benjamin
All propositions are of equal value
join:2000-12-20
New York, NY
clubs:
 ·BroadVoice
 ·RCN CABLE
| reply to fortissimo
Re: [BroadVoice] Calls to Hong Kong are down
said by fortissimo  :Does calling to Hong Kong include mobile (cellular) phone numbers in the World or the World Plus plans? I thought they didn't include them, or I am wrong about that? You are correct. Calls to mobile phones are not included to any country. |
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im_chandave
join:2005-07-28
Cleveland, OH
| For those of you that call Hong Kong on a regular basis, why not just subscribe to Hong Kong Broadband's softphone?
It's a standard SIP-based VoIP system. It costs $48 HKD/month ($6.15 USD). You get an Hong Kong DID with unlimited incoming and outgoing minutes. Additionally, since it's a Hong Kong phone number, there isn't a difference in price for landline or mobile (cellular).
If signing up is a problem (the online registration is in Chinese and you need a Hong Kong residential address), then you can get the Hong Kong person you normally call to sign you up.
More info about Hong Kong Broadband's softphone service is available at:
»www.voip-info.org/tiki-index.php···roadband
See ya...
d.c. |
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hkvalet
Specialist
join:2004-10-18
Las Vegas, NV
| said by im_chandave :It's a standard SIP-based VoIP system. It costs $48 HKD/month ($6.15 USD). You get an Hong Kong DID with unlimited incoming and outgoing minutes. Additionally, since it's a Hong Kong phone number, there isn't a difference in price for landline or mobile (cellular). Thanks for your great info. So are you using it right now? I am interested about this service, but I have some questions for you:
1. Can I use my own device? I'm using SPA-2000.
2. How's the sound quality if I use in the U.S.?
3. Their proxy is in Hong Kong, so I think there will be a long delay for all calls if I call from the U.S., right?
Thanks for your time.
--
**Students are the CUSTOMERS of teachers** »RateMyProfessors.com |
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im_chandave
join:2005-07-28
Cleveland, OH
| said by hkvalet :said by im_chandave :It's a standard SIP-based VoIP system. It costs $48 HKD/month ($6.15 USD). You get an Hong Kong DID with unlimited incoming and outgoing minutes. Additionally, since it's a Hong Kong phone number, there isn't a difference in price for landline or mobile (cellular). Thanks for your great info. So are you using it right now? I am interested about this service, but I have some questions for you: 1. Can I use my own device? I'm using SPA-2000. 2. How's the sound quality if I use in the U.S.? 3. Their proxy is in Hong Kong, so I think there will be a long delay for all calls if I call from the U.S., right? Thanks for your time. Yes, I'm using it right now. I have 2 accounts. One for personal use (hosted in HK) and one for testing (hosted in the US). Both are on Asterisk boxes.
•You can use your own device. I've had success with a SPA-3000 from the US to HKBN. Unfortunately, I didn't keep the configuration. But, it should be easy to do. I've written information about how to configure an Asterisk box for HKBN. You should be able to adapt it to the SPA. Just remember to turn on STUN/VIA/RPORT support on the SPA if you are behind a NAT/Firewall.
•Sound quality is fine. They support G.711a so you are using 80kbps (each way) for audio. Also, their Nortel Voice Packet Gateway is hooked into a decent backend infrastructure. They are not running a SIP Express Router, an Asterisk box, and a few PRI cards (like some VoIP companies in the US).
•Average RTT are around 220-250ms for a IP packet. There is some latency but not so much that conversations are undecipherable. I've had no problems with US calling a HK landline. You will notice slightly more latency when calling to a mobile phone (especially when that person is on the KCR entering and exiting Lion Rock tunnel).
And for those that are wondering...no I don't work for HKBN. I just happen to be a satisfied customer.
Oh. One more thing...if you sign up for the softphone and want to BYOD, don't expect any support from HKBN's tech support. They will not provide any support for any softphone/hardphone/ATA except their Nortel IAD and their Nortel Multimedia PC Client. I had to reverse everything by watching the SIP conversations between the PC client and HKBN.
