UK Obi110 PSTN working fine, SIP inbound calls ring but cannot answer them

[derketo-rick]

UK, using BT landline and DECT hansets, PSTN connection absolutely fine.

Using Voipfone, SIP registers fine and I can call out on SIP fine with **1 prefix, but with 40 second delay before other end starts ringing (with silence in handset earpiece whilst waiting).

I would like to shorten that delay significantlly, so any suggestions welcome.

Main problem is for inbound SIP calls - handset rings but answering on handset does not answer the SIP call - eventually times out to voice mail.

I may have firewall port forwarding issues or config issues - do not know how to tell.

I know there is insufficient data here but if anyone has seen similar issues please advise what to try next, but otherwise can someone tell me what config data/pages I need to post here to help diagnose my problems and/or where I can activate/see logging in sufficient detail to figure out what is happening ?

Thanks very much.

This looks like a great little box with thousands of options but I am a bit lost as to where to start diagnosing these two problems.

[ianobi]

derketo-rick - welcome to the forum.

This may take a few postings. I'm also in the UK, though I use sipgate.co.uk as my main sip service.

UK, using BT landline and DECT hansets, PSTN connection absolutely fine.

I bet we can reduce outgoing and incoming delays here (digitmap tuning), but that can wait for a while.

Using Voipfone, SIP registers fine and I can call out on SIP fine with **1 prefix, but with 40 second delay before other end starts ringing (with silence in handset earpiece whilst waiting).

Ten seconds of the delay is probably digitmaps again. If not already set, then try setting Outbound Proxy:

Service Providers > ITSP Profile A > SIP > OutboundProxy: sip.voipfone.net
Service Providers > ITSP Profile A > SIP > OutboundProxyPort: 5060

Main problem is for inbound SIP calls - handset rings but answering on handset does not answer the SIP call - eventually times out to voice mail.

Outboundproxy may have helped with this. More likely is that your router is blocking the ports used for RTP, which is the speech part of the call. Most routers will allow you to put one or more devices in "DMZ". This effectively forwards all ports to the device. If you need to find the local ip address of your OBi, then dial ***1 from the phone connected to the OBi. If this fixes things, then take the OBi out of DMZ and set up port forwarding in your router:

TCP Ports: 6800, 5222, 5223
UDP Ports: 5060, 5061, 10000 to 11000, 16600 to 16998, 19305



Make changes to your OBi settings via the Obi Expert Configuration pages. From your OBi Dashboard, click on your OBi number and follow the prompts to get there. To change a value uncheck both boxes to the right of the value and leave them unchecked. After changing the values on one page, press submit at the bottom of the page and wait a few minutes for the Obi to reboot. Then move on to the next page if required.


Hopefully, the above will get your OBi working roughly, then we can do the fine tuning!

[derketo-rick]

Thanks for that ianobi

I already had the outbound proxy set up as you described.

I put the various ports you suggested into the router for pass through and that was exactly the same for inbound - phone rings but pickup does not work. In case we missed one I also put the Obi into DMZ

Logs show pickup, but handset does not receive the call (ringing continues, voipfone eventually goes to voicemail:

|Call 1|---|09/11/2013 00:18:33|   (ignore time - unit time not set up)
|Terminal ID|---|SP1|---|PHONE1|
|Peer Name|---|07767******|---||
|Peer Number|---|07767******|---||
|Direction|---|Inbound|---|Inbound|
|00:18:33|---|Ringing|---||
|00:18:37|---||---|Call Connected|
|00:18:44|---||---|End Call|

Any idea which parameters affect the signalling of pickup to the SIP provider ?

Thanks again.

Rick

[ianobi]

Rick,

Try and have a look at Status > Call Status while the call is ongoing during the "ringing" stage. Have a look at the rows showing Peer RTP Address and Local RTP Address. Both should have an ip address:port showing. Test while in DMZ.

The next step might be to set up STUN, but I seem to remember that voipfone does not like that idea.

[derketo-rick]

Thanks again for your help.

I had set up STUN - and turning it off fixed the problem.

I now have both voipfone and sipgate connected OK and working in both directions.

There is still a delay when dialling out on SIP - what should I try to reduce that ?

Thanks again
Rick

[Shale]

There is still a delay when dialing out on SIP - what should I try to reduce that ?

Step 1 would be to enter # after you have finished dialing other digits. If that eliminates the delay you are talking about, then you could eliminate the need for dialing the # by adjusting a string or two.

[ianobi]

Rick,

The starting point is to decide which service will be your "Primary Line". That is the default line where you don't have to dial any ** codes. The aim is to set up digitmaps that match the number formats for each service. This will reduce or eliminate delays. The Primary Line can be used to automate the routing of calls without the need for ** codes. Here is a UK setup that uses sp1 as the Primary Line.

http://www.obitalk.com/forum/index.php?topic=5362.msg34764#msg34764

Have a read through and come back with any questions specific to your setup.