This post is intended to collect and share info on how to configure HKBN 2B in the Obi110 or Obi100.
I could get the device to register, but I will get an error message code "500" when I tried to dial out.
There is no ringing on my side when someone is calling me...
My router is Asus RT-N16. It is flashed with Tamagochi. I have forwarded 5060 and RTP ports in my router.
I have tried stun.xten.com and stun01.sipphone.com, stun.voipbuster.com, stun.ideasip.com.
I am using the following dial plan:
([2-9]xxxxxx|011xx.|1[2-9]xx[2-9]xxxxxx|<852:>[236789]xxxxxxxS0|[236789]xxxxxxxS0|1850xS0|188xS0)
My OBi110 is running 1.2.1 (Build: 2103).
Is there anyway we could have debug log for this device?
Could anyone help?
Thank you!
Below shows my settings:
(http://i54.tinypic.com/2dl1x6q.png)
(http://i52.tinypic.com/2hcngw0.png)
(http://i52.tinypic.com/2d95qmr.png)
(http://i51.tinypic.com/2ekrzp3.png)
(http://i53.tinypic.com/2i8vcc4.png)
(http://i55.tinypic.com/1zr0378.png)
Below shows the log
I tried to call a number as you could see the "OFF HOOK" Line
After dialing, an error message saying it is rejected by the service provider Code "500"
Thank you!!!!!!!!!!
<7> sendto cb505987:5060(842)
192.168.1.127 10/04 01:40:57.746
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:40:57.746
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:40:57.746
<7> TCP:Accept OK (HTTPD)
192.168.1.127 10/04 01:41:02.073
<145> Reboot is scheduled in 1 seconds
192.168.1.127 10/04 01:41:02.075
<7> TCP:Accept OK (HTTPD)
192.168.1.127 10/04 01:41:02.115
<7> TCP:Accept OK (HTTPD)
192.168.1.127 10/04 01:41:02.116
<0> Reboot checking..... 192.168.1.127 10/04 01:41:03.069
<143> Compare the version now
192.168.1.127 10/04 01:41:03.070
<0> Final Cleanup before reboot....
192.168.1.127 10/04 01:41:03.070
<144> Goodbye! Reboot Now. (reason: 9)
192.168.1.127 10/04 01:41:04.070
<6> ==== Networking is ready ====
192.168.1.127 10/04 01:41:10.274
<0> IP Address= 192.168.1.127 192.168.1.127 10/04 01:41:10.280
<0> Gateway = 192.168.1.1 192.168.1.127 10/04 01:41:10.281
<0> Netmask = 255.255.255.0 192.168.1.127 10/04 01:41:10.283
<3> SYSTEM REBOOTED (Reason: 9, lifecycle: 1496)
192.168.1.127 10/04 01:41:10.284
<0> SLIC_init ... 192.168.1.127 10/04 01:41:10.329
<0> Reset SLIC... 192.168.1.127 10/04 01:41:10.331
<150> Setup Provisioning for system start! 1200
192.168.1.127 10/04 01:41:10.342
<0> SLIC & DAA is initialized 192.168.1.127 10/04 01:41:12.000
<6> Start Main Service Now 192.168.1.127 10/04 01:41:14.052
<7> Voice Main
192.168.1.127 10/04 01:41:14.053
<7> [CPT] --- FXS s/w tone generator (0) ---
192.168.1.127 10/04 01:41:14.059
<7> BASESSL:load cert:5
192.168.1.127 10/04 01:41:14.162
<7> BASESSL:Load certificate ok
192.168.1.127 10/04 01:41:14.167
<7> sendto cb505987:5060(569)
192.168.1.127 10/04 01:41:14.452
<7> XMPP:Invalid cfg use for xmpp
192.168.1.127 10/04 01:41:14.453
<7> GTT:xmpp not configured
192.168.1.127 10/04 01:41:14.456
<7> XMPP:Invalid cfg use for xmpp
192.168.1.127 10/04 01:41:14.457
<7> GTT:xmpp not configured
192.168.1.127 10/04 01:41:14.458
<7> [SLIC] FAX MODE --- OFF
192.168.1.127 10/04 01:41:15.392
<7> [SLIC]:Slic#0 ON HOOK
192.168.1.127 10/04 01:41:15.394
<7> sendto cb505987:5060(569)
192.168.1.127 10/04 01:41:15.454
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:41:15.456
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:41:15.458
<7> sendto cb505987:5060(837)
192.168.1.127 10/04 01:41:15.472
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:41:15.743
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:41:15.761
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:41:15.801
<7> sendto cb505987:5060(285)
192.168.1.127 10/04 01:41:15.803
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:41:15.805
<7> [CPT] --- FXS s/w tone generator (7133) ---
192.168.1.127 10/04 01:41:21.194
<147> PROV: Auto Update invalid server name()
192.168.1.127 10/04 01:41:22.344
<7> [CPT] tone compression done - 12283 !!
192.168.1.127 10/04 01:41:26.342
<7> sendto cb505987:5060(843)
192.168.1.127 10/04 01:42:15.819
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:42:16.102
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:42:16.108
<7> [PR] prompt transcoding done!!
