Problem with connecting IPphone to OBi110

[Siteweb65]

hi All

Hopefully someone here can dig me out of the mess Im in here.

I have a Gigaset DX800a system that works in the small office/house setup I have here, I have an ISP provider that gives me a standard PSTN line with ADSL and an ATA incorporated into the ADSL modem, (ie a standard phone plugged into the modem gives me a second line over IP).

My DX800 only has facility to take one PSTN line and 6 IP ports, so I started my search for a way to convert one of the IP ports to a PSTN line (the one offered from the modem)

with me so far ?

So I found the Obi110, which seems to be capable of doing what I require, even found a couple of threads of people telling me exactly how to set up an IP phone with the Obi to get the functionallity I require.

However the DX800 has no way I can see of enabling a call on non-registration of the server, which would seem to be required as the Obi cant register itself with the Phone.

So Im stuck, I ve sent an email to gigaset, but dont really expect much in the way of return from them, so I'm turning the the experts here for ideas, all I want is to be able to connect an IP line from the DX800a to the phone input I have on my ADSL modem.

Any ideas ?

Arhh yes forgot to say ... I cant simply use the SIP server of the ISP as they dont release the details, you have to use their build in ATA in the modem.

TIA

[azrobert]

The IP phone needs to call out without registration to work directly to the OBi110.
If the phone doesn't have a No Registration option, maybe leaving the Registration Server blank will setup the phone for no registration.

Do you already own an OBi110?
The OBi200 with an OBiLine adapter will work.
You can register a device to an OBi200.

You can use another service between the phone and the OBi110 like a PBX.

http://www.obitalk.com/forum/index.php?topic=5411.0

http://www.obitalk.com/forum/index.php?topic=7606.0

Edit:
If you already own an OBi110 you can place an OBi200 between the phone and the OBi110. Register the phone to the OBi200 and then the OBi200 talks to the OBi110. Obviously, this is a more complex setup.

[SteveInWA]

I have a somewhat similar Gigaset C610A IP phone system.  The Gigasets do NOT support non-registered use.  I realize it doesn't help with your particular VoIP ITSP, but I simply use my Gigaset system independently of the OBi, by registering multiple extensions with my ITSP, Callcentric.

[Siteweb65]

@azrobert

Thanks for your reply, yes its slightly more complicated but its all local, which I like rather than having to use an external service and redirecting back to the Obi, which was my original plan.

Before I place an order for an Obi200, there isnt any other device that will do what I want all in one box is there, to confirm the Phone will not work without registration, i'd already tried leaving fields blank etc.

Thanks for your reply.

@steveInWA

Thanks foryour confirmation, Im awaiting a response from gigaset, but never really expected anything more than a standard reply from them. Its a pity as as with just a bit more effort the phone system could have been made more cusomizable and really been up there with the Ciscos etc.
Yes I use a few of the IP lines to connect to other SIP providers that offer me great rates in certain areas, but the reason Id like to use the ISP Voip is that it provides me free calls to mobiles within my country (something that no one else provides)

Thanks again

[azrobert]

Before I place an order for an Obi200, there isnt any other device that will do what I want all in one box is there

The OBi200 with the USB OBiLine adapter is the only way I know.

http://www.amazon.com/gp/product/B00BUV7C9A/ref=s9_simh_gw_p229_d5_i2?pf_rd_m=ATVPDKIKX0DER&pf_rd_s=center-3&pf_rd_r=079R1201C196MQYN1NBB&pf_rd_t=101&pf_rd_p=1688200422&pf_rd_i=507846

http://www.amazon.com/OBiLINE-FXO-Phone-Line-Adapter/dp/B00DP4YOJ6/ref=sr_1_1?s=office-products&ie=UTF8&qid=1413208411&sr=1-1&keywords=obiline

Disclaimer:
I recently learned about the registration feature on the OBi2xx
See: http://www.obitalk.com/forum/index.php?topic=8718.msg57547#msg57547

I could NOT get the 3CX softphone to register.

The Phonerlite softphone registered.

I did help someone with a Cisco IP phone register to an OBi200.

