Hi, First Thanks in advance for reply to this forum.
I have a OBI 202 and I want to set up Voxbeam for SP3.
I use following link to configure voxbeam.
http://www.obitalk.com/forum/index.php?topic=2898.0
Only difference I see is :
Physical Interfaces -> PHONE Port -> DigitMap:
([1-9]x?*(Mpli)|[1-9]S9|[1-9][0-9]S9|911|**0|***|#|**1(Msp1)|**2(Msp2)|**3(Mvg3)|
**8(Mli)|**9(Mpp)|(Mpli))
=======================
On 202 is as following:
([1-9]x?*(Mpli)|[1-9]S9|[1-9][0-9]S9|911|**0|***|#|##|**70(Mli)|**8(Mbt)|**81(Mbt)|**82(Mbt2)|**1(Msp1)|**2(Msp2)|**3(Mvg3)|**4(Msp4)|**9(Mpp)|(Mpli))
=======================
Physical Interfaces -> PHONE Port -> OutboundCallRoute:
{([1-9]x?*(Mpli)):pp},{(<#:>|911):li},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},
{(<**3:>(Mvg3)):vg3},{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}
==============================
On my OBI 202 is
{([1-9]x?*(Mpli)):pp},{(<##:>):li},{(<#:>):ph2},{(<**70:>(Mli)):li},{(<**82:>(Mbt2)):bt2},{(<**81:>(Mbt)):bt},{(<**8:>(Mbt)):bt},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{(<**3:>(Mvg3)):vg3},{(<**4:>(Msp4)):sp4},{(<**9:>(Mpp)):pp},{(Mpli):pli}
===============================
It was not in instrucation but I modified CallReturnDigitalMap as following:
{pli:(xx.)},{sp1:(<**1>xx.)},{sp2:(<**2>xx.)},{vg3:(<**3>xx.)},{sp4:(<**4>xx.)},{bt:(<**8>xx.)},{bt2:(<**82>xx.)},{pp:(<**9>xx.)}
My Problem is that I am trying to call India and I dialed **3 91 XXX XXX XXXX but i get error message that "There is no call route available to complete your call".
What am I doing wrong? Can someone help with this?
Thanks,
Raj
Raj - welcome to the forum.
Your OBi202 details look correct. The original post looks like it was configured for an OBi110.
The example post you quoted seems to have a mistake in it – a rare occurrence for RonR! The AccessNumber for a Voice Gateway needs reference to an spX that is configured for SIP (not Google Voice). The example quotes sp2, so it should read:
Voice Services -> Gateways and Trunk Groups -> Voice Gateway3
Name : Voxbeam
AccessNumber :
sp2(sbc.voxbeam.com)DigitMap : (<ppppppp>(<1aaa>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|1xxxxxxxxxx|91xxxxxxxxxx|xx.S4))
AuthUserID : (CallerID)
where ppppppp is 0011103 for standard routes and 0011101 for premium routes, and aaa is your local area code.
I have made some changes to the DigitMap, which should be more responsive to numbers starting with 91 followed by ten digits.
You can use any working spX configured for SIP (sp2 in this example) or set up a fake spX as shown in the post you quoted.
Hi, Thanks for your reply.
I set it up SP3 and I check on log messages but I think problem now is that it duplicate voxbeam digit string.
Here is the log.
<7> CCTL:Filter=13 2 _g3sp3*0011101001110191xxxxxxxxxx
<7> CCTL:NewCallOn Term 12[0] ->0011101001110191xxxxxxxxxx,_g3sp3*0011101001110191xxxxxxxxxx
<7> [SLIC] Command: 0, 1, 0, 0, 0, 0,
Somehow I am expecting only one 0011101 string but I do not know where is it duplicating it? Can you please help me to find out where is it duplicating?
Thanks for your help.