192.168.1.127 10/04 01:43:03.707
<7> [SLIC]:Slic#0 OFF HOOK
192.168.1.127 10/04 01:43:10.202
<7> [CPT] --- FXS SiLab h/w tone generator ---
192.168.1.127 10/04 01:43:10.203
<7> sendto cb505987:5060(843)
192.168.1.127 10/04 01:43:12.123
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:43:12.414
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:43:12.443
<7> [DSP]: ---- H/W DTMF ON (level:1) : * @ 119320 ms---- 192.168.1.127 10/04 01:43:13.373
<7> [DSP] ---- H/W DTMF OFF @ 119470 ms ---- 192.168.1.127 10/04 01:43:13.522
<7> [DSP]: ---- H/W DTMF ON (level:1) : * @ 119640 ms---- 192.168.1.127 10/04 01:43:13.693
<7> [DSP] ---- H/W DTMF OFF @ 119790 ms ---- 192.168.1.127 10/04 01:43:13.843
<7> [DSP]: ---- H/W DTMF ON (level:1) : 1 @ 120320 ms---- 192.168.1.127 10/04 01:43:14.373
<7> [DSP] ---- H/W DTMF OFF @ 120470 ms ---- 192.168.1.127 10/04 01:43:14.522
<7> [DSP]: ---- H/W DTMF ON (level:1) : 2 @ 122660 ms---- 192.168.1.127 10/04 01:43:16.712
<7> [DSP] ---- H/W DTMF OFF @ 122810 ms ---- 192.168.1.127 10/04 01:43:16.862
<7> [DSP]: ---- H/W DTMF ON (level:1) : 4 @ 123000 ms---- 192.168.1.127 10/04 01:43:17.052
<7> [DSP] ---- H/W DTMF OFF @ 123150 ms ---- 192.168.1.127 10/04 01:43:17.202
<7> [DSP]: ---- H/W DTMF ON (level:1) : 5 @ 123360 ms---- 192.168.1.127 10/04 01:43:17.412
<7> [DSP] ---- H/W DTMF OFF @ 123510 ms ---- 192.168.1.127 10/04 01:43:17.563
<7> [DSP]: ---- H/W DTMF ON (level:1) : 5 @ 123720 ms---- 192.168.1.127 10/04 01:43:17.772
<7> [DSP] ---- H/W DTMF OFF @ 123870 ms ---- 192.168.1.127 10/04 01:43:17.922
<7> [DSP]: ---- H/W DTMF ON (level:1) : 5 @ 124040 ms---- 192.168.1.127 10/04 01:43:18.092
<7> [DSP] ---- H/W DTMF OFF @ 124190 ms ---- 192.168.1.127 10/04 01:43:18.242
<7> [DSP]: ---- H/W DTMF ON (level:1) : 1 @ 124360 ms---- 192.168.1.127 10/04 01:43:18.412
<7> [DSP] ---- H/W DTMF OFF @ 124510 ms ---- 192.168.1.127 10/04 01:43:18.562
<7> [DSP]: ---- H/W DTMF ON (level:1) : 2 @ 124650 ms---- 192.168.1.127 10/04 01:43:18.703
<7> [DSP] ---- H/W DTMF OFF @ 124810 ms ---- 192.168.1.127 10/04 01:43:18.862
<7> [DSP]: ---- H/W DTMF ON (level:1) : 6 @ 125040 ms---- 192.168.1.127 10/04 01:43:19.092
<7> [DSP] ---- H/W DTMF OFF @ 125190 ms ---- 192.168.1.127 10/04 01:43:19.243
<7> [DSP]: ---- H/W DTMF ON (level:3) : # @ 125480 ms---- 192.168.1.127 10/04 01:43:19.532
<7> [DSP] ---- H/W DTMF OFF @ 125610 ms ---- 192.168.1.127 10/04 01:43:19.663
<142> PARAM Cache Write Back(256 bytes) 192.168.1.127 10/04 01:43:20.084
<7> STUNC:ERR=TO,mtd=1
192.168.1.127 10/04 01:43:58.754
<7> sendto cb505987:5060(952)
192.168.1.127 10/04 01:43:58.757
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:43:59.038
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:43:59.043
<7> sendto cb505987:5060(358)
192.168.1.127 10/04 01:43:59.044
<7> RTP:Del Channel
192.168.1.127 10/04 01:43:59.046
<7> [CPT] --- FXS SiLab h/w tone generator ---
192.168.1.127 10/04 01:43:59.055
<7> [CPT] --- FXS s/w tone generator (165005) ---
192.168.1.127 10/04 01:43:59.056
<7> sendto cb505987:5060(843)
192.168.1.127 10/04 01:44:12.460
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:44:12.752
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:44:12.756
<7> [SLIC] FAX MODE --- OFF
192.168.1.127 10/04 01:44:17.540
<7> [SLIC]:Slic#0 ONHOOK
192.168.1.127 10/04 01:44:17.541
<7> [SLIC]:Slic#0 OFF HOOK
192.168.1.127 10/04 01:44:21.740
<7> [CPT] --- FXS SiLab h/w tone generator ---
192.168.1.127 10/04 01:44:21.742
____________________________________________________________
2011-04-11
We finally made some progress after OBIHAI Support suggested to turn off X_InsertRemotePartyID.
I could receive call and dial out. However I got cut off every 10 mins.
Does it have to do with X_KeepAliveExpires or RegistrationPeriod?
(http://i53.tinypic.com/2hhdifl.png)
baor,
1. What kind of home router you have?
You can check whether the router has a SIP ALG function.
If it has, try TURNING OFF the feature and call again.
2. Make sure the number you try to call is acceptable by HKBN.
Have you tried to call the same number from another
device or softphone with HKBN? and successful?
3. We can also check your device configuration remotely. For privacy you can
email support@obihai.com and provide your OBi number (printed on the back of the unit),
and we can provide you further assistance.
SIP error code 500 is : Internal Server Error.
You have a STUN server configured : stun01.sipphone.com
This STUN server was shut down a couple of weeks ago and will not return. Unless the current firmware has changed the timeout, you will experience a 40 second delay on each call you attempt with a non-existent STUN server.
In fact I am one of HKBN 2b users in Hong Kong. I have been able to install it in my SPA3000, PAP2T and Asterisk server. The settings of HKBN 2b is given by
http://www.hkepc.com/forum/viewthread.php?tid=1141462&page=218#pid18904520 (http://www.hkepc.com/forum/viewthread.php?tid=1141462&page=218#pid18904520)
I have already tried several times to register the 2b server with my OBi110 in Shenzhen but still failed. I believe the information provided in the above link should provide a good reference of how to convert the settings in Linksys SPA to OBi devices.
YH
I just reconnected my old Linksys PAP2T ATA with the HKBN 2b settings in LINE 1 tab. The HKBN 2b settings in the PAP2T works very well. It can make HK PSTN outbound call and receive HK PSTN call with caller ID shown on the phone attached to it.
My network configuration is shown as follows:
Interent --> China Telecom --> modem --> Router (Draytek 2200E Plus) --> i) PCs, ii) Asterisk, iii) SPA devices including SPA941, PAP2T, iv) OBi110.
Based on the information shown in the above post, the PAP2T, configured to connect HKBN 2b server, can make and receive call without any problem encounter. Please note that I do not need to set any UDP ports forwarding (5060 and RTP number range) to my PAP2T.
Also based on my pass experience of using Linksys ATA, the settings for HKBN 2b in PAP2T which can make the SIP client be visible from the WAN port. That's to say that the SIP client looks like sitting in the fornt of the WAN port of the router.
So back to the OBi110 or OBi100, how can we make the internal SIP client (behind router) be in front of the WAN port of the router although it is connected to the LAN port of the router?
