Belgacom BBox

I am trying to configure a Voip enabled BBOX router. It has the following SIP configuration page:


SIP Setting


Configure the following SIP-related parameters. And press Save button.

SIP Parameters
SIP Stack Mode NGN Stack IMS Stack
SIP UserAgent
UserAgent Domain:
UserAgent Port:
Proxy Setting
Proxy Domain:
Proxy Port:
Registrar Setting
Registrar Domain:
Registrar Port:
Outbound Proxy
Outband Proxy Domain:
Outband Proxy Port:
Re-Registration Time Interval
Support PRACK (RFC3262) Enable
Support SIP Session Timer (RFC 402 Enable
'Session Expires' timer:
'Minimum Session Expires' timer:

It also has a "Phone Number Setting" Page which is a little more straight forward:
Phone Number
Display Name
Realm
Username
Password

I would like to ask for some help with these settings, because registration fails. I tried both pbxes.org and 188.40.65.148 for User Agent, Proxy and Registrar settings with port 5060 everywhere.
On the status page I see this :
SIP URL ---------------------------------Registration
sip:luxapo-111@188.40.65.148 ---Fail

This post has been edited 1 time(s), it was last edited by lux on 08.10.2010 at 16:09.

RE: Belgacom BBox

If you have already tried successfully to setup other ITSPs on the specific IAD, but cannot setup a PBXes extension, chances are it does not support the hyphen (-) in the user part of the SIP URI.

Try to figure out who the manufacturer of the Bbox router is, and inquire with them if the hyphen is a valid character. Most ITSPs only use numbers in the user part or the SIP URI, so the IAD vendors elect to forget the requirements of the SIP RFCs.

RE: Belgacom BBox

REGISTRATION PROBLEM SOLVED.

I had to change the setting of VPI/VCI on the ATM Interface of MAC Encapsulated Routing (under ATM PVC) from 0/40 to 1/40.


UNFORTUNATELY,
There is another problem, though: Every time I dial out, I get a busy tone. Even if I just dial another extension. I thought it might be a problem with DTMF, I tried fiddling with the telephone settings: Disabled 'Support Out of Band DTMF', and tried different codecs (G729 and G.723.1) to no avail. I can call from 101 to 111 no probs, but calls from 111 to 101 (or to any number, really) get a busy response. Here is the log:


Oct 28 11:49:54 VERBOSE[106684] logger.c: -- SIP/luxapo-111-01c5 is ringing
Oct 28 11:49:54 VERBOSE[106669] logger.c: -- Local/111@from-internal-cont/n-c988,1 is ringing
Oct 28 11:50:20 VERBOSE[62714] chan_sip.c: SIP response 200 to standard invite
Oct 28 11:50:20 VERBOSE[62714] logger.c: Found RTP audio format 8
Oct 28 11:50:20 VERBOSE[62714] logger.c: Peer audio RTP is at port 87.64.23.165:5002
Oct 28 11:50:20 VERBOSE[62714] logger.c: Peer video RTP is at port 87.64.23.165:65535
Oct 28 11:50:20 VERBOSE[62714] logger.c: Found description format PCMA
Oct 28 11:50:20 VERBOSE[62714] logger.c: Capabilities: us - 0x18061e (gsm|ulaw|alaw|g726|speex|ilbc|h263|h263p), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Oct 28 11:50:20 VERBOSE[62714] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Oct 28 11:50:20 VERBOSE[62714] chan_sip.c: Oooh, we need to change our formats since our peer supports only 0x8 (alaw) and not 0x4 (ulaw)
Oct 28 11:50:20 VERBOSE[106684] logger.c: -- SIP/luxapo-111-01c5 answered Local/111@from-internal-cont/n-c988,2
Oct 28 11:50:20 VERBOSE[106669] logger.c: -- Local/111@from-internal-cont/n-c988,1 stopped sounds
Oct 28 11:50:20 VERBOSE[106669] logger.c: -- Local/111@from-internal-cont/n-c988,1 answered SIP/luxapo-101-b5b9
Oct 28 11:50:20 VERBOSE[106669] logger.c: We're at 88.198.69.250 port 39786
Oct 28 11:50:20 VERBOSE[106669] logger.c: Video is at 88.198.69.250 port 41336
Oct 28 11:50:20 VERBOSE[106669] logger.c: Adding codec 0x200 (speex) to SDP
Oct 28 11:50:20 VERBOSE[106669] logger.c: Adding codec 0x2 (gsm) to SDP
Oct 28 11:50:20 VERBOSE[106669] logger.c: Adding codec 0x100000 (h263p) to SDP
Oct 28 11:50:20 VERBOSE[106669] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Oct 28 11:50:22 VERBOSE[106684] chan_sip.c: Hangup call SIP/luxapo-111-01c5, SIP callid 3e136efd2d1bc88133e8bfae25572f89@88.198.69.250
Oct 28 11:50:22 VERBOSE[106669] chan_sip.c: Hangup call SIP/luxapo-101-b5b9, SIP callid 730981728392@172.24.80.168
Oct 28 11:50:26 VERBOSE[62714] logger.c: -- Registered SIP 'luxapo-101' expires 3600