I registered my OBi1032 IP phone to an OBi200.

I don't know if you will have problems registering your IP phone.

Edit:

I helped someone with an OBi1xx and an IP phone that also required registration. He tried calling out after registration failed and it WORKED. I don't remember the make of the IP phone.

[Siteweb65]

OK, guess im going to have to order one up and see :-)

I cant use the ObiLine, well more precisely I cant 'order' the ObiLine as they wont deliver one here to France, so it will be the         Gigaset DX800 > Obi200 > Obi110 > PSTN    that i'll be trying to set up.

No doubt i'll be back for help with the setup once Ive the parts available ;-)

Thanks

[azrobert]

Below is a sample configuration for your devices.
The OBi110 defaults are for a N. American landline.
I suggest first testing with a phone attached to the OBi110 phone port.
When you take the phone off-hook you get dial tone from the OBi110.
Dial # to get dial tone from the PSTN line.
Call someone.
Hang up.
Does the OBi110 disconnect?
If you have problems I'm NOT the one to help.


OBi200 (IP address=192.168.1.100)

Service Provider -> ITSP Profile A SIP
ProxyServer: 127.0.0.1

Voice Services -> SP1 Service
X_InboundCallRoute: {OBi200>((#|XX.)<:@192.168.1.110:5060>):sp1}
X_Proxy: Checked    (This allows registration)
X_RegisterEnable: Unchecked
AuthUserName: OBi200
AuthPassword: password1

192.168.1.110 is the IP address of the OBi110
5060 is the X_UserAgentPort of OBi110 SP1 trunk                    
------------------------------------------------------------------------

OBi110 (IP address=192.168.1.110)

Service Provider -> ITSP Profile A SIP
ProxyServer: 127.0.0.1

Voice Services -> SP1 Service
X_InboundCallRoute: {OBi200>(<#:>|XX.):li1)
X_RegisterEnable: Unchecked
AuthUserName: OBi110

------------------------------------------------------------------------

IP Phone

Proxy: 192.168.1.100  (IP address of the OBi200)
Proxy Port: 5060     (X_UserAgentPort of OBi200 SP1 trunk)
UserID: OBi200
Password: pasword1

Dial any combination of digits and *
The dialed number will be routed out the Line port on the OBi110.
Dial # to get dial tone from the PSTN line.
The config can be altered to restrict what can be dialed.

[hwittenb]

Acrobert,

Good work.  I tested your configuration and it works for me.  I was testing with an OBi202 and the XLite softphone for a sip client.

I did notice that the OBi's do not pass rfc2833 dtmf.  I had to set the sip client to use INBAND dtmf.

I haven't seen any comment on this, but it looks to me like this configuration, where you are registering a client to the OBi202, as opposed to unregistered direct ip calling, will only work when you are dealing with everything on the local network.  I haven't been able to get a sip client to register to my OBi202 over the internet.  In other words it probably wouldn't work with an IPhone or Android smartphone remote from the local network.

INCOMING PSTN LINE CALLS

For calls going the other way, i.e. incoming PSTN Line call that you want to go to the IP Phone, it looks like that could be a challenge if you have to go over the registered path.  It is more of a challenge because the OBi110 is a separate ata.  I was using pg 118 of the OBihai Administration Guide for an idea of how to do it.  

If it works, you could setup the OBi110 Line Port X_InboundCallRoute to send the call directly to the local network ip address of the IP Phone.  That should be straight forward assuming the IP Phone has a static ip address on the local network.  Depending on the IP Phone client it may not work.  

If the IP Phone requires that the incoming call come from the OBi200 then you may have to use a spare SPx on the OBi200.  After a struggle, I got the following to work where the IP Phone is registered to SP1 and I am using SP3 to route the call to the IP Phone:

OBi200
SP3
X_UserAgentPort: 5023
X_IncomingCallRoute: {sp1(OBi200@local_client)}

Where the SPx defined to allow the IP Phone to register is on SP1
Where 5023 is the X_UserPort port number setup on OBi200 SP3 and "local_client" is just that.  5023 is just an arbitrary number used for a port number.