YH
Quote from: obi-support2 on April 09, 2011, 06:52:51 PM
baor,
1. What kind of home router you have?
You can check whether the router has a SIP ALG function.
If it has, try TURNING OFF the feature and call again.
2. Make sure the number you try to call is acceptable by HKBN.
Have you tried to call the same number from another
device or softphone with HKBN? and successful?
3. We can also check your device configuration remotely. For privacy you can
email support@obihai.com and provide your OBi number (printed on the back of the unit),
and we can provide you further assistance.
1. My router is Asus RT-N16. It is flashed with Tamagochi.
I will post some of my router setting. I don't see the "SIP ALG".
2. I tried to call weather report, which should be a valid HK number (1878-200)
3. Nice. Do I have to put a reference number (ticket number) in my email? So I could get your attention and support?
Quote from: RonR on April 09, 2011, 06:59:16 PM
SIP error code 500 is : Internal Server Error.
You have a STUN server configured : stun01.sipphone.com
This STUN server was shut down a couple of weeks ago and will not return. Unless the current firmware has changed the timeout, you will experience a 40 second delay on each call you attempt with a non-existent STUN server.
I have tried stun01.sipphone.com and stun.xten.com.
I use the # key at the end of dial.
The error code is immediately if I called 1878-200.
Dialing regular phone number will have the 40 seconds delay you talked about. Does it mean stun.xten.com is closed?
Quote from: yhfung on April 09, 2011, 07:01:06 PM
In fact I am one of HKBN 2b users in Hong Kong. I have been able to install it in my SPA3000, PAP2T and Asterisk server. The settings of HKBN 2b is given by
http://www.hkepc.com/forum/viewthread.php?tid=1141462&page=218#pid18904520 (http://www.hkepc.com/forum/viewthread.php?tid=1141462&page=218#pid18904520)
I have already tried several times to register the 2b server with my OBi110 in Shenzhen but still failed. I believe the information provided in the above link should provide a good reference of how to convert the settings in Linksys SPA to OBi devices.
YH
I also have a Linksys PAP (original Vonage version, Unprovisioned?) and I could use my PAP fine with HKBN. I also know how to reset the registration as HKBN allows only
1 register.
There were some settings I could not find in the OBi110.
【NAT Settings】Heading
NAT Mapping Enable: yes NAT Keep Alive Enable: yes
NAT Keep Alive Msg: $PING
【RTP Parameters 】Heading
RTP Packet Size: 0.020
【NAT Support Parameters】Heading
Handle VIA received: yes Handle VIA rport: yes
Insert VIA received: yes Insert VIA rport: yes
Substitute VIA Addr: yes Send Resp To Src Port: yes
STUN Enable: yes STUN Test Enable: yes
STUN Server: stun.xten.com
Is there anyway we could have debug log for this device?
Quote from: baor on April 09, 2011, 08:33:40 PMDialing regular phone number will have the 40 seconds delay you talked about. Does it mean stun.xten.com is closed?
I don't know the status of stun.xten.com, but I've always avoided it as it's often down. I just know that the OBi has a 40 second timeout on STUN server responses, so if it's not there, you wait 40 seconds before the OBi resumes processing of the call. I reported the problem to support via email back on 3/22 but got no reply and it was also discussed here in the forum, but apparently no action was taken.
I'm currently using stun.ideasip.com with no problems.
Quote from: baor on April 09, 2011, 08:43:12 PM
Is there anyway we could have debug log for this device?
There is SYSLOG capability at:
System Management -> Device Admin
You can run a syslog server on your PC.
On the OBi, under SP1/SP2 Service, enable X_SipDebugOption.
Under Device Admin, point Debug Server and port to the IP address and
port of the PC where you run your syslog server.
This could be a SIP Interop issue. May be HKBN does not like our INVITE request.
Please email your capture log file to support@obihai.com and we should be able
to figure it out and do something about it.
To help you remotely trouble shoot the problem, all we need is your 9-digit OBI number;
and you must add your device to OBiTALK if you haven't done so yet.
Thank you.
Quote from: RonR on April 09, 2011, 08:46:58 PM
Quote from: baor on April 09, 2011, 08:33:40 PMDialing regular phone number will have the 40 seconds delay you talked about. Does it mean stun.xten.com is closed?
I don't know the status of stun.xten.com, but I've always avoided it as it's often down. I just know that the OBi has a 40 second timeout on STUN server responses, so if it's not there, you wait 40 seconds before the OBi resumes processing of the call. I reported the problem to support via email back on 3/22 but got no reply and it was also discussed here in the forum, but apparently no action was taken.
I'm currently using stun.ideasip.com with no problems.
I have tried stun.xten.com and stun01.sipphone.com, stun.voipbuster.com, stun.ideasip.com.
Still could not dial out.
Quote from: obi-support2 on April 09, 2011, 08:51:01 PM
You can run a syslog server on your PC.
On the OBi, under SP1/SP2 Service, enable X_SipDebugOption.
Under Device Admin, point Debug Server and port to the IP address and
port of the PC where you run your syslog server.
This could be a SIP Interop issue. May be HKBN does not like our INVITE request.
Please email your capture log file to support@obihai.com and we should be able
to figure it out and do something about it.
To help you remotely trouble shoot the problem, all we need is your 9-digit OBI number;
and you must add your device to OBiTALK if you haven't done so yet.
Thank you.
IF I added the device, it will erase all of my settings.
I tried to use the portal in the very beginning. However, it would not register.
I could get it to register using manual settings......
I have no experience with syslog server. Any easy tutorial?
baor,
Hi
Here are 2 free Windows syslog servers I have used in the past. They are easy to use.
http://tftpd32.jounin.net/
http://kin.klever.net/pumpkin
Quote from: obi-support2 on April 09, 2011, 08:51:01 PM
You can run a syslog server on your PC.
On the OBi, under SP1/SP2 Service, enable X_SipDebugOption.
Under Device Admin, point Debug Server and port to the IP address and
port of the PC where you run your syslog server.
This could be a SIP Interop issue. May be HKBN does not like our INVITE request.
Please email your capture log file to support@obihai.com and we should be able
to figure it out and do something about it.
To help you remotely trouble shoot the problem, all we need is your 9-digit OBI number;
and you must add your device to OBiTALK if you haven't done so yet.
Thank you.
I made the settings and reboot.