Raj
Looks like we are adding it in the Phone Port DigitMap and then again in the Phone Port OutboundCallRoute. Try this:
Voice Services -> Gateways and Trunk Groups -> Voice Gateway3
Name : Voxbeam
AccessNumber : sp2(sbc.voxbeam.com)
DigitMap : (ppppppp(<1aaa>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|1xxxxxxxxxx|91xxxxxxxxxx|xx.S4)|<ppppppp>(<1aaa>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|1xxxxxxxxxx|91xxxxxxxxxx|xx.S4))
AuthUserID : (CallerID)
If you only use Voxbeam for international, then you could simplify it to:
Voice Services -> Gateways and Trunk Groups -> Voice Gateway3
Name : Voxbeam
AccessNumber : sp2(sbc.voxbeam.com)
DigitMap : (ppppppp(91xxxxxxxxxx|xx.S4)|<ppppppp>(91xxxxxxxxxx|xx.S4))
AuthUserID : (CallerID)
Thanks for your reply.
It works and I got following log messages but still I am not able to contact voxbeam.
Here is the Information, I worked on it.
====================================
Voice Services -> Gateways and Trunk Groups -> Voice Gateway3
Name : Voxbeam
AccessNumber : sp3(sbc.voxbeam.com)
DigitMap : (ppppppp(91xxxxxxxxxx|xx.S4)|<ppppppp>(91xxxxxxxxxx|xx.S4))
AuthUserID : (CallerID)
====================
This gives me following results.
<7> CCTL:Filter=13 2 _g3sbc.voxbeam.com*001110191xxxxxxxxxx
<7> CCTL:NewCallOn Term 10[2] ->0011101xxxxxxxxxx,_g3sbc.voxbeam.com*001110191xxxxxxxxxx
String looks good to me but I hear "The number you dial was rejected by service provider and reason is 500".
What is the problem?
Thanks,
Raj
Your OBi is sending the correct digits, so it looks like the problem is with Voxbeam. Have you set up your public IP address with Voxbeam - see your original example post:
http://www.obitalk.com/forum/index.php?topic=2898.msg19186#msg19186
The public IP address may also need the port number. For sp3 the default port number is 5062.
I don't use Voxbeam, so I can't tell you exactly what they require. Your OBi looks good now, so you might need to contact Voxbeam directly.
To help others, please let us know how you get on.
Thanks for your reply. I correctly set up my public IP in voxbeam. They do not have port set up in voxbeam account. I opened a case with them and let's see what are they coming up with it? I will update complete information once it will work correctly.
Thanks,
Raj
I called Voxbeam technical support with log messages and they reply me back with following message.
=====================================
Thank you for contacting Technical Support. The reason for the call failure is because your setting the caller-ID information as "sbc.voxbeam.com" which is not acceptable. All caller-ID cannot be in the form of a .com string. You will need to ensure your using a valid e.164 formatted CLI when your sending calls to us.
====================================
I corrected following information:
====================================
Voice Services -> Gateways and Trunk Groups -> Voice Gateway3
Name : Voxbeam
AccessNumber : sp3(+1XXXXXXXXXX)
DigitMap : (ppppppp(91xxxxxxxxxx|xx.S4)|<ppppppp>(91xxxxxxxxxx|xx.S4))
AuthUserID : (CallerID)
I corrected as above and got following log.
<7> [SLIC] Command: 1, 10, 3, 0, 0, 0,
<7> [SLIC] Command: 1, 10, 4, 0, 0, 0,
<7> CCTL:Filter=13 2 _g3+1xxxxxxxxxx*001110191xxxxxxxxxx
<7> CCTL:NewCallOn Term 10[2] ->001110191xxxxxxxxxx,_g3+1xxxxxxxxxx*001110191xxxxxxxxxx
<7> [SLIC] Command: 0, 1, 0, 0, 0, 0,
<7> [SLIC] Command: 0, 4, 5, 0, 0, 0,
Now I am getting error message that "number is rejected by service provider with error code 404"
Can you please let me know, now what is wrong with this request?
Thanks,
Raj
They seen to be talking about CallerID so I think this should work:
Voice Services -> Gateways and Trunk Groups -> Voice Gateway3
Name : Voxbeam
AccessNumber : sp3(sbc.voxbeam.com)
DigitMap : (ppppppp(91xxxxxxxxxx|xx.S4)|<ppppppp>(91xxxxxxxxxx|xx.S4))
AuthUserID : +1xxxxxxxxxx
Edit: Parentheses removed from AuthUserID - they are not required.
