[Newbie] Forward Calls to Landline, Answer Using SIP Client/App/Other Methods
ianobi:
How you make changes to your OBi config is a big subject. There are two methods, both have pros and cons. For now I suggest you stay with the Obihai preferred method, especially as your OBi may be remote from you:
Make changes 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.
1. The above info and giqcass’ advice should have cured this.
3. There are many ways to go! If you don’t want to go with a paid account, then I suggest this for android:
Download the CSipSimple app to your cell phone (free).
Register for a sip2sip account (free). Install it on CSipSimple. I’ll assume your sip2sip account id is alalalal@sip2sip.info
Now your cell phone can be called via your sip2sip account. Your OBi110 now needs a “fake” SIP account, I’ll use sp2:
Service Providers -> ITSP Profile B -> SIP -> ProxyServer : 127.0.0.1
Service Providers -> ITSP Profile B -> SIP -> X_SpoofCallerID : checked
Service Providers > ITSP Profile B > General > DigitMap:
([^*]@@.'@'@@.)
Voice Services -> SP2 Service -> Enable : (checked)
Voice Services -> SP2 Service -> AuthUserName : Something (don’t leave blank)
Voice Services -> SP2 Service -> X_RegisterEnable : (unchecked)
Voice Services -> SP2 Service -> X_ServProvProfile : B
Voice Services -> SP2 Service -> X_UserAgentPort : 5061
Voice Services -> SP2 Service -> CallerIDName : Whatever
Voice Services -> SP2 Service -> MaxSessions : 4
Change the Line Port InboundCallRoute:
Physical Interfaces > LINE Port > InboundCallRoute:
{12345678901:sp2(alalalal@sip2sip.info)},{ph}
Now incoming calls to your OBi110 with CallerID 12345678901 should call your cell phone via your sip2sip account on CSipSimple.
There are other settings on CSipSimple to ensure it’s “kept alive” etc, but we can do more detail if you choose to go this way and get this far! Sorry, it’s quite a learning curve at the beginning. The good thing about CSipSimple is that it does integrate with the native android dial pad and will use your existing contacts.
4. Making an outgoing call from your cell phone via the OBi110:
OBiON app method: In your OBi Dashboard click on your softphone number – 290123456 – make sure that you have filled in the “OBi Voice Gateway” box with your OBi110 number – 200123456. Click save, wait for the OBi110 to reboot.
Now anything you dial from the OBiON app on your cell phone will be sent out from your OBi110 using its Primary Line. Default Primary Line for an OBi110 is PSTN Service, so calls will go out via the Line Port. This can be changed if required.
CSipSimple method: In your CSipSimple sip2sip account you need to tell it where your OBi110 is. Either your OBi110 is at a fixed IP address (your router’s public ip address) followed by the OBi110’s port (5061) so 123.12.112.10:5061 or you need a ddns address such as alalalal.ddns.me.com:5061. Then you set up a filter rule for your sip2sip account on CSipSimple, which says “All – Suffix with @alalalal.ddns.me.com:5061.
Now when you dial from your native android dial pad it will ask you to choose between “mobile” or “sip2sip”. Choose “sip2sip” and dial say 12345678901. This will be sent to the sip2sip servers as 12345678901@alalalal.ddns.me.com:5061 and your router will remove all but 12345678901 and send that number to port 5061, which is the OBi110 sp2. (Phew, nearly there!)
Change sp2 InboundCallRoute:
Voice Services > SP2 Service > X_InboundCallRoute:
{(alalalal)>(xx.):li}
Any call coming into sp2 with a CallerID of “alalalal” will be sent out of the Line Port to PSTN. This can be changed to include ringing the OBi110 Phone Port, or connect to the Auto Attendant etc.
There are some potential problems. For example sip2sip won’t allow through numbers starting with “0”. Many countries (including where I live) use “0” rather than “1” as a national prefix. This can be overcome, but you have enough to think about for now!
As said earlier, these are only two options – using OBiON or CSipSimple. Both are free, but obviously use your data plan. The OBiON app is a bit clunky and is in need of updating. CSipSimple is a learning curve all of its own, but can be very useful.
5. sip2sip does have a voicemail facility. I’ve not used it with my sip2sip accounts, so I cannot comment on it. I guess we should not expect too much from a free account!
I’m no expert on PBX etc. You may wish to post that separately after you get past all of the other issues in this post.
By now I expect you are banging your head on the nearest wall or have fallen asleep – don’t despair – many of us have been through this and remained almost sane :D
ALAL:
Oh wow, that is a lot to take in. But I'd like to thank both of you, giqcass and ianobi! I tackled most of the problems!!!
1. Thanks giqcass! it's working now!
3. Yay! thanks ianobi. I successfully used csipsimple to receive calls!
4. Outgoing call: having a bit of trouble here, I'm stuck at this stage. I'm trying to keep everything uniform and so i'll use csipsimple only.
CSipSimple method: In your CSipSimple sip2sip account you need to tell it where your OBi110 is. Either your OBi110 is at a fixed IP address (your router’s public ip address) followed by the OBi110’s port (5061) so 123.12.112.10:5061 or you need a ddns address such as alalalal.ddns.me.com:5061. Then you set up a filter rule for your sip2sip account on CSipSimple, which says “All – Suffix with @alalalal.ddns.me.com:5061.