I tried the TFTPD software, nothing is going on???
(http://i52.tinypic.com/sl03zp.png)
(http://i53.tinypic.com/4qn8ch.png)
(http://i54.tinypic.com/2zso1zd.png)
What should I do now?
baor,
Hi
You may need to port forward port 514 in your router to the IP of the server. I just made a test on my computer. You should see activity that the OBi is generating.
<See Image>
(//)
Quote from: QBZappy on April 09, 2011, 10:27:47 PM
baor,
Hi
You may need to port forward port 514 in your router to the IP of the server. I just made a test on my computer. You should see activity that the OBi is generating.
<See Image>
(//)
Could you how you set the software? Do I have to set the DHCP server? How?
baor,
What is your router model name and number?
EDIT: Can you log on the the web page of your router?
Quote from: QBZappy on April 09, 2011, 10:30:58 PM
baor,
What is your router model name and number?
My router is Asus RT-N16. It is flashed with Tamagochi.
Do you have MSN or other Instant messeger?
It is all internal, why do I have to forward port????
If the OBi and the PC the Syslog server is running are on the same LAN, no port forwarding or anything else is required. Just fire up the Syslog server on the PC and try to make a call on the OBi. The Syslog server should record all the events. You can email the resulting log file for analysis.
baor,
No port forwarding should be necessary as Ron suggested. I just checked my router settings.
Quote from: RonR on April 09, 2011, 10:36:24 PM
If the OBi and the PC the Syslog server is running are on the same LAN, no port forwarding or anything else is required. Just fire up the Syslog server on the PC and try to make a call on the OBi. The Syslog server should record all the events. You can email the resulting log file for analysis.
It is working now when I am using the 32bit version....
baor,
I had noticed that on your image capture. I assumed that you had installed the correct version.
Off topic. By the way what is Tamagochi firmware? I could not find anything on Google.
Below shows the log
I tried to call a number as you could see the "OFF HOOK" Line
After dialing, an error message saying it is rejected by the service provide Code "500"
Thank you!!!!!!!!!!
<7> sendto cb505987:5060(842)
192.168.1.127 10/04 01:40:57.746
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:40:57.746
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:40:57.746
<7> TCP:Accept OK (HTTPD)
192.168.1.127 10/04 01:41:02.073
<145> Reboot is scheduled in 1 seconds
192.168.1.127 10/04 01:41:02.075
<7> TCP:Accept OK (HTTPD)
192.168.1.127 10/04 01:41:02.115
<7> TCP:Accept OK (HTTPD)
192.168.1.127 10/04 01:41:02.116
<0> Reboot checking..... 192.168.1.127 10/04 01:41:03.069
<143> Compare the version now
192.168.1.127 10/04 01:41:03.070
<0> Final Cleanup before reboot....
192.168.1.127 10/04 01:41:03.070
<144> Goodbye! Reboot Now. (reason: 9)
192.168.1.127 10/04 01:41:04.070
<6> ==== Networking is ready ====
192.168.1.127 10/04 01:41:10.274
<0> IP Address= 192.168.1.127 192.168.1.127 10/04 01:41:10.280
<0> Gateway = 192.168.1.1 192.168.1.127 10/04 01:41:10.281
<0> Netmask = 255.255.255.0 192.168.1.127 10/04 01:41:10.283
<3> SYSTEM REBOOTED (Reason: 9, lifecycle: 1496)
192.168.1.127 10/04 01:41:10.284
<0> SLIC_init ... 192.168.1.127 10/04 01:41:10.329
<0> Reset SLIC... 192.168.1.127 10/04 01:41:10.331
<150> Setup Provisioning for system start! 1200
192.168.1.127 10/04 01:41:10.342
<0> SLIC & DAA is initialized 192.168.1.127 10/04 01:41:12.000
<6> Start Main Service Now 192.168.1.127 10/04 01:41:14.052
<7> Voice Main
192.168.1.127 10/04 01:41:14.053
<7> [CPT] --- FXS s/w tone generator (0) ---
192.168.1.127 10/04 01:41:14.059
<7> BASESSL:load cert:5
192.168.1.127 10/04 01:41:14.162
<7> BASESSL:Load certificate ok
192.168.1.127 10/04 01:41:14.167
<7> sendto cb505987:5060(569)
192.168.1.127 10/04 01:41:14.452
<7> XMPP:Invalid cfg use for xmpp
192.168.1.127 10/04 01:41:14.453
<7> GTT:xmpp not configured
192.168.1.127 10/04 01:41:14.456
<7> XMPP:Invalid cfg use for xmpp
192.168.1.127 10/04 01:41:14.457
<7> GTT:xmpp not configured
192.168.1.127 10/04 01:41:14.458
<7> [SLIC] FAX MODE --- OFF
192.168.1.127 10/04 01:41:15.392
<7> [SLIC]:Slic#0 ON HOOK
192.168.1.127 10/04 01:41:15.394
<7> sendto cb505987:5060(569)
192.168.1.127 10/04 01:41:15.454
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:41:15.456
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:41:15.458
<7> sendto cb505987:5060(837)
192.168.1.127 10/04 01:41:15.472
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:41:15.743
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:41:15.761
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:41:15.801
<7> sendto cb505987:5060(285)
192.168.1.127 10/04 01:41:15.803
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:41:15.805
<7> [CPT] --- FXS s/w tone generator (7133) ---
192.168.1.127 10/04 01:41:21.194
<147> PROV: Auto Update invalid server name()
192.168.1.127 10/04 01:41:22.344
<7> [CPT] tone compression done - 12283 !!
192.168.1.127 10/04 01:41:26.342
<7> sendto cb505987:5060(843)
192.168.1.127 10/04 01:42:15.819
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:42:16.102
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:42:16.108
<7> [PR] prompt transcoding done!!
