Setting up an Obi 202 + Obiline in the UK (Draft)
Mouse:
I have just added another UK setting, adding 17070 (the BT test number) to the list of outbound numbers (like 999) passed through to PSTN in the dialplan. I have added this in phone 1 and Phone 2 settings - is that right?
(It would seem to me there needs to be somewhere to put dialplan settings that apply to all outbound/inbound calls to avoid duplication. Is that what gateways are for?)
17070 does not work with my sip provider.....
ianobi:
Quote
(It would seem to me there needs to be somewhere to put dialplan settings that apply to all outbound/inbound calls to avoid duplication. Is that what gateways are for?)
In OBi-speak a dial plan is a DigitMap. If you use the same DigitMap in several places, then it's useful to create a User Defined DigitMap. For example:
User Settings > User Defined Digit Maps > User Defined Digit Map2 >
Label: tel
DigitMap: (0[4568]xx.S3|0[123]xxxxxxxxx|07[1-9]xxxxxxxx|116xxx|xx.)
Replace existing DigitMaps with the new User Defined DigitMap:
Service Providers > ITSP Profile A > General > DigitMap:
(Mtel)
Service Providers > ITSP Profile B > General > DigitMap:
(Mtel)
Any change made to the User Defined DigitMap "tel" will automatically be made to all DigitMaps defined as (Mtel).
Voice Gateways are a different subject. They are used to provide access to VOIP service providers that do not require registration and are used for outgoing calls only. In the UK I use registered VOIP providers on various spXs, but I use a Voice Gateway for a voipcheap.co.uk service which I use for outgoing mobile calls as they only charge 3p per minute. In the UK DigitMaps can be arranged to automatically route any number beginning with "07" to any specific Voice Gateway.
Mouse:
Quote from: ianobi on July 18, 2015, 02:22:51 am
Quote
(It would seem to me there needs to be somewhere to put dialplan settings that apply to all outbound/inbound calls to avoid duplication. Is that what gateways are for?)
In OBi-speak a dial plan is a DigitMap. If you use the same DigitMap in several places, then it's useful to create a User Defined DigitMap. For example:
User Settings > User Defined Digit Maps > User Defined Digit Map2 >
Label: tel
DigitMap: (0[4568]xx.S3|0[123]xxxxxxxxx|07[1-9]xxxxxxxx|116xxx|xx.)
Replace existing DigitMaps with the new User Defined DigitMap:
Service Providers > ITSP Profile A > General > DigitMap:
(Mtel)
Service Providers > ITSP Profile B > General > DigitMap:
(Mtel)
Any change made to the User Defined DigitMap "tel" will automatically be made to all DigitMaps defined as (Mtel).
Thanks that's quite neat.
Quote
Voice Gateways are a different subject. They are used to provide access to VOIP service providers that do not require registration and are used for outgoing calls only. In the UK I use registered VOIP providers on various spXs, but I use a Voice Gateway for a voipcheap.co.uk service which I use for outgoing mobile calls as they only charge 3p per minute. In the UK DigitMaps can be arranged to automatically route any number beginning with "07" to any specific Voice Gateway.
OK like the SPA 3102 in a way then
Thanks very much for your feedback
Mike
Mouse:
Hmm I said above that the inbound PSTN echo problem is solved with the UK settings but maybe not quite.
1. A loud sound with a sharp rise time still attracts an echo, though it is truncated and diminished by the canceller.
2. The echo canceller still seems to have the SPA3102 problem of giving a half duplex flavor to conversations. When you start talking the voice of the person at the other end gets a bit suppressed.
1. is at an acceptable level I think but my wife reports that 2 is problematic as it leads to missing things that people are saying, and people talking over each other without realizing. Both effects I think are better than on the SPA.
Any ideas (if symmetric RTP reduces signal delay then maybe it will help with echo and hence suppression symptoms?)
On the SPA the best approach to reducing the echo tail was to reduce the RTP packet size or Jitter buffer size as this reduced processing lag through the device. You could then sometimes switch off the canceller. I guess packet size is G711a/u packetisation period on the 202, but I dunno if this applies to the Line input. Can you switch off the canceller and adjust the jitter buffer - I cannot find the settings......?
Maybe preferring 711u would help as the device is probably optimized for it?
On PSTN does inbound work like the SPA? Processing the analogue signal into a digital one, then back to analogue. This was the source of delay on the SPA.
Kind regards
Mike
Mouse:
I just changed CLI trigger on phone 1 and phone 2 to 'before first ring', which I think is correct for the UK. Also changed in the setup guide in the first post.
This solved a problem with incoming calls constantly being detected as unanswered and thus added to unanswered call logs for me.
Did not seem to be in UK settings file?
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