Thanks for your reply.
Now I am able to successfully connect it. But problem is voice is not transmitting. I am not able to hear other person and he is not able to hear me also. So is there any setting in OBI that won't transmit voice? Or it is an issue with Voxbeam. I tried both premium as well as direct line. I check in my account that call is connected and they deducted money also. So there must be some issue.
I again appreciate your help. I am going to write complete manual for voxbeam set up, if other person wants use it with OBI.
Thanks,
Raj
It's obvious from this thread and the one that you quoted originally that your manual is much needed!
Looks like we are making some progress. As you may know, a SIP call comes in two parts: Firstly, the SIP signalling which routes the call, sets it up , disconnects it etc. This we have now achieved. Secondly, the media part (our voices) which rely on Real Time Protocol (RTP) after SIP has set the call up. This seems to be our problem now. There are two ways to improve RTP connection:
1. Ensure that the correct ports are forwarded in your router. The Obihai FAQ section is out of date on this issue. As you are using sp3 as a "carrier" for Voice Gateway 3, in your router you need to forward the range of RTP ports defined here:
Service Providers > ITSP Profile C > RTP > LocalPortMin
to
Service Providers > ITSP Profile C > RTP > LocalPortMax
The protocol defined in your router need only be UDP.
Also if your router has a SIP ALG (Application Layer Gateway) function, then turn it off.
2. Use a STUN server. You can use any public STUN server, I use ideasip as it seems reliable. Set up as follows:
Service Providers > ITSP Profile C > General > STUNServer: stun.ideasip.com
Service Providers > ITSP Profile C > General > STUNServerPort: 3478 (default)
No need to check STUNEnable. I know this sounds strange, but it is not needed and may confuse later providers using sp3 or other voice gateways that use sp3.
One more change:
Voice Services -> Gateways and Trunk Groups -> Voice Gateway3
Name : Voxbeam
AccessNumber : sp3(sbc.voxbeam.com
;op=s)
DigitMap : (ppppppp(91xxxxxxxxxx|xx.S4)|<ppppppp>(91xxxxxxxxxx|xx.S4))
AuthUserID : +1xxxxxxxxxx
When the AccessNumber is used op=s tells sp3 (which you are using as a carrier for Voice Gateway 3) to use the STUN server set up on sp3 (ITSP Profile C).
If you do write the manual, then remember that the DigitMap will depend on what numbers the user wants to use Voxbeam to dial. Good luck!
Thanks for your help.
I did open port and set up STUN server also. Still I voice is not able to transmit. Do you know why? Any debug method to find out this issue?
Appreciate your help.
Thanks,
Raj
To prove there is no problem with ports being open, can you put your OBi202 in DMZ in your router? This is just for testing purposes.
While a call is in progress, have a look at Status > Call Status. In the RTP rows you should see something like this:
Peer RTP Address 217.10.77.241:38176
Local RTP Address 192.168.1.10:16026
RTP Transport UDP
Audio Codec tx=G711A; rx=G711A
RTP Packetization (ms) tx=20; rx=20
RTP Packet Count tx=238; rx=240
RTP Byte Count tx=40936; rx=41280
If the call fails, then I suspect some of the "rx= " numbers will be missing.
I'm running out of ideas now ??? I do suggest checking all the settings in our last few posts. It's easy to miss a semicolon or put a space in by mistake.
Next step may be a support ticket to Obihai. I've never known no speech at all after doing all the things we have tried!
Thanks for your reply.
I got following Information in log.
Call 1
Terminal 1 Terminal 2
Terminal ID PHONE1 SP3
State peer-ringing peer-ringing
Peer Name
Peer Number **3001110191xxxxxxxxxx 001110191xxxxxxxxxx@sbc.voxbeam.com;op=s
Start Time 14:27:53 14:27:53
Duration 00:00:38 00:00:38
Direction Outbound Outbound
Peer RTP Address 10.49.139.61:20144
Local RTP Address 192.168.1.138:17004
RTP Transport UDP
Audio Codec tx=G711U; rx=
RTP Packetization (ms) tx=20; rx=0
RTP Packet Count tx=1778; rx=0
RTP Byte Count tx=305816; rx=0
I tried to put my OBI in DMZ and still I got the same response. Does it mean that I need to set something in RTP? or Do you think it is a router issue? We are almost there...Thanks again for your help.