192.168.1.127 10/04 01:43:03.707
<7> [SLIC]:Slic#0 OFF HOOK
192.168.1.127 10/04 01:43:10.202
<7> [CPT] --- FXS SiLab h/w tone generator ---
192.168.1.127 10/04 01:43:10.203
<7> sendto cb505987:5060(843)
192.168.1.127 10/04 01:43:12.123
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:43:12.414
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:43:12.443
<7> [DSP]: ---- H/W DTMF ON (level:1) : * @ 119320 ms---- 192.168.1.127 10/04 01:43:13.373
<7> [DSP] ---- H/W DTMF OFF @ 119470 ms ---- 192.168.1.127 10/04 01:43:13.522
<7> [DSP]: ---- H/W DTMF ON (level:1) : * @ 119640 ms---- 192.168.1.127 10/04 01:43:13.693
<7> [DSP] ---- H/W DTMF OFF @ 119790 ms ---- 192.168.1.127 10/04 01:43:13.843
<7> [DSP]: ---- H/W DTMF ON (level:1) : 1 @ 120320 ms---- 192.168.1.127 10/04 01:43:14.373
<7> [DSP] ---- H/W DTMF OFF @ 120470 ms ---- 192.168.1.127 10/04 01:43:14.522
<7> [DSP]: ---- H/W DTMF ON (level:1) : 2 @ 122660 ms---- 192.168.1.127 10/04 01:43:16.712
<7> [DSP] ---- H/W DTMF OFF @ 122810 ms ---- 192.168.1.127 10/04 01:43:16.862
<7> [DSP]: ---- H/W DTMF ON (level:1) : 4 @ 123000 ms---- 192.168.1.127 10/04 01:43:17.052
<7> [DSP] ---- H/W DTMF OFF @ 123150 ms ---- 192.168.1.127 10/04 01:43:17.202
<7> [DSP]: ---- H/W DTMF ON (level:1) : 5 @ 123360 ms---- 192.168.1.127 10/04 01:43:17.412
<7> [DSP] ---- H/W DTMF OFF @ 123510 ms ---- 192.168.1.127 10/04 01:43:17.563
<7> [DSP]: ---- H/W DTMF ON (level:1) : 5 @ 123720 ms---- 192.168.1.127 10/04 01:43:17.772
<7> [DSP] ---- H/W DTMF OFF @ 123870 ms ---- 192.168.1.127 10/04 01:43:17.922
<7> [DSP]: ---- H/W DTMF ON (level:1) : 5 @ 124040 ms---- 192.168.1.127 10/04 01:43:18.092
<7> [DSP] ---- H/W DTMF OFF @ 124190 ms ---- 192.168.1.127 10/04 01:43:18.242
<7> [DSP]: ---- H/W DTMF ON (level:1) : 1 @ 124360 ms---- 192.168.1.127 10/04 01:43:18.412
<7> [DSP] ---- H/W DTMF OFF @ 124510 ms ---- 192.168.1.127 10/04 01:43:18.562
<7> [DSP]: ---- H/W DTMF ON (level:1) : 2 @ 124650 ms---- 192.168.1.127 10/04 01:43:18.703
<7> [DSP] ---- H/W DTMF OFF @ 124810 ms ---- 192.168.1.127 10/04 01:43:18.862
<7> [DSP]: ---- H/W DTMF ON (level:1) : 6 @ 125040 ms---- 192.168.1.127 10/04 01:43:19.092
<7> [DSP] ---- H/W DTMF OFF @ 125190 ms ---- 192.168.1.127 10/04 01:43:19.243
<7> [DSP]: ---- H/W DTMF ON (level:3) : # @ 125480 ms---- 192.168.1.127 10/04 01:43:19.532
<7> [DSP] ---- H/W DTMF OFF @ 125610 ms ---- 192.168.1.127 10/04 01:43:19.663
<142> PARAM Cache Write Back(256 bytes) 192.168.1.127 10/04 01:43:20.084
<7> STUNC:ERR=TO,mtd=1
192.168.1.127 10/04 01:43:58.754
<7> sendto cb505987:5060(952)
192.168.1.127 10/04 01:43:58.757
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:43:59.038
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:43:59.043
<7> sendto cb505987:5060(358)
192.168.1.127 10/04 01:43:59.044
<7> RTP:Del Channel
192.168.1.127 10/04 01:43:59.046
<7> [CPT] --- FXS SiLab h/w tone generator ---
192.168.1.127 10/04 01:43:59.055
<7> [CPT] --- FXS s/w tone generator (165005) ---
192.168.1.127 10/04 01:43:59.056
<7> sendto cb505987:5060(843)
192.168.1.127 10/04 01:44:12.460
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:44:12.752
<7> RxFrom:cb505987:5060
192.168.1.127 10/04 01:44:12.756
<7> [SLIC] FAX MODE --- OFF
192.168.1.127 10/04 01:44:17.540
<7> [SLIC]:Slic#0 ONHOOK
192.168.1.127 10/04 01:44:17.541
<7> [SLIC]:Slic#0 OFF HOOK
192.168.1.127 10/04 01:44:21.740
<7> [CPT] --- FXS SiLab h/w tone generator ---
192.168.1.127 10/04 01:44:21.742
Is it related to these lines?
<7> XMPP:Invalid cfg use for xmpp
192.168.1.127 10/04 01:41:14.453
<7> GTT:xmpp not configured
192.168.1.127 10/04 01:41:14.456
<7> XMPP:Invalid cfg use for xmpp
192.168.1.127 10/04 01:41:14.457
<7> GTT:xmpp not configured
baor,
Ah, Tamagochi firmware = Tomato firmware.
There are a few on this forum I have noticed using this firmware. It seems that we both are using Asus RT N-16 router+Tomato firmware.
If you're into that router and firmware, you gotta check out this site. These guys have simplified the installation of Optware on this router model. They have an automated install of Asterisk, web servers, bit torrent clients, etc..
http://www.xtremecoders.org/forums/f78/tomato-optware-package-valerakvb-ver-11-4-a-169/
Thanks
baor,
Quote from: baor on April 09, 2011, 10:56:25 PM
Is it related to these lines?
<7> XMPP:Invalid cfg use for xmpp
192.168.1.127 10/04 01:41:14.453
<7> GTT:xmpp not configured
192.168.1.127 10/04 01:41:14.456
<7> XMPP:Invalid cfg use for xmpp
192.168.1.127 10/04 01:41:14.457
<7> GTT:xmpp not configured
XMPP are references to Google Talk. I see from your screen capture you have only a SIP account configured. I just remove my Google Voice account and this is my syslog. I think it just means you have not setup the GV account. This should not be the problem you have.
