
Registrar: sip.voipbuster.com

When I dial a number the other phone rings but as soon as it is picked up the connection ends!

Can anyone suggest how I can get this working?

Hi p*j*b,

Welcome to the forum. Honestly, I don't have a solution for you, let alone with the Draytek 2600VG router. However, this symptom sounded like the device isn't setup and/or registered properly. Can you please provide us with your VoIPBuster SIP setup as well as how you connect this Draytek 2600VG to the Internet feed?

The Draytek Vigor 2600VG is 4 port and wireless router with a built in ADSL modem and two FXS ports. (http://www.draytek.com/product/voip/vigor2600vg/vigor2600vg_html_spec.php)

So I have set the SIP Port to 5060 and the Registrar to sip.voipbuster.com.

I then plugged a phone into FXS port 1 and dialled my uk home number (including international dialling code), My home phone rings but as soon as I pick it up the call terminates.

According to the router I am registered:

> voip sip reg
% voip sip registrar <server>
% Now: sip.voipbuster.com

% Status of registration:
endpt1 = Yes
endpt2 = No

This is the output from the routers voip debugging:

<--Receive Message <18:18:55>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP myip:5060;branch=z9hG4bK-JEG-12119;received=myip; rport=5060
From: user
<sip:user@sip.voipbuster.com:5060>;tag=PfW-29421
To: <sip:mynumber@sip.voipbuster.com>;tag=as29d77d83
Call-ID: ocN-22318@myip
CSeq: 2 INVITE
User-Agent: SipProxy
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:mynumber@213.61.187.147>
Content-Type: application/sdp
Content-Length: 419

v=0
o=root 29062 29062 IN IP4 213.61.187.147
s=session
c=IN IP4 213.61.187.147
t=0 0
m=audio 48764 RTP/AVP 4 3 0 8 111 5 10 7 18 110 97
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=silenceSupp:off - - - -

<--Receive Message <18:19:03>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP myip:5060;branch=z9hG4bK-JEG-12119;received=myip; rport=5060
From: username <sip:username@sip.voipbuster.com:5060>;tag=PfW-29421
To: <sip:mynumber@sip.voipbuster.com>;tag=as29d77d83
Call-ID: ocN-22318@myip
CSeq: 2 INVITE
User-Agent: SipProxy
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:mynumber@213.61.187.147>
Content-Length: 0

I dont understand much of this but it would seem that a session has started but then gets terminated by with the 403 Forbidden message.

Any ideas?

Hi p*j*b,

Can you please tell us how you setup your Draytek Vigor 2600VG with the VoIPBuster service? Since Draytek Vigor 2600VG is a complete package, i.e. modem, router + VoIP, I need to know if you have mistakenly enabled the device with any Outbound Proxy and/or STUN Server features.

Also, you may want to check this article on a different forum where another Draytek Vigor 2600VG owner has run into a bit different problem that probably due to SIP reregister Proxy time period.

OK there are two screens that configue voip.
The first is configured as follows:

sip port 5060
registrar sip.voipbuster.com
username ****
password ****
expiry time 1 hour

The other screen has:

Default Codec: G.711MU
Packet size: 20s
DTMF: "Payload 101" SIP Info
Dynamic RTP port start: 5004
Dynamic RTP port end: 5006

There are no settings to configure STUN or an Outbound proxy.

Hi p*j*b,

AFAIK, your setup does not need any Outbound Proxy and STUN Server mainly because the VoIP/SIP setup on your Draytek Vigour 2600VG modem/router is based on the IP address assigned by your ISP (and I assumed it is a real/public IP address).

I re-read all your posts in this thread again and had just spotted the error code of 403 Forbidden from your previous post. According to this post, the VoIPBuster server had refused to fulfill your request. The reason is either your account has no permission to call or the PSTN number you tried to call is restricted by VoIPBuster. You may want to try to call other numbers to see if the VoIP setup on your 2600VG really works.

Please go to read this post. It has some other information that may interest you to know. At any rate, it's not a bad idea to spare some of your time to read the available FAQs as well as other posts and replies. I reckon you will learn a lot from those posts.

hey there...

not sure if this is the problem, but by looking at the specs for the router, I noticed that it doesn't support GSM.

If recall correctly, voipbuster, on their initial connection will say, "zero cent per minute".

That's sent over the GSM codec, but then the call resumes to G711 or whatever your settings might be.

just a thought...


.h

hemantp :

That's sent over the GSM codec, but then the call resumes to G711 or whatever your settings might be.

Hello Hemantp,

Welcome to the forum. This sure is very interesting. I am sure some of us will have some interests in this. It will be nice if you would be able to provide us the link on this subject.

Thanks for the info hemantp,

I guess that would make sense in terms of the line going dead as soon as the phone is picked up.

I have checked and the router only supports the following codecs:

G.711MU (64Kbps)
G.711A (64Kbps)
G.729A/B (8Kbps)
G.723 (6.4kbps)
G.726_32 (32kbps)

Can anyone suggest an alternative to VoipBuster that will e compatible with this router?

hemantp :

hey there... not sure if this is the problem, but by looking at the specs for the router, I noticed that it doesn't support GSM. If recall correctly, voipbuster, on their initial connection will say, "zero cent per minute". That's sent over the GSM codec, but then the call resumes to G711 or whatever your settings might be. just a thought... .h

Yes correct.

When the operator says"zero cents per minute " it's in GSM and then when the phone starts ringing changes to another codec.
rgds.hilario.

I have got this working!

I entered stun.voipbuster.com as the Registrar instead of sip.voipbuster.com and it just works.


I found the solution on the Draytek UK Support Forum, the solution wasn't posted until 25th June, so I missed it when I did my first trawl for answers.

Thanks to everyone that replied.