Raj
The following is from Voxbeam FAQ:
What audio codecs does Voxbeam support?
We support the G.729a and G.723 codecs for all of our routes. Additionally, the G.711u, G.711a, and G.726 codecs are supported for most.
Maybe Voxbeam doesn't support G11U for the route you are using.
You can try disabling all the codecs in the OBi except G729a.
azrobert may be on to something here. It's worth trying different codecs. I would leave the OBi in DMZ while you test.
Voxbeam does seem to be quite a challenge. I was hoping that we would solve it this year, but time has run out in my time zone, so I'm off to celebrate the New Year! We have all of 2014 to finish the task :)
Happy New Year to both of you.
Here is the latest update.
Call 1
Terminal 1 Terminal 2
Terminal ID PHONE1 SP3
State peer-ringing peer-ringing
Peer Name
Peer Number **3001110191xxxxxxxxxxx 001110191xxxxxxxxxxx@sbc.voxbeam.com;op=s
Start Time 13:44:49 13:44:49
Duration 00:00:43 00:00:43
Direction Outbound Outbound
Peer RTP Address 10.49.138.20:23360
Local RTP Address 192.168.1.138:17002
RTP Transport UDP
Audio Codec tx=G729; rx=
RTP Packetization (ms) tx=20; rx=0
RTP Packet Count tx=2116; rx=0
RTP Byte Count tx=67712; rx=0
My OBI is in DMZ and I disable all the codecs and set it up G729 Codec for SP3. You can see this in log. Still no luck. Any Idea?
Thanks for your help.
Raj
Happy New Year to you too!
99% of the time STUN either helps or does no harm. Rarely, I have seen it make things worse. You could try removing the ";op=s" leave the OBi in DMZ and try again.
If that fails, then put ";op=s" back, make a call, then ask Voxbeam support to debug it. Give them the exact time of the call and explain that your OBi is in DMZ.
I can't think of anything more to do at your end.
I have also seen that Voxbeam can be picky with some of the settings
and how your endpoint address is discovered.
A couple of options I have checked off with Voxbeam trunks, and that you may
want to re-review are:
1. ITSP profile -> SIP -> X_PublicIPAddress
enter the public IP address of your home/office connection. If this changes, you'll have
to keep an eye on it and make sure it's correct.
2. As others mentioned I have STUN configured but I also have enabled ICE and symmetric port
under ITSP -> General
3. I saw flakiness when Voxbeam is on a trunk that uses a SIP agent port that is not set to 5060.
This may as well be an issue with my setup/firewall/etc etc. But worth experimenting.
4. Could be an issue with port forwarding configuration with your firewall.
In my setup I have seen that even if I place the Obi in DMZ, it wouldn't make a difference (and not sure why).
hope one of the above helps.
L.
OK, Good. It is working now. I use another router and open port into router in the range from 17000 to 17098.
Just one more problem.
I am trying to call through my OBI app through smartphone and I get following log and it change VoiceCodec. I want to use G729. It become bridge and that's why It is not working.
Can you please let me know, how to configure it so that my softphone use correct codec?
Thanks,
Raj
======================================
Number of Active Calls: 1
Call 1
Terminal 1 Terminal 2
Terminal ID OBiTALK1 SP3
State ringing peer-ringing
Peer Name Raj iPhone
Peer Number 290xxxxxxx; GW=xxxxxxxx 001110191xxxxxxxxxxxx@sbc.voxbeam.com;op=s
Start Time 20:01:36 20:01:37
Duration 00:00:04 00:00:04
Direction Inbound Outbound
Peer RTP Address 0.0.0.0:0 169.132.139.41:20598
Local RTP Address 10.1.10.185:32542 10.1.10.185:17012
RTP Transport UDP UDP
Audio Codec tx=G711U; rx=G711U (bridged) tx=G711U; rx=G711U (bridged)
RTP Packetization (ms) tx=0; rx=0 (bridged) tx=20; rx=0 (bridged)
RTP Packet Count tx=0; rx=0 tx=0; rx=2
RTP Byte Count tx=0; rx=0 tx=0; rx=26
First of all it's not clear what you mean by "it's not working" in the scenario you mention.