OBi110
LinePort
X_IncomingCallRoute: SP1(OBi200@192.168.1.117:5023)

Where 192.168.1.117 is (my) local network address of the OBi200 and 5023 is the X_User port number of SP3 on the OBi200 that has the Incoming Call Route is setup to call the IP Phone

Where SP1 is a Sip defined SPx on the OBi110
Where OBi200 is the UserID setup on the SP1 on the OBi200

Maybe someone can make and test an improvement.

[azrobert]

I got CSipSimple on an Android to register to my OBi200 using the WiFi hotspot at my health club. I believe an outbound call connected, but I had a slight problem of no audio. LOL.  I didn't try to fix the problem.

ianobi got CSipSimple to work without registration.
See: http://www.obitalk.com/forum/index.php?topic=8511.0
If I have to do the same to get the above to work, why bother with registration.

I just used a speed dial on my OBi110 to call the Phonerlite softphone using the IP address of the computer running Phonerlite. Phonerlite was registered to my OBi200, so I think routing the inbound PSTN call directly to the IP address of the smartphone/IP_Phone should work.

[Siteweb65]

@azrobert

Thanks for your detailed instructions, I recieved the OBi200 yesterday and have just connected up and programmed as per you instructions.

The DX800a registers OK .... phew :-)

However when I go to dial out on on the IP1 service (that ive set up) I just get engaged tone and line busy :-(

Looking at the SP service stats on the OBi200, shows no packets sent or recieved on any of the SP services ...  

any ideas ?

[azrobert]

Temporarily try this in the OBi200
X_InboundCallRoute: ph

Make a call.
The OBi200 phone port should ring.
You don't need a phone attached, the phone port LED will flash.

The inbound route is comparing for the Cicso's UserID.
Sometimes the IP Phone transmits something other than the UserID, like the IP address.

If the phone rings look at the OBi200's call history.

Log directly into the OBi using the local interface.
Key the IP address of the OBi into a Web Browser.
Hit Enter
The UserID and default Password are both "admin".
Click Status on the left column.
Then click Call History.

The Peer Number is the ID the Cisco is sending.
Change the inbound route to:
{192.168.1.120>((#|XX.)<:@192.168.1.110:5060>):sp1}

   

[Siteweb65]

Sorry Azrobert

I relised my mistake after I posted, and immeadiately removed the post ... but you were too quick ;-)

OK, Ive got a connected call

OBi200 log shows

Call 1 10/19/2014    00:26:19

00:26:19 From 'obi200' SP1(obi200) To SP1(3103@192.168.1.101:5060) 
00:26:21  Call Connected
00:26:53 Call Ended


OBi110 log shows

Call 1 10/19/2014    00:26:18

Terminal ID SP1 LINE1
Peer Name obi200 
Peer Number obi200 3103
Direction Inbound Outbound
00:26:18 Ringing 
00:26:20  Call Connected
00:26:53 End Call


but so far no audio ... 3103 is my message service
192.168.1.100  is my phone
192.168.1.101  is OBi110
192.168.1.102  is OBi200

Off to bed now its late, but will try again in morning with a clearer head ;-)

[azrobert]

I assigned different RTP ports to the OBi's.
I used the defaults 16600 thru 16798 for the OBi200 and 17600 thru 17698 for the OBi110.
Then used port forwarding in my router.

The RTP ports are found here:
Service Providers -> ITSP Profile A RTP -> LocalPortMin/Max

I also assigned different listening ports.
This is a good idea for security.

Voice Services -> SP1 Service -> X_UserAgentPort
Pick a number like 7560 and 7660.

I used a softphone to test and didn't port forward its RTP ports.
If the above doesn't work you can try the same for the Cisco.   

[hwittenb]

but so far no audio ... 3103 is my message service
192.168.1.100  is my phone
192.168.1.101  is OBi110
192.168.1.102  is OBi200

Audio problems are frequently caused inside the call signalling by either or both OBi's including an external ip address when it should be an internal ip address or vice versa.

The call path between the OBi110 and the OBi202 is over your local network so the signalling should be using the local network ip addresses.

