[Newbie] Forward Calls to Landline, Answer Using SIP Client/App/Other Methods

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ALAL:
Thank you ianobi for your help.
1. I'm currently testing it under different situations and connections (wifi vs 3G vs landline with different phones).  Thanks.
2. Currently, it's set to 4000, CallerIDMethod is FSK(Bell 202).  Under these settings, the Caller ID can be detected and seen.  I played around with those variables, such as lowering the RingDelay and changing the CallerIDMethod, but nothing seems to work.  The country is Hong Kong, which I think (not sure) should follow the UK standard.  If you know the appropriate settings, please let me know.  I've done some search but can't find any posts relating to this issue.
3. Calling out using CSipSimple:
I tried to dial out, and it says "404 / User not found".  The current filter settings are:
a. All Prefix by **7
b. All Suffix with [SIP2SIP username].[DDNS address]:[port]

In regards to 3.b., should it be "All Suffix with [DDNS address]:[port]"?

I further tested with different combinations without any filters set up:
- **7[phone number]@[SIP2SIP username].[DDNS address]:[port] => "500 / Server error occurred"
- **7[phone number]@[DDNS address]:[port] => "486 / Busy here"

Just to clarify, what exactly is "**7"?
Thanks.

ianobi:
2. Wikipedia seems to think that Hong Kong uses the same CallerID format as USA (FSK Bell 202) so you should be ok on the OBi default settings. Unfortunately, that means you are stuck with the four seconds delay to decode the CallerID. If you were willing to allow all calls to be forwarded to your cell phone regardless of CallerID, then the RingDelay could be set to 0 and this changed to:

Physical Interfaces > LINE Port > InboundCallRoute:
{sp2(alalalal@sip2sip.info)}

This would send all calls incoming on the line port to CSipSimple on your cell phone.

Or:

Physical Interfaces > LINE Port > InboundCallRoute:
{ph,sp2(alalalal@sip2sip.info)}

This would send all calls incoming on the line port to CSipSimple on your cell phone and ring the phone attached to the OBi110.

I’m not sure if CallerID will make it through the OBi110 to CSipSimple using these methods.


3. Calling out using CSipSimple:

The correct formats are:

a. All Prefix by **7
b. All Suffix with @[DDNS address]:[port]


This is hopeful:

**7[phone number]@[DDNS address]:[port] => "486 / Busy here"

It suggests that the call got the OBi110, but the format or CallerID was not correct.

**7 is just my invented prefix.  It needs to be anything that the sip2sip servers do not recognise as a number that they should process, so they pass it through with no processing. The main thing here is that the prefix we add from CSipSimple needs to be the same as we remove in the OBi110 InboundCallRoute. We added **7 at the CSipSimple end so we remove it here at the OBi110 end:


Voice Services > SP2 Service > X_InboundCallRoute:
{(alalalal)>(<**7:>(xx.)):li}

Any call coming into sp2 with a CallerID of “alalalal” and a prefix of **7 will have the prefix **7 removed and then be sent out of the Line Port to PSTN.

I don’t know what you are actually using as your CallerID (where I have used alalalal). If it contains reserved characters (m, M, s, S, x, X which have special meaning in the digit map syntax) then this can cause problems. If that is so, then use single quotes like so:

Voice Services > SP2 Service > X_InboundCallRoute:
{(‘alalalal’)>(<**7:>(xx.)):li}

ALAL:
thank you!!!
let me try these settings first. i'll let you know if it works.
just a quick question, is there an iPhone/iOS equivalent of CSipSimple?
Thanks!

ianobi:
I'm no expert on iOS, but Acrobits seems popular. It has number rewriting like the CSipSimple filter rules. It costs a few $ and like CSipSimple, if you want to use it on 3G, then I recommend spending a few more $ on the G729 codec.

ALAL:
Just want to share with everyone, after months of using this. I successfully made and received calls.  However, the call quality varied.  I tried testing on T-mobile (US) using 2G/3G, and wifi.  The calls are sometimes acceptable but there were some interruptions in between or dropped calls.
Used:  Samsung S3, Obi110, CSipSimple (native phone support for android, which is nice)

Question:  Does anyone know how to use an iOS SIP client for this purpose?  If so, I would be grateful if someone could give me some pointers.

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