hkvalet
Specialist
join:2004-10-18
Las Vegas, NV
| reply to im_chandave
Re: [BroadVoice] Calls to Hong Kong are down
said by im_chandave  :I received mine within an hour. The "From:" line was: From: "cs2b@hkbn.com.hk" <cs2b@hkbn.com.hk> You might want to contact them using the above email address. See ya... d.c. There is a list of toll-free number you can call for 2b customer support.
»www.2b.com.hk/eng/faq06.htm
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im_chandave
join:2005-07-28
Cleveland, OH | reply to VOIPnYC
I received mine within an hour. The "From:" line was:
From: "cs2b@hkbn.com.hk" <cs2b@hkbn.com.hk>
You might want to contact them using the above email address.
See ya...
d.c. |
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VOIPnYC
join:2005-06-26
New York, NY | reply to im_chandave
I signup at www.2b.com.hk about a weeks ago, but still have not receive the password yet? anyone know how long it takes to receive the password? thanks |
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tillama
Premium
join:2005-10-22
China | reply to kitkitnet
KitKitNet, good to know that you have some success on the incoming call problem for 2B.
I am interested to know how did you set SPA-3000? is it possible to post the setting (in HTML dump) for everyone's reference?? |
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kitkitnet
join:2001-01-25
Oxford
| reply to im_chandave
nc22,
I am aware of that, if you read the entire thread, you'll find I am using Sipura SPA-3000 (I don't use *) and posted the Sipura's Syslog to Dave, I never had a successful incoming before the 2B user (which he's using 2B softphone), I was really amazed by the fact that it works when a fellow 2B user called me.
I'll try to ask him to call again for a log when he is available. I'll suggest anyone with a Sipura try to find a fellow 2B user (with TA or 2B's softphone) to test and post the log to help the community.
Cheers |
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nc22
join:2004-04-16
Everett, WA | reply to kitkitnet
kitkitnet, if you are not using sipura/pap2 but using *box, dave said it works for incoming calls, it works for sipura to incoming call rings and caller ID, but the call drop off in 2 secs after you pick up the phone. |
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kitkitnet
join:2001-01-25
Oxford
| reply to im_chandave
I am kind of busy this weekend, haven't tested dave's setting. But something strange just happened! I received an incoming calls from a fellow 2B user today, it works!
Too bad I am not running syslog at that time, I have no idea why it worked this time! The caller ID display the phone number with Caller Name "HKBN HKBN".
I'll keep testing dave's setting later this week, hopefully Sipura incoming call issue can be resolved soon. |
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im_chandave
join:2005-07-28
Cleveland, OH
| reply to cplus
said by cplus :...now, i get line 1 Registration State: Can't connect to login server oh, i have sv 3.1.3. You most likely need to "reset" your 2b account via »pa.2b.com.hk and power-cycle your PAP2. You should get re-REGISTERed again.
See ya...
d.c |
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cplus
join:2005-09-30
Fullerton, CA
| reply to im_chandave
thanks. I've updated dial plan and added STUN server. I noticed that my stanaphone on line 2 does not like some of the STAN servers as it causes it to go offline.
I rechecked the settings and was able to get a dial tone - but by the time i found a HK test number, the line went dead.
now, i get line 1 Registration State: Can't connect to login server
oh, i have sv 3.1.3. |
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pktill
@com.hk | reply to gkong
I tried to setup HKBN with my SPA-3000.
I have also the same problem that inbound call drops immediately right after picking up the phone.
I hope somebody can resolve this problem soon... |
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im_chandave
join:2005-07-28
Cleveland, OH
| reply to pktill
said by pktill :Hi chandave, I tried to use HKBN's 2B service with my SPA-3000. Outbound calls works fine.... The phone rings when there is inbound calls, but it disconnect immediately after pick up. I followed all setting you have suggested in » www.voip-info.org/wiki/view/Sipu···+HKBN+2b. Can I send you the log and helps to check what is happening? THanks a lot Till If you send me your log and your configuration (goto http://IP_of_your_SPA3k/admin/advanced and save the HTML page), I'll try and help you diagnose the problem. No promises.
But, before you can do anything, you need to register with broadbandreports.com.
See ya...
d.c. |
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pktill
@com.hk
| reply to im_chandave
Hi chandave,
I tried to use HKBN's 2B service with my SPA-3000.
Outbound calls works fine....
The phone rings when there is inbound calls, but it disconnect immediately after pick up.
I followed all setting you have suggested in »www.voip-info.org/wiki/view/Sipu···+HKBN+2b.
Can I send you the log and helps to check what is happening?
THanks a lot
Till |
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im_chandave
join:2005-07-28
Cleveland, OH
| reply to cplus
I've updated my HKBN SPA config on voip-info.org to reflect support for a HK dialplan:
»www.voip-info.org/wiki/view/Sipu···+HKBN+2b
You can choose any STUN server from the list provided in »www.voip-info.org/wiki/view/STUN.
As for "Registration State: Offline", you might need to reset your "2b" account status. Again, check the "Sipura settings HKBN 2b" page on voip-info.org.
See ya...
d.c. |
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cplus
join:2005-09-30
Fullerton, CA
| reply to im_chandave
HI. just found 2b VOIP and need some help configuring PAP2. I found the info at voip-info.org and changed all the required settings. I cannot get a dial tone.
what is the correct DIAL PLAN? I have ([2-9]xxxxxx|011xx.|1[2-9]xx[2-9]xxxxxx) which i figure is wrong since i was given a hong kong number.
which usable STUN server do you recommend to use?