The call is established but you can't hear anything? or one direction isn't working?
What happens if you dial into the AA from your obion app and you press the AA options?
does the tone go through?
Now some more comments:
1. I have given up on the Obion app some time ago for the simple reason that it seems
that the Obihai folks seem to have abandoned it and feels totally unsupported. There hasn't been
a release or an enhancement for a long long time. However, from what I recall there wasn't a
direct way to control the codec. Rather, you could specify the "quality" level and that
indirectly would give a hint to the app to use one of the supported codecs. So you have to
experiment and try a few of the settings and then monitor on your obi which codec turns out to be 729.
In case you care to explore more, I'm using the csipSimple VOIP client on my phone to dial
into the AA and then dial out. And obviously there are tons of other open sourced sip clients.
2. I think (but haven't tested) that you could possibly force the use of G729 by disabling all the codecs
except for g729 in the voice services -> obitalk -> codecprofile setting (you need to adjust the
corresponding codec profile to have only g729 enabled).
3. I don't think that "bridging" is your issue. Bridging here means that the two parts of the
call are "connected" at the obi. The way it works is that: you place a call into obi (that is one session)
and then the obi places another call to the remote end you want to dial (this is the 2nd session). When
these two sessions are established, Obi bridges them (which means connects them back to back)
and performs necessary transcoding if needed.
What I think is your problem is that the peer address of your Obion app shows up as 0.0.0.0
That makes me suspect that obi doesn't know how to reach the obion client and this most likely result
in a partially setup call. Now that's sometimes a bit hard to figure out.
In some cases I had seen this happening when the phone running obion
was behind the same home firewall as the obi.
As soon as I was switching to my 3G connection (which means that obion
was coming into obi from the outside and not discovered on the local network, things start working. So do make sure this is not the cause.
L.
My voxbeam set up is working fine but after 15 minutes my call drop from Voxbeam. I called there technical support guy and they told me that after 15 minutes server use reinvite and OBI device is not allowing to reinvite and after several try it drop my call. do you know where can I change configuration so that reinvite acknowledge call and it will continue my call? Thanks for your help.
Raj
that's strange. I didn't have to do anything with re-invite settings. I have them
at whatever defaults and didn't experience the problem you see.
some of the SIP settings are under if you want to check them out.
ITSP Profile X -> SIP
I should point out two things though about my config with Voxbeam:
1. Voxbeam's SBC server doesn't support registration. So unmark the following;
Voice Services -> SPn service -> X_registerEnable
2. Under the same voice services screen, I have enabled keepalives. Not sure if
it has anything to do with me seeing different behavior.
Thanks for your Reply.
I called Voxbeam and they reply me with following answer.
Per our phone discussion please see below the following trace. If you notice in the INVITE dialog there's an internal IP address that you setting in the Via header. When the re-INVITE gets sent back it's being pointed back to your internal IP address instead of your public. You need to either remove the internal IP address and replace it with your public address or completely remove your Via header from your INVITE dialog.
U 2014/03/02 16:06:30.802256 xx.xx.xx.x:5062 -> 108.59.2.133:5060
INVITE sip:0011101917600665927@sbc.voxbeam.com SIP/2.0.
Call-ID: 71456966@10.1.10.185.
Content-Length: 171.
CSeq: 8001 INVITE.
From: "Raj Patel" <sip:+14848759610@sbc.voxbeam.com>;tag=SP37f937057ffff0131.
Max-Forwards: 70.
To: <sip:0011101917600665927@sbc.voxbeam.com>.
Via: SIP/2.0/UDP 10.1.10.185:5062;branch=z9hG4bK-5cf5f497;rport.
User-Agent: OBIHAI/OBi202-3.0.1.4303.
Contact: "Raj Patel" <sip:sbc.voxbeam.com@10.1.10.185:5062>.
Expires: 60.
Supported: replaces.
Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE.
Content-Type: application/sdp.
How can I remove header Information? Anybody has any idea?
Thanks,
Raj