Getting SIP to work on Obi 202

[Hortoristic]

Below my question, is a snippet from RonR on how to get SIP configured.

• On my router, I forwarded ports 5060-5061 to the IP of my OBI202
• My DID is set up as sip:441903300093@myname.dyndns.info:5061
• When I call the above number, my phone rings on Obi202, however when I answer; there is no voice in either direction - this happened before I followed RonR advice to use a SP and set up proxy server as 127.0.0.1 - which was really odd, I would have though the phone wouldn't have even rang until I somewhat configured a SP (SP3 in my case) - so whether I configured something on SP3 or not - the phone rang - just not getting any voice in either direction, but is ringing as expected.
  

I vaguely remember some other router settings I can try - I'm using Dlink DIR645 - however; all my other VOIP stuff works, so not sure that even needs messing with.

RonR:
If you have one or more DID's that forward via SIP URI to an ATA, softphone, or hardphone, they can be forwarded to your OBi instead.

Your OBi must have SP1 and/or SP2 configured for SIP.  It need not be a working provider.  You can set the ITSPx proxy server to anything (like 127.0.0.1) and disable X_RegisterEnable on SPx.

Forward ports 5060 and 5061 in your router to your OBi's IP address.

Forward your DID to your public IP address and the appropriate port number (SP1 = 5060. SP2 = 5061).  If you don't have a static IP address, a Dynamic DNS host name works fine.  Use any phone number:

12345@xxx.dyndns.org:5061

Calls will arrive at the PHONE Port, but CallerID can used on the SPx -> X_InboundCallRoute to perform the normal routing tricks.

I'm using this with IPComms and IPKall and both work very reliably.

[QBZappy]

On the SPx account, you may need to forward these ports as well.

ITSP Profile X
    General
    SIP
    RTP <------------ Port Forward this range. Note that these should be different for every SPx account.

[Hortoristic]

The Min and Max ports here for ITSP Profile C RTP are 17000 and 17098 - those ports come back as invalid in my port forwarding.

Port forwarding as: 5060-5062 works but 5060-5062, 17000-17098 won't save on router.

So my phone still rings, just can't hear voice on either side.

Vaguely remember I should have some of these router settings unchecked?
PPTP
IPSEC
RTSP
SIP

Any unchecked?  They are all checked at moment

[QBZappy]

Port forwarding as: 5060-5062 works but 5060-5062, 17000-17098 won't save on router.

SIP

These could be set up as separate rules
UDP 5060-5062
UDP 17000-17098
SIP setting on the router might be your sip alg which might need to be disabled. Uncheck it. It might be enough to fix everything.

[Hortoristic]

I disabled the SIP in Dlink router and ALL IS WORKING GREAT!  Thanks!

What's weird is my router help text says having SIP checked is supposed to help and improve VOIP services, not hurt it.

Something else to note; I have GV on SP1 and voip.ms on SP2 - and nothing configured on SP3 or SP4 - yet when I call my number as below, it really works and rings and I can talk both ways: sip:441903300093@myname.dyndns.info:5061 - SP2 is actually ringing - seems to be controlled by the fact I specified 5061 - is that true?

[ianobi]

You are correct. 5061 is the UserAgentPort for sp2 so the call is directed to sp2.

It might be interesting for you to direct the call to a spare sp maybe sp3 or sp4, after setting it up for sip as sugested by RonR. I presume default UserAgentPorts for an OBi202 range from 5060 to 5063. You might have to look again at port forwarding.

You can set UserAgentPort to other values, I use 5070 and 5071 in my OBi110 to try to avoid sip scanners.

[rsriram22]

keep rocking @qbzappy.. your solution worked for me.. i had same issue as @hortoristic (sip forwarding from CC to my obi, call comes through,including caller id and then no voice after that from both directions)

..i fwded both ITSP profiles port sets in my router (just in case)

On the SPx account, you may need to forward these ports as well.

ITSP Profile X
    General
    SIP
    RTP <------------ Port Forward this range. Note that these should be different for every SPx account.

[EVRMINC]

Port forwarding as: 5060-5062 works but 5060-5062, 17000-17098 won't save on router.

SIP

These could be set up as separate rules
UDP 5060-5062
UDP 17000-17098
SIP setting on the router might be your sip alg which might need to be disabled. Uncheck it. It might be enough to fix everything.

QBzappy:

I am an OBiPLUS Beta user and use two OBi202s for two separate busineses. All is basically ok except I am trying to figure out if I can change the port assignments on the OBi202 to use ports other than 5060-5062 and 17000-17098. Why? I have a Fonality phone on the same LAN that uses 5060-5061, 10000-20000.
Any insight is appreciated, thanks.