Please advise.

This post has been edited 2 time(s), it was last edited by lux on 28.10.2010 at 11:59.

RE: Belgacom BBox

is that the log showing the busy?

RE: Belgacom BBox

I thought it was, but now I tried to reproduce it and I can't: Nothing gets logged. I turned SIP trace on, still nothing -- obviously the above log is not for the dial attempt. On the status page (on the flash GUI) the call attempt does NOT get displayed either. Looks like the busy tone I get immediately after dialing is generated on the Bbox itself. Any suggestions?

RE: Belgacom BBox

Based on your latest comments, I would suggest that you connect the Bbox to a hub, not a switch, along with a PC running a packet tracing utility. Then you will be able to see all the packets originating from the Bbox to the rest of the network.

If no SIP packets originate from the Bbox, then you might need to reset it, and reconfigure its' settings from scratch. If that does not help either, then I suggest you ask for a replacement unit.

RE: Belgacom BBox

Diafora,

I do not think you read my posts very carefully.

My bbox connects, it rings on incoming calls and I can take these calls. I can tell you that there are a lot of SIP packets involved in this process without ethereal. My problem is that DTMF signals don't seem to go through for some reason. I don't suppose I will find a solution for this but I made a point of reporting the problem on this forum so that YOUR other users in the future might find it helpful.

The BBOX is otherwise a provider branded user premise adsl equipment and as such it is not intended for third party VoIP provider services. Reseting, reconfiguring or replacing is not really a viable option.

RE: Belgacom BBox

Hey luxapo,

I believe that I read all your comments properly:

Zitat:

I am trying to configure a Voip enabled BBOX router. It has the following SIP configuration page: ... I would like to ask for some help with these settings, because registration fails. I tried both pbxes.org and 188.40.65.148 for User Agent, Proxy and Registrar settings with port 5060 everywhere. On the status page I see this : SIP URL ---------------------------------Registration sip:luxapo-111@188.40.65.148 ---Fail

You then resolved the issue, by fiddling with the VPI / VCI settings, and now are facing DTMF transmission issues. To which I replied that if you can get a packet trace off what the BBox is transmitting, that would point to whether the BBox creates the OOB DTMF signaling. It would also help us identify which OOB method it employs for DTMF transmission.

Zitat:

On the status page (on the flash GUI) the call attempt does NOT get displayed either. Looks like the busy tone I get immediately after dialing is generated on the Bbox itself. Any suggestions?

Since you mentioned the PBXes based SIP trace was not working for you, I suggested you setup an on-premise packet trace, based on your comment which insinuated no packets originated from your BBox towards PBXes at all. But since the ADSL modem is built-in to the BBox, if the PBXes SIP trace utility is not working for you, there is no other way to troubleshoot your DTMF issue.

Zitat:

I would like to ask for some help with these settings, because registration fails.

If your comment was purely informative for other PBXes users as you claim, you should had stated your intentions in your first post of this thread. Instead, you were asking for help to get this working. Hardly an informational only post.