<144> Goodbye! Reboot Now. (reason: 4)
172.16.240.52 10/04 02:16:29.023
<6> ==== Networking is ready ====
172.16.240.52 10/04 02:16:35.242
<7> XMPP:Invalid cfg use for xmpp
172.16.240.52 10/04 02:16:39.218
<7> GTT:xmpp not configured
172.16.240.52 10/04 02:16:39.222
<7> XMPP:Invalid cfg use for xmpp
172.16.240.52 10/04 02:16:39.225
<7> GTT:xmpp not configured
172.16.240.52 10/04 02:16:39.228
Quote from: QBZappy on April 09, 2011, 11:23:15 PM
baor,
Quote from: baor on April 09, 2011, 10:56:25 PM
Is it related to these lines?
<7> XMPP:Invalid cfg use for xmpp
192.168.1.127 10/04 01:41:14.453
<7> GTT:xmpp not configured
192.168.1.127 10/04 01:41:14.456
<7> XMPP:Invalid cfg use for xmpp
192.168.1.127 10/04 01:41:14.457
<7> GTT:xmpp not configured
XMPP are references to Google Talk. I see from your screen capture you have only a SIP account configured. I just remove my Google Voice account and this is my syslog. I think it just means you have not setup the GV account. This should not be the problem you have.
<144> Goodbye! Reboot Now. (reason: 4)
172.16.240.52 10/04 02:16:29.023
<6> ==== Networking is ready ====
172.16.240.52 10/04 02:16:35.242
<7> XMPP:Invalid cfg use for xmpp
172.16.240.52 10/04 02:16:39.218
<7> GTT:xmpp not configured
172.16.240.52 10/04 02:16:39.222
<7> XMPP:Invalid cfg use for xmpp
172.16.240.52 10/04 02:16:39.225
<7> GTT:xmpp not configured
172.16.240.52 10/04 02:16:39.228
I don't see the log revealing anything special????
Please help!
baor,
1) I have setup two different service providers on the OBi. I did not need to use Stun, are you certain you need it. Try removing Stun setting. Any change?
2) You have it registered, so it looks like you have the correct credentials. Go back to basics and double check if you port forwarded the SIP and RTP ports correctly to the OBi. In the router make sure they are the UDP ports being forwarded. I don't think this would give you Server error 500. The voice you hear on the phone is from the OBi I assume.
3) Do a hard reset by using a paper clip on the back of the OBi unit. Reconfigure everything over again.
3) After rechecking the basics and if it is still not resolved you need to send a syslog file to OBi support. Give them your OBi unit number and make sure you have registered on the portal so that they can look into your unit from their end.
Good luck.
(http://i54.tinypic.com/33jn7m9.png)
Not working...
I see this in your syslog:
<7> STUNC:ERR=TO,mtd=1
Quote from: QBZappy on April 10, 2011, 12:25:38 AM
I see this in your syslog:
<7> STUNC:ERR=TO,mtd=1
What does it mean?
That's what I see when you have a STUN server configured and it's unreachable for whatever reason. This is from my Syslog when stun01.sipphone.com was dismantled:
STUNC:ERR=TO,mtd=1
--------- 40 SECONDS OF NO ACTIVITY ---------
sendto 80ff79b:5060(1029)
I put in: --------- 40 SECONDS OF NO ACTIVITY ---------
baor,
It looks like problem might be associated with stun.
RonR,
Since I don't use Stun I'm not sure if you need to port forward the stun port to the OBi. What do you think?
STUN servers are used to determine one's public IP address. This is needed mostly for RTP, the audio side of things. The STUN server timeout problem in the OBi simply delays the outgoing call for 40 seconds, which can certainly confuse troubleshooting a problem significantly. I've never encountered a case where forwarding of port 3478 was required.
It's not clear why baor was using one to start with. When configuring a new provider, it's best to start out with the simplest, bare minimum, configuration possible, then build from there.
First of all I would like to let people know a little bit of HKBN 2b. Why do we need the service from HKBN 2b? This is because of HKBN 2b can provide the best VoIP service available in Hong Kong. It allows you to have mulitple inbound and outbound calls simultaneously. However the technical service provided by HKBN 2b is for SOFTPHONE ONLY. This is no support given for people who use either ATA or Asterisk.
Nevertheless some experts in the field of VoIP based on the packets from the softphone, they finally came up the solution of how to put HKBN 2b parameters into a Sipura/Linksys ATA box or an Asterisk box. You may find more details on the www.voip-info.org.
Sipura/Linksys ATA
If Sipura/Linksys ATAs are used, the settings for HKBN 2b is as described clearly in the followin link:
http://www.hkepc.com/forum/viewthread.php?tid=1141462&page=218#pid18904520
Asterisk
If Asterisk boxes are used, the settings for HKBN 2b is shown below:
1) Add the following and ip address in the /etc/hosts
203.80.89.135 s2hkbntel.net s21.hkbntel.net
(please note that other numbers or names may be required for other HKBN 2b numbers, #203.80.89.139 s2hkbntel.net s22.hkbntel.net)
2) In /etc/asterisk/sip.conf, we have
[general]
srvlookup=yes
nat=yes
realm=sip.hongkong.com
externhost=sip.hongkong.com
fromdomain=sip.hongkong.com
localnet=192.168.1.0/255.255.255.0 ;change it as per your Asterisk network address
externrefresh = 1
defaultexpirey=120
bindport=5060
qualify=yes
disallow=all
allow=ulaw,alaw,gsm
alwaysauthreject=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
context=front-desk
pedantic = yes
;to register the HKBN 2b server for inbound calls
register => 33445566hk:password@s2hkbntel.net:5060/33445566
[hkbn2b]
type=peer
username=33445566hk
secret=password
port=5060
host=s2hkbntel.net
fromuser=333445566hk
fromdomain=s2hkbntel.net
canreinvite=no
insecure=port,invite
dtmfmode=auto
context=from-hkbn
3) In /etc/asterisk/extensions.conf, we have
;HKBN 2b HK PSTN Trunk
[trunk_hk_pstn]
exten => _9XX.,1,Dial(SIP/${EXTEN:1}@hkbn2b,,r)
exten => _9XX.,n,Hangup()
[internal]
include => trunk_hk_pstn
.
.
.
.
.
;HKBN 2b dial-in
[from-hkbn]
exten => 33445566,1,Dial(SIP/2001,,r)
exten => 33445566,n,Hangup()
What do we (OBi users) want?
What is the settings in OBi devices such that we are able to use the HKBN 2b?
boar,
HKBN is using an old Nortel MCS server; and it seems to me that it
does not like the OBi's remote-party-id header.
Please try this work around:
Under your ITSP Profile (A or B), disable the option X_InsertRemotePartyID
(uncheck the box).
This should make your call go further.
