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Struggling with a DigitMap/Outgoing Route for UK...

Started by agsteele, July 02, 2018, 01:04:24 AM

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agsteele

I've been reconfiguring my OBi device to help my daughter with her move to using just an IP phone and abandoning her landline.

All has gone well in the initial phase. I got regular calls passing out to the service provider (Sipgate UK) and receive incoming calls OK.

However, I've been struggling with anything slightly out of the unusual.  For example Sipgate has voice mail by dialing 50000. I can't seem to pass the number to Sipgate. Just silence.  I can map 50000 to, for example 1571, and that passes through OK but 50000 just sits with silence for much longer than the 10 sec timeout. (1571 generates an error since that number isn't supported by Sipgate). Calls to the Sipgate test number 10000 also fail.

I've found lots of useful posts on Sipgate Digitmaps which I've borrowed from without success.

I'm fearful that even emergency calls (999/112) might not pass through.

Perhaps I should bin my trusty old OBi and get my daughter something else but if anyone can provide insight before I do then I'd be immensely grateful. The OBi configuration is very powerful but I'm clearly struggling to get things going. 

I'm using SP1 for ALL calls. There is nothing connected to the PSTN.  All calls including emergency calls should go through SP1.  So the only code that needs to be captured is *** to access the voice menu via the attached phone.

ISTPA DigitMap:

(999S0|112S0|50000S0|0[1237]xxxxxxxxxS0|<01392>[2-9]xxxxxx|0[4568]xx.|00xx.|xx.)

Phone:

DigitMap:

([1-9]x?*(Mpli)|[1-9]S9|[1-9][0-9]S9|999|112|**0|***|#|**1(Msp1)|**2(Msp2)|**9(Mpp)|(Mpli))

OutboundCallRoute:

{([1-9]x?*(Mpli)):pp},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{(<**9:>(Mpp)):pp},{(Mpli):pli}

drgeoff

#1
1. Your mention of PSTN implies you have an OBi110. Those default to having the Primary Line set to  LINE. Try changing it to SP1.  I suspect the silent calls are being routed to the LINE port.

2. You can see what digits are being sent and to where by looking at Call History. Point a browser at the OBi's IP address on your LAN. Log in, click on Status, then on Call History.

agsteele

Thanks for your comment.  The device is set to default to SP1.

The Call History shows each of the failed calls are being sent from the phone to SP1


Terminal ID           PHONE1     SP1
Peer Name      
Peer Number      50000        50000
Direction           Outbound     Outbound
18:58:11           New Call   
18:58:11                        Call Connected
18:58:52           End Call   


drgeoff

Quote from: agsteele on July 02, 2018, 04:46:56 AM
Thanks for your comment.  The device is set to default to SP1.

The Call History shows each of the failed calls are being sent from the phone to SP1


Terminal ID           PHONE1     SP1
Peer Name      
Peer Number      50000        50000
Direction           Outbound     Outbound
18:58:11           New Call   
18:58:11                        Call Connected
18:58:52           End Call   
Strange.  That should work.  I have a Sipgate UK account and I tried 50000 and 10000 immediately prior to typing this.  Both worked, returning the recorded male messages.

drgeoff


agsteele


agsteele

Quote from: drgeoff on July 02, 2018, 06:19:47 AM
Strange.  That should work.  I have a Sipgate UK account and I tried 50000 and 10000 immediately prior to typing this.  Both worked, returning the recorded male messages.
Would you be willing to share your DigitMaps for SP1 and Phone and the OutboundCallRoute for the PHONE?

I have a Linksys unit here for my own lines and when I put my daughter's credentials in the LinkSys the Voice Mail number works as it should.

So there is definitely something adrift.


azrobert

Your call history shows the call correctly routed to SP1 and it connects. This indicates your digit map and outbound route are correct. You shouldn't have audio problems because regular calls work. Try displaying the Call Status when you have an active VM call. Maybe this will give you an idea what's wrong.

agsteele

I have made a little progress.  I think I may have a one way audio issue when I initiate calls.  The audio goes out but I hear nothing coming back which is why I don't hear the test calls or VoiceMail.

At least I have a better sense of what might be happening.

I'll report back when I find more information to complete the history.

Thanks for the help.

Andrew

agsteele

I'm pleased to report that the issue was with the RTP port range rather than my DigitMaps.

Thanks for all the advice that nudged me in the right direction.