I would try setting the following in the ITSP Profile-->SIP for both the OBi110 and the OBi220
Service Providers -> ITSP Profile A-->X_DiscoverPublicAddress (uncheck)
The default setting is for this setting to be checked.

The recommended port number changes are good configuration practice.

[azrobert]

Service Providers -> ITSP Profile A-->X_DiscoverPublicAddress (uncheck)

I found if the trunks used on both OBi's are not registered or registered to a local service like Asterisk this change is not needed.

Both OBi's in this setup have their trunks unregistered.

[Siteweb65]

Hi Guys

Sorry for the long delay, life took over for a while ;-)

OK, Ive tried all the above fixes, but still have no audio :-(

The phone connects and dials the number OK, I can listen in on a seperate phone plugged into the PSTN line after dialing and hear my message service, but still no audio on the DX800.

Is there any way of drilling down deeper to see if voice packets are being passed between the two Obi devices ?

[hwittenb]

Hi Guys

Sorry for the long delay, life took over for a while ;-)

OK, Ive tried all the above fixes, but still have no audio :-(

The phone connects and dials the number OK, I can listen in on a seperate phone plugged into the PSTN line after dialing and hear my message service, but still no audio on the DX800.

Is there any way of drilling down deeper to see if voice packets are being passed between the two Obi devices ?

I would run a syslog trace on the call to see the sip signalling to see if that gives a clue to the problem.

As I said audio problems are usually caused by one side or the other sending a local ip address instead of an external ip address or vice versa.  It can also be caused by incomplete call signalling where one side doesn't receive an expected acknowledgement.  A further problem can be caused when routers are changing port numbers.  A sip trace is used to try to pin down the cause of the problem.

To run a syslog trace you download and install a syslog program on your computer and make some configuration settings.  Obihai has instructions here and a simple pc syslog program for download:
http://www.obihai.com/faq/Troubleshooting-sec/collect-syslog-from-OBi

A more sophisticated trace can be obtained with the WireShark packet trace program but that requires that the computer running the program to be able to see all the packets going and coming from the OBi.

[Siteweb65]

Hi hwittenb 

Thanks for your reply, I already have Wireshark so I'll see what I can do with that, my question was more a wonder if the Obi's had internal logging of packets that I could view.

My setup here is 100% internal on the LAN, no external IP's are referenced within this set up (though I have other SIP services running that use external SIP providers), thats why I find it strange, I'd have thought that packets would have naturally been routed to other internal devices.

i'll come back with any findings, thanks again

[hwittenb]

If WireShark can see all the packets it has a feature where after a capture you can click on VoIP calls and it will highlight the calls and then the signalling for a particular call.  It is really a useful tool.  If rtp packets are being sent they are also easy to see because they dominate a capture.

In my opinion Obihai never intended for people to use direct ip addressing for calling.  I don't believe it is part of their software regression testing, the admin manuals don't discuss it.

[Siteweb65]

OK, wireshark coulnt see all packets so Im using the suggested logger

Log from OBi101

[Oct 31 16:19:14][192.168.1.101]<7> FXO:make new call
[Oct 31 16:19:14][192.168.1.101]<7> [DAA]: FXO OFFHOOK
[Oct 31 16:19:14][192.168.1.101]<7> FXO:NewTermState:offhook
[Oct 31 16:19:15][192.168.1.101]<7> FXO:Start Tone 3
[Oct 31 16:19:15][192.168.1.101]<7> [CPT] --- FXO s/w tone generator (3) ---
[Oct 31 16:19:15][192.168.1.101]<7> FXO:Stop Tone
[Oct 31 16:19:15][192.168.1.101]<7> FXO:Start Tone 1
[Oct 31 16:19:15][192.168.1.101]<7> [CPT] --- FXO s/w tone generator (1) ---
[Oct 31 16:19:15][192.168.1.101]<7> FXO:Stop Tone
[Oct 31 16:19:16][192.168.1.101]<7> FXO:Start Tone 0
[Oct 31 16:19:16][192.168.1.101]<7> [CPT] --- FXO s/w tone generator (0) ---
[Oct 31 16:19:16][192.168.1.101]<7> FXO:Stop Tone
[Oct 31 16:19:16][192.168.1.101]<7> FXO:Start Tone 3
[Oct 31 16:19:16][192.168.1.101]<7> [CPT] --- FXO s/w tone generator (3) ---
[Oct 31 16:19:16][192.168.1.101]<7> FXO:Stop Tone
[Oct 31 16:19:16][192.168.1.101]<7> FXO:ConnectRightAway
[Oct 31 16:19:16][192.168.1.101]<7> FXO:OutboundConnected
[Oct 31 16:19:16][192.168.1.101]<7> RTP:DtmfTxMtd:1(1),0
[Oct 31 16:19:16][192.168.1.101]<7> RTP:Start->c0a80166:16606(81);0;0;0:0:0;0(26