[QBZappy]

I think this is what you are trying to control:

Voice Services
    SP1 Service->X_UserAgentPort = Whatever

ITSP Profile X
    General
    SIP
    RTP->LocalPortMin = Whatever
         ->LocalPortMax = Whatever

[lk96]

I recently got a DID from Voxbeam and thought it would be straightforward.

The symptoms I experience are identical to the original posting in this thread.

What I have so far:
1. UDP port forward is configured in the Dlink DIR-655 router I'm using for the appropriate range of ports.
2. SIP ALG is disabled on the router
3. I use SP4 for this service as such I forward incoming SIP traffic to port 5063.
Appropriate rules is in the Dlink router.

As such all suggestions posted in this thread are taken care of.

my URI looks like: 1234567890@name.dyndns.org:5063

Incoming calls ring the phone attached to the 202 at the correct service (SP4). But once I answer no audio
is there in either direction.

Checking the call status in progress I see one strange thing: the Rx audio codec is empty.
The Tx audio codec is G711u. The Rx stats keep incrementing. But the Tx stats are 0.

Anybody has any ideas/thoughts of what may be the problem ?

thanks

L.

[ianobi]

lk96,

It looks like an RTP problem. I would recheck that you have forwarded the correct UDP range for SP4.

If that's ok, then it might be worth trying STUN:

Service Providers > ITSP Profile D > General:
STUNEnable: check
STUNServer: stun.ideasip.com  (or stun server of your choice)

[lk96]

ianobi:

thanks. Have cross checked that the appropriate port ranges
are open on the firewall.

And I already had STUN setup (pointing to ekiga.net). Will retry a different STUN server
(just for the heck of it). But I suspect the result wouldn't be much different.

I even put the obi202 device on the DMZ to see if that makes a different: no change either.

So either the Dlink router/firewall is acting up or i have some different type of issue.

thanks again for the suggestions

L.

[azrobert]

You can try some config settings discussed here:
http://www.obitalk.com/forum/index.php?topic=5987.0

[lk96]

azrobert

thanks for the pointer.

My issue was resolved by entering my actual public IP address, as indicated
in some of the comments in the post you forwarded,
in the X_publicIPAddress manually.
After that, things started working fine. So that was it !

So this is strange given that I tried both STUN and ICE and apparently both of them
were probably sending the internal obi IP address. But apparently others have experienced
this issue before.

L.

[hwittenb]

azrobert

My issue was resolved by entering my actual public IP address

So this is strange given that I tried both STUN and ICE and apparently both of them
were probably sending the internal obi IP address. But apparently others have experienced
this issue before.

ik96

The sip protocol has buried in the packet the ip address the caller should use for a contact response.  Your problem here has to do with the undocumented OBihai technique they employ as to whether to send a sip response showing an external ip address or a local network address. 

You fixed the problem by setting the numeric public IP address.  I can also duplicate your problem using an IPKall DID by setting up a voip account in SP4.  The voip account doesn't necessarily have anything to do with your Voxbeam DID.  If you check the setting to Register the voip account the incoming call using the DynDNS symbolic ip address will work fine.  If you uncheck the setting to Register the voip account the incoming call will not have audio due to the OBi sending a local network ip address in the sip 200 OK response when the call is answered.

Using a STUN server shows the OBi adapter the external ip address, but OBi does not use it in certain cases.  In fact the Admin manual says:

It should be noted that the STUN feature used in this context is only for RTP packets, not SIP signaling packets (which typically does not require STUN).

[lk96]

Thanks for the correction about STUN.

I had actually attempted to enable registration. But it seems that Voxbeam doesn't support SIP registration.
So I unchecked that option since it was failing to register.

Unless you are saying that it's ok to enable registration
even if the voip provider doesn't support it, with the whole intent for obi to send the external/public
IP address as part of the SIP message.

[lk96]

I just attempted what hwittenb hinted:

I enabled registration with the VOIP provider even though my provider doesn't
support registration (when I go to the Status page of the Obi, it shows that "register failed" for SP4).
And on top of that, I removed from the X_PublicIPAddress my external IP address.

Apparently things continue working which confirms that the Obi will sent the public IP address
when registration is enabled, regardless if registration succeeds or not. And apparently
the Obi was sending the internal IP address when registration was disabled.

This was quite enlightening ...

[hwittenb]

It's good to know that it works both for register succeed and register fail.  I didn't previously test register fail.

The OBi will continue to try to register.  If you had any other voip provider that would register you could configure that configuration under SP4 to register successfully.  You could get a free CallCentric or Sip-to-Sis account if you wished.  It wouldn't affect your incoming calls.

You previous stated that you forwarded the sip signalling port for SP4 in your router to the OBi202.  It should be noted for anyone else that this forwarding is required in this type of use.