When I check my PAP2 info page, Registration State: Offline
Any ideas? |
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im_chandave
join:2005-07-28
Cleveland, OH
| reply to kitkitnet
said by kitkitnet :Dave, I have sent you the log via IM. And I tried playing around codec before, but it also failed. I'll try the NAT Support Parameters in this weekend. And about log, I am not an expert in SIP/RTP, but I found 2B's incoming session can't init a RTP Rx Session. I have no idea if it is related. : : You can find a normal incoming session in the last part of log, via voipfone.co.uk, bascially, I think they are pretty the same except the RTP session. I don't know if it is the reason why I can't receive any calls? BTW, I am using Draytek Vigor V2600We Router. The logs was just what I needed. The problem is that you are behind a NAT and your PAP2 is telling HKBN's SIP Proxy to send the RTP stream to our Private network number (192.168.1.240). HKBN is trying to initiate the RTP stream to your private network number and fails. It then tears down the call.
I wrote up this issue in my description of HKBN's service at »www.voip-info.org/wiki/view/Hong···roadband :
quote:
When early media (SIP response 183) occurs as a result of an incoming INVITE from HKBN, the Packet Voice Gateway will terminate the call if the source UDP port for the RTP connection differs from the source port specified in SDP of the SIP 200 response from your SIP UA. Therefore, if your router/firewall has a tendency to re-NAT or re-PNAT the source address and/or port, you must have STUN or UPNP support on your SIP UA in order to make sure you put in the right IP address and/or port in your SDP.
In your case, the problem is not the port (although, I'm not sure about whether your router supports source port preservation through the NAT), but the address you are using in your SDP. It should be your public IP address instead of the private address of 192.168.1.240.
Try turning on NAT support via the following:
•SIP > NAT Support Parameters > Send Resp To Src Port = "yes"
•SIP > NAT Support Parameters > STUN Enable = "yes"
•SIP > NAT Support Parameters > STUN Test Enable = "yes"
•SIP > NAT Support Parameters > STUN Server = stun.xten.com
•Line 1 > NAT Settings > NAT Mapping Enable = "yes"
•Line 1 > SIP Settings > Restrict Source IP = "yes"
Reboot the PAP2 after changing these settings (I've seen my SPA-3k not act on these changes until I power-cycled the unit).
Also, if you do successfully establish a connection with G729a codec, your DTMF tones will not work if you still have InBand as the TxMethod.
The reason why your incoming calls from voipfone.co.uk work is because they are using Asterisk. Most likely, they have "nat=yes" in their sip.conf configuraton for your account. This tells Asterisk to ignore the IP addr and port you specify in your SDP and just send the RTP stream to the source of the RTP stream you are currently using to connect to them.
See ya...
d.c. |
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kitkitnet
join:2001-01-25
Oxford
| reply to im_chandave
Dave, I have sent you the log via IM. And I tried playing around codec before, but it also failed. I'll try the NAT Support Parameters in this weekend.
And about log, I am not an expert in SIP/RTP, but I found 2B's incoming session can't init a RTP Rx Session. I have no idea if it is related.
Normal:
[0:0]RTP Tx Up (pt=8->d4bba2b2:15978)
[0:0]RTCP Tx Up
[0:0]RTP Rx 1st PKT @16468(2)
[0:0]DEC INIT 8
[0:5060]<<212.187.162.179:5065
[0:5060]<<212.187.162.179:5065
2B:
[0:0]RTP Tx Up (pt=18->3d5c7706:58066)
[0:0]RTCP Tx Up
[0:5060]<<203.80.89.135:5060
[0:5060]<<203.80.89.135:5060
You can find a normal incoming session in the last part of log, via voipfone.co.uk, bascially, I think they are pretty the same except the RTP session. I don't know if it is the reason why I can't receive any calls?
BTW, I am using Draytek Vigor V2600We Router.
Thanks |
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im_chandave
join:2005-07-28
Cleveland, OH
| reply to kitkitnet
Hi kitkitnet,
Since you have the SIP converstation, would you IM it to me? Also, try forcing your PAP2 to only use G711a and turn off all the other G-series codecs.
About the only other thing I can think of is that you ensure you have "SIP > NAT Support Parameters > Send Resp To Src Port" set to "yes".
Are you behind a D-Link DI-series router?
See ya...
d.c. |
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kitkitnet
join:2001-01-25
Oxford
| reply to im_chandave
I just want to add that I also can't receive any call from 2b (outgoing is ok). I've made sure the DMZ is pointed to SPA-3000, STUN on/off, NAT Mapping on/off, and many more combinations with service/codec/RTP but incoming call still failed.
I tried to read SPA-3000's debug log and I found nothing wrong with it, The call just drop (with normal BYE) right after the phone is picked up. As first I was thinking if problem with Codec, but seem it is not the case.
I tried to use a computer with X-Lite, income call is working fine. Is it a problem with Sipura, I already tried both 2.0.x and 3.0 firmware? And I have basically tested for many days but not a single time I can get an incoming call, the call drop right after I pickup the phone. Really really weird.
Sure, other VoIP work fine in my SPA-3000. |
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nc22
join:2004-04-16
Everett, WA
| reply to im_chandave
Thanks ChanDave, I turned everything on without a stun server, and working for calling somebody, however, I could not receive any calls and I could not add a stun server (could not registered once stun server was added). You have the same problem too? |
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im_chandave
join:2005-07-28
Cleveland, OH | reply to gkong
If you remember, I could successfully call you. I will have to play around with my PAP2 when I get it.
See ya...
d.c. |
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