Please also try to use a working STUN server.
Thank you.
Thank for obi-supprt2's response.
I have EXACTLY followed boar's instrudctions into my OBi110 box but still failed to register the HKBN 2b server.
Also I uncheced the box as your information, the result is still the same.
I wonder whether these is any missing information iin the first post.
YH
Quote from: obi-support2 on April 11, 2011, 01:50:54 PM
boar,
HKBN is using an old Nortel MCS server; and it seems to me that it
does not like the OBi's remote-party-id header.
Please try this work around:
Under your ITSP Profile (A or B), disable the option X_InsertRemotePartyID
(uncheck the box).
This should make your call go further.
Please also try to use a working STUN server.
Thank you.
Using your advice.
I think I got it to work. I am still testing it. Thank!
I will post my settings later, so other users would benefit.
We finally made some progress after OBIHAI Support suggested to turn off X_InsertRemotePartyID.
I could receive call and dial out. However I got cut off every 10 mins.
Does it have to do with X_KeepAliveExpires or RegistrationPeriod?
(http://i53.tinypic.com/2hhdifl.png)
Please explain what you meant by cut-off.
Is it no audio, but call still up?
or the Call is ended by the server after 10 min?
Exactly 10 minute every time?
From what city call to what city? HK -> HK?
Thank you.
Quote from: obi-support2 on April 11, 2011, 06:31:46 PM
Please explain what you meant by cut-off.
Is it no audio, but call still up?
or the Call is ended by the server after 10 min?
Exactly 10 minute every time?
From what city call to what city? HK -> HK?
Thank you.
At 10 mins, each side could only hear silence.
I guess Call is ended by the server.
HK -> HK
BTW, I wanted to post screen as attachment. But the 128K limit is too small.
Tinypic someday will delete my uploads and our records. So please consider increasing the limit to 200k.
Thank you!
Dude,
If you guys are testing my device, please let me know.
Could you add me in MSN or something?
It rebooted several times when I was using it.
This random thing scares the crap out of me?
If call is ended by the server, 2 things should happen:
1. Your call will disappear from the call status page (if you refresh the page)
2. If you don't hang up, you will hear some fast busy tone after 5s or so.
Did the last 2 happen?
Silence on both ends may not be a good indication the server has ended the call.
Or if the call still shows on the call status page (the one you attached),
what does it show for the number of packets sent and received?
If the call is still up, try this: hook flash on your phone, then hook flash again.
Does the audio resume?
Hey, I was trying the bridge function.
It does not work.
I use my cell phone to call the Obi110, enter my pin.
I could dial **9 222 222 222 to test the bridge.
However, when I tried to bridge with HKBN and call 18503 (weather report), I could not hear anything.....
Any idea?
(http://i51.tinypic.com/n5sb35.png)
Quote from: obi-support2 on April 11, 2011, 06:50:17 PM
If call is ended by the server, 2 things should happen:
1. Your call will disappear from the call status page (if you refresh the page)
2. If you don't hang up, you will hear some fast busy tone after 5s or so.
Did the last 2 happen?
Silence on both ends may not be a good indication the server has ended the call.
Or if the call still shows on the call status page (the one you attached),
what does it show for the number of packets sent and received?
If the call is still up, try this: hook flash on your phone, then hook flash again.
Does the audio resume?
Do you have a HK number? I could call you.
Hey, you reminded me something interesting.
I was talking to someone. At around 10 mins time frame, she said she could not hear from me. So she decided to flash and call me again. Her call hooked right back to my end while I was still talking to the phone (I didn't know it was dead on her end).... Does it help?
baor,
Hi
Just a suggestion. Better to send separate images in separate posts than loose them on tinypic, and loose the trail of information.
Quote from: QBZappy on April 11, 2011, 07:09:28 PM
baor,
Hi
Just a suggestion. Better to send separate images in separate posts than loose them on tinypic, and loose the trail of information.
>:( Come one, how come they could not up the size?
It is easy to exceed 128K with image, which is better than a thousand words.
I am just trying to help...
I think server space should be cheap enough when you are running a real business.
Sorry about the rant.... ;D
I am not sure if Obitalk Support team is testing my OBi110 or what.
My Obi110 is rebooting randomly????
baor,
Off topic:
I use this to resize my images. You probably can reduce that image to under 128K and still be usable. I have done it. It works. It is Freeware.
http://www.faststone.org/FSViewerDetail.htm
Quote from: QBZappy on April 11, 2011, 07:21:55 PM
baor,
Off topic:
I use this to resize my images. You probably can reduce that image to under 128K and still be usable. I have done it. It works.
http://www.faststone.org/FSViewerDetail.htm
I know how to resize. It is about usability...
Please don't get offended. I am talking my time to capture screen. I wish I could upload and share them immediately.
Doing Post processing kills my mood....
Quote from: baor on April 11, 2011, 07:01:38 PM
Quote from: obi-support2 on April 11, 2011, 06:50:17 PM
If call is ended by the server, 2 things should happen:
1. Your call will disappear from the call status page (if you refresh the page)
2. If you don't hang up, you will hear some fast busy tone after 5s or so.
Did the last 2 happen?
Silence on both ends may not be a good indication the server has ended the call.
Or if the call still shows on the call status page (the one you attached),
what does it show for the number of packets sent and received?
If the call is still up, try this: hook flash on your phone, then hook flash again.
Does the audio resume?
Do you have a HK number? I could call you.
Hey, you reminded me something interesting.
I was talking to someone. At around 10 mins time frame, she said she could not hear from me. So she decided to flash and call me again. Her call hooked right back to my end while I was still talking to the phone (I didn't know it was dead on her end).... Does it help?
You are right.....
The call is still on. Packets are sending. But the phone is dead at exactly 10 mins...
baor,
To answer your question regarding why bridging HKBN and GV has no audio...
From you previous HKBN call status, you were using G729 with HKBN.
However GV can only do G711U. To make this work, make sure your call to
HKBN is using G711U (you have to force it by disabling G729 and G726 codecs in the
codec profile). If HKBN indeed does not support G7111U codec at all, then there
is not way to bridge it with GV at the moment.
About the call has no audio on both ends (in your words "dead" or "cutoff") after 10 min, I cannot figure it out yet. All I can see is the OBi still sending out packets (tx packets increasing), but
no packets are coming from the server (rx packets not increasing). You can try hook flash, and hook flash again to see if the incoming packets will resume.