See ya...
d.c. |
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nc22
join:2004-04-16
Everett, WA | reply to im_chandave
HKBN is good and cheap, however, I want to get more from it by using my sipura 2000, however, it was not successful, and anyone can post a full working config for sipura 2000? (it keeps failed to register)
Thanks. |
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im_chandave
join:2005-07-28
Cleveland, OH
| I just helped out another fellow bbr member get his SPA-1001 working with HKBN. Here are the settings I suggested:
Line 1 > Proxy and Registration
Proxy: s2hkbntel.net
Use Outbound Proxy: yes
Outbound Proxy: s21.hkbntel.net
Use OB Proxy In Dialog: yes
Register Expires: 120
Use DNS SRV: no
Line 1 > Subscriber Information
User ID: HKBN phone number
Password: Your softphone/Web password
One thing that might be a problem is that s2hkbntel.net does not resolve to a real domain so I'm not sure what the SPA will do. We resolve it by setting up a DNS server to service that domainname. We then pointed the SPA to use that DNS server for the Primary DNS Server.
Additionally, if you are behind a NAT/Firewall, turn on STUN support, VIA header support, and most of all, "Send Resp To Src Port" = "yes".
If you noticed that you cannot SIP REGISTER, then the problem is that your previous SIP REGISTER is blocking you from re-REGISTERing. HKBN's "2b" SIP Registrar does not support multiple simultaneous registrations of different devices. You will need to go to your Personal Assistance website »pa.2b.com.hk and "2b Reset" your SIP registration.
If you register successfully but get "486 Busy Here" messages when you try calling into your HKBN phone number, then the problem might be anonymous blocking is enabled on your SPA. We tried turnning off the Anonymous Block server on the SPA but it still would not allow the call into the SPA. Still trying to resolve this problem.
Anonymous Calls will occur when someone in HK dials your phone number with a "133" prefix. It can also happen for an incoming International Direct Dial call (i.e. someone from outside of HK calling your phone number).
See ya...
d.c. |
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nc22
join:2004-04-16
Everett, WA
| Thanks a lot ChanDave, your info was so useful, I can register now after I logout manually on hkbn web page,
however, after I set it up, I could only call somebody, if someone called this sip, my phone was ringing too but when I answered the call, the call dropped and was cut off, and the caller at another side never heard my phone picked up. What's the setting problem, why I can call out but can not answer the phone? Do I need to enter a stun server? |
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nc22
join:2004-04-16
Everett, WA | reply to im_chandave
also you mentioned many times we should use msc1.hkbntel.net, why sipura need s2hkbntel.net instead? |
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hkvalet
Specialist
join:2004-10-18
Las Vegas, NV
| said by nc22 :also you mentioned many times we should use msc1.hkbntel.net, why sipura need s2hkbntel.net instead? If you signed up at www.2b.com.hk, you will need s2hkbntel.net (Proxy) & s21.hkbntel.net (Outbound Proxy).
If you signed up at www.hkbntel.net, you will need mcs1.hkbn.net for both Proxy.
--
**Students are the CUSTOMERS of teachers** »RateMyProfessors.com |
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nc22
join:2004-04-16
Everett, WA
| Thanks a lot experts ChanDave and HKVALET, if anyone could solve the call-in calls dropping problem, please let every of us know, I tried many combinations in my SPA, and still can answer any calls (Call somebody is fine using ChanDave's suggestion) |
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im_chandave
join:2005-07-28
Cleveland, OH
| reply to nc22
said by nc22 :Do I need to enter a stun server? Turning on all the VIA supports will help you register and get and initiate calls through HKBN. Adding STUN will allow the audio channels to get past your firewall/NAT system that's between your ATA and HKBN. Yes, you need to fill in the STUN server input box if you plan on using STUN. You can find a list of public STUN servers at »www.voip-info.org/wiki/view/STUN.
In order for STUN to work, you need to enable "Gaming Mode" on your firewall/router/NAT. I'm not sure the exact terminology used by your firewall/router/NAT vendor. Also, from what I gather from another post, your "Gaming Mode" might be triggered by enabling UPnP (strange....).
See ya...
d.c. |
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gkong
join:2001-10-06
Edmond, OK
·Cox HSI
| reply to im_chandave
im_chandave: I tried using somebody phone with Caller ID and the SPA still sends a 486 Busy Here signal. Maybe with so many of us are having problem, the solution is not available until Sipura/Linksys fix some SIP handling problem with Nortel MCS server in their new firmware, Thanks for helping us out! |
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im_chandave
join:2005-07-28
Cleveland, OH | 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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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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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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im_chandave
join:2005-07-28
Cleveland, OH
| 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
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
| 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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cplus
join:2005-09-30
Fullerton, CA
| 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
| 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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