[Oct 31 16:19:54][192.168.1.101]<7> [DAA]: FXO ONHOOK MONITOR
[Oct 31 16:19:54][192.168.1.101]<7> FXO:NewTermState:onhook
[Oct 31 16:19:54][192.168.1.101]<7> RTP:Del Channel
[Oct 31 16:19:54][192.168.1.101]<7> [JB] call overall status --
peer:             192.168.1.102:16606,
local:            192.168.1.101:17610,
pkt_tx:           1874,
pkt_rx:           1867,
bytes_tx:         322328,
bytes_rx:         321124,
clk_diff:         -48 PPM,
pkt_in_jb:        3,
pkt_ooo:          0,
pkt_lost:         0,
pkt_late:         0,
pkt_loss_rate:    0 %,
pkt_drop_rate:    0 %,
jb_len:           180 ms,
curr_rcvd_jitter: 4 ms,
rcvd_digits:      0,
underruns:        0,
overruns:         0,
seq_num_broken:   0,
pkt_interp:       0,
skew_comp:        0 ms,
frm_in_pkt:       2
[Oct 31 16:20:26][192.168.1.101]<7> [DAA]: FXO ring on
[Oct 31 16:20:26][192.168.1.101]<7>

• Ring On
  [Oct 31 16:20:26][192.168.1.101]<7> FXO:NewTermState:ringing
  [Oct 31 16:20:26][192.168.1.101]<7> ------ caller id (pcm_id: 1) received! -----
  ------
  [Oct 31 16:20:26][192.168.1.101]<7>
• DAA CND ,,,,,
  [Oct 31 16:20:26][192.168.1.101]<7>
• DAA CND ,,,,,
  [Oct 31 16:20:30][192.168.1.101]<7> fxo: cp proceeding
  [Oct 31 16:20:31][192.168.1.101]<7> [DAA]: FXO ring off
  [Oct 31 16:20:31][192.168.1.101]<7>
• Ring Off
  [Oct 31 16:20:31][192.168.1.101]<7> FXO:NewTermState:onhook
  [Oct 31 16:20:31][192.168.1.101]<7> FXO tell cc end-call
  [Oct 31 16:20:31][192.168.1.101]<7> [DAA]: FXO ONHOOK MONITOR
  
  
  Log from OBi200
  
  [Oct 31 16:19:14][192.168.1.102]<7> CCTL:NewCallOn Term 10[0] obi200->3103@192.1
  68.1.101:5060,3103@192.168.1.101:5060
  [Oct 31 16:19:16][192.168.1.102]<7> RTP:DtmfTxMtd:1(1),0
  [Oct 31 16:19:16][192.168.1.102]<7> RTP:DtmfTxMtd:1(1),0
  [Oct 31 16:19:16][192.168.1.102]<7> RTP:Start->c0a80165:17610(80 0);0;0;0:0:0;0(40)
  [Oct 31 16:19:16][192.168.1.102]<7> RTP:Start->c0a80164:5016(80 0);0;0;0:0:0;0(38)
  [Oct 31 16:19:54][192.168.1.102]<7> RTP:Del Channel
  [Oct 31 16:19:54][192.168.1.102]<7> RTP:Del Channel
  
  
  Does that make any sense to you ?