Again, a complete debug log will help with SIP Debug Options enabled will help us diagnose the problem further. If there is a size limit in posting, please email to support@obihai.com.
Quote from: obi-support2 on April 12, 2011, 09:28:59 AM
baor,
To answer your question regarding why bridging HKBN and GV has no audio...
From you previous HKBN call status, you were using G729 with HKBN.
However GV can only do G711U. To make this work, make sure your call to
HKBN is using G711U (you have to force it by disabling G729 and G726 codecs in the
codec profile). If HKBN indeed does not support G7111U codec at all, then there
is not way to bridge it with GV at the moment.
About the call has no audio on both ends (in your words "dead" or "cutoff") after 10 min, I cannot figure it out yet. All I can see is the OBi still sending out packets (tx packets increasing), but
no packets are coming from the server (rx packets not increasing). You can try hook flash, and hook flash again to see if the incoming packets will resume.
Again, a complete debug log will help with SIP Debug Options enabled will help us diagnose the problem further. If there is a size limit in posting, please email to support@obihai.com.
Not all call can be done at G711U. I dunno why. Some calls were using G711U. Most of them were 711A or 729. If disabling 711A and 729, "503 error" would result.
Can this be fixed in future firmware result, ie bridging to CODEC other than 711U?
baor,
Fundamentally, the OBi110 cannot solve this problem completely b/c transcoding btw different codecs may be prohibitedly complex in general. And IMO, it's also unlikely GV will support g729 in the near future.
I have put in a request to add support for G711A for GV; and hopefully that will alleviate this situation.
Quote from: obi-support2 on April 14, 2011, 01:58:23 PM
baor,
Fundamentally, the OBi110 cannot solve this problem completely b/c transcoding btw different codecs may be prohibitedly complex in general. And IMO, it's also unlikely GV will support g729 in the near future.
I have put in a request to add support for G711A for GV; and hopefully that will alleviate this situation.
That means your bridging function is unpredictable......
There are 2 ends here.
1. Calling to Obi1x0
2. Obi1x0 bridging to Receiving end
Is route 1 fixed in what codec is used?
Quite the contrary. The unit will try to match both call-legs wherever possible.
You just picked a case that is not possible to match at the moment.
Quote from: obi-support2 on April 14, 2011, 08:56:19 PM
Quite the contrary. The unit will try to match both call-legs wherever possible.
You just picked a case that is not possible to match at the moment.
Good to hear it is still possible.
Thanks!
Can I request a function?
Can the call history retain those call status information, e.g. codec, packet tx/rx, packet loss, etc???
I have tried many times and still failed to do so.
I do not know wheather other forum user is able to register the HKBN 2b server without any problem encountered.
YH
Quote from: yhfung on April 15, 2011, 08:00:00 AM
I have tried many times and still failed to do so.
I do not know wheather other forum user is able to register the HKBN 2b server without any problem encountered.
YH
Did you follow my settings?
I have tried again on my OBi110 in SZ and am not able make OBI110 register the HKBN 2b server. However when I switched to my PAP2T, the registration works very well.
Also I have tried another OBi110 in Hong Kong, the result is the same and still not able to make HK OBi110 register.
YH
Quote from: yhfung on April 22, 2011, 04:11:28 AM
I have tried again on my OBi110 in SZ and am not able make OBI110 register the HKBN 2b server. However when I switched to my PAP2T, the registration works very well.
Also I have tried another OBi110 in Hong Kong, the result is the same and still not able to make HK OBi110 register.
YH
Did you reset at http://pa.2b.com.hk/login.jsp ???
Did you try different servers, s21.hkbntel.net/s22.hkbntel.net ??
I have followed your suggestions but still failed to make OBi110 register the HKBN 2b server.
I do not know whether other OBi110 users are able to get the registration. No matter the answer is yes or no, just let me know, thanks.
Regards,
YH
If you're trying to use HKBN with the OBi, you may be interested to know that the latest f/w 1.2.1(2286) has some enhancement that should make it work better with HKBN.
On OBi web configuration page, set the new parameter X_ProxyRequire (under ITSP Profile A/B - SIP) to this value: com.nortelnetworks.firewall
You should also disable STUN and ICE when setting up this way.
NOTE: This parameter is not available in the OBITALK Expert Configuration yet. You must use the OBi's own web configuration page to make this change.
Quote from: baor on April 11, 2011, 05:57:42 PM
We finally made some progress after OBIHAI Support suggested to turn off X_InsertRemotePartyID.
I could receive call and dial out. However I got cut off every 10 mins.
Does it have to do with X_KeepAliveExpires or RegistrationPeriod?
baor,
We have some idea why the call might be dropped by the server after 10 min or so.
If you still experience this call drop problem after 10 min, we can patch the
f/w on your unit so you can try out a fix we put in to deal with this issue.
Just send us an email at support@obihai.com w/ your unit's OBI number.
Thank you.
The full installation method for HKBN 2b on OBi device can be found in the following link:
http://www.obitalk.com/forum/index.php?topic=886.0
YH
Quote from: obi-support2 on May 14, 2011, 08:54:58 PM
If you're trying to use HKBN with the OBi, you may be interested to know that the latest f/w 1.2.1(2286) has some enhancement that should make it work better with HKBN.
On OBi web configuration page, set the new parameter X_ProxyRequire (under ITSP Profile A/B - SIP) to this value: com.nortelnetworks.firewall
You should also disable STUN and ICE when setting up this way.
NOTE: This parameter is not available in the OBITALK Expert Configuration yet. You must use the OBi's own web configuration page to make this change.
I have updated the firmware, it broke my service...
RX is not receiving any packet.
TX kept sending...
Update:
The web portal does not work with HKBN but it works when I configured it using the local login.
I wonder if someone can help, I've been using the Obitalk 110 with HKBN 2b fine for a long time. It was still working fine last month or so. However it seems like something happened and it stopped working. I just noticed this when mom told me she hasn't been able to make calls. I am getting this in the line status:
Register Failed: No Response From Server (server=203.80.89.135:5060; retry in 195s)
I have always been using s21.hkbntel.net. I have done the reset on pa.2b.com.hk as usual with no avail. I tried switching to s22, s23, but it just says "moved permanently" etc., which means they aren't the right servers for me.
Just to make sure my account is working, I downloaded the softphone and it logged in without issue. I was also able to make calls.
Is anyone else having issue with 2b and obi? Please help. Thanks!