- does it mean I have to change port forwarding settings on the router to the Obi110? So, forward all 5061 to the obi110?
- filter rule: in csipsimple, under "filters" > "sip2sip" > "add filter" > ??? Do I select "can't call" "rewrite" "stop processing" "directly call" "Auto answer"? Then, "All"? After selecting "all", I can't add "All – Suffix with @alalalal.ddns.me.com:5061"
There are some potential problems. For example sip2sip won’t allow through numbers starting with “0”. Many countries (including where I live) use “0” rather than “1” as a national prefix. This can be overcome, but you have enough to think about for now!
Can I set sip2sip or csipsimple to dial a non-zero prefix before the actual phone number? say "123"-[actual phone number]?
Thanks!!!!
ianobi:
CSipSimple is a complex beast, but that’s because it is very configurable, which suits our needs. Here’s some general settings I recommend in addition to the defaults:
Settings > Network > Lock WiFi: check
Settings > Network > Hi perfs lock: check
Settings > Network > Resolve DNS SRV: check
Settings > User Interface > Use partial wake lock: check
Quote
filter rule: in csipsimple, under "filters" > "sip2sip" > "add filter" > Huh Do I select "can't call" "rewrite" "stop processing" "directly call" "Auto answer"? Then, "All"? After selecting "all", I can't add "All – Suffix with @alalalal.ddns.me.com:5061"
Like me, I think you need two filter settings:
Settings > Filters > sip2sip (account name) > Add Filter > Rewrite > All > Prefix by: **7
Settings > Filters > sip2sip (account name) > Add Filter > Rewrite > All > Suffix with:
@alalalal.ddns.me.com:5061
Now if you dial 12345678901 from CSipSimple, then **712345678901@alalalal.ddns.me.com:5061 will be sent to your router. **712345678901 will make it through to your OBi110 sp2.
Change to this:
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.
Quote
does it mean I have to change port forwarding settings on the router to the Obi110? So, forward all 5061 to the obi110?
Yes, it’s recommended to port forward the “UserAgentPorts”. By default these are 5060 for sp1 and 5061 for sp2. Personally I like to change the UserAgentPorts to a random number as it helps to avoid sip scanners. You might pick something non-standard like 5490 and 5491. NB: If you do change the UserAgentPort for sp2 you must change the CSipSimple filter rule to agree with it.
If you wish to use this method with a 3G data plan, then it would be worth investing a few $ and buying the app CSipSimpleCodecG729. I find it works pretty well with 3G. Set the codec settings in CSipSimple so that 3G is classed in the “Slow” list. Then set the slow list to prioritise G729 codec. Your OBi110 also supports G729 so you could also prioritise that in the relevant OBi110 codec list. Sip2sip does not support the G729 codec, but that’s ok as it will simply pass the call through with any codec that’s being used.
I don’t think that any of us got all this working on our first attempt! Took me a while, and I’m still on a bit of a learning curve regarding CSipSimple!
ALAL:
Going back to receiving calls on CSipSimple, I have two problems:
1. We can't hear each other
I managed to receive phone calls using CSipSimple BUT sometimes I can't hear anything at all, and same with the caller. Or sometimes it's one way only, meaning I can hear what the caller is saying, but the caller can't hear me at all. So it's really on and off.
2. CSipSimple or Obi110 doesn't pick up the call right away
When the phone call is redirected/call-forwarded to my landline (connected to Obi110), the phone call isn't picked up right away, the caller needs to let it ring a few more times before CSipSimple picks it up. Is there a way to make CSipSimple or OBi110 pick it up right away?
Side note: I'm still trying to figure out the dialing out part. I think my version of CSipSimple is different from yours, I don't see settings you're recommending. The version I'm using is 1.02.00 r2330.
Thanks!
ianobi:
1. This is usually a NAT traversal problem. Try to do your testing when CSipSimple is connected via your cell phone wifi connection. If this works reliably, then your router / OBi are probably set up correctly. Make sure you have the correct ports forwarded in your router. Obihai recommend:
TCP Ports: 6800, 5222, 5223
UDP Ports: 5060 to 5061, 10000 to 11000, 16600 to 16998, 19305
This assumes that you are still using 5060 & 5061 as your sp1 & sp2 UserAgentPorts. For speech 16600 to 16998 are the most important – these carry speech data using RTP (Real Time Protocol). To test if port forwarding is the problem, you could put the OBi LAN connection in your router’s DMZ temporarily. This would leave it totally open to the internet for testing purposes.
2. An incoming call to the OBi110 Line Port is subject to a delay set by:
Physical Interfaces > LINE Port > LINE Port > RingDelay
The default is usually 4000 (4 seconds). This is to allow time for the OBi110 to decode the incoming CallerID. In North America CallerID is sent after the first ring and 4 seconds allows enough time to receive the first ring followed by the CallerID. Where I live in the UK CallerID comes in before the first ring, so I can set my delay to 0. If you tell us which country your OBi110 is in, then we should be able to work out what delay if any is needed. Or it might just be a question of trial and error. CallerIDMethod also needs to be set correctly for your country.
Apologies – I should have told you this part sooner: I am using the same version of CSipSimple as you, but here is yet another of its complications, you need to see “Expert Settings” See here:
https://code.google.com/p/csipsimple/wiki/ExpertSettingMode
Having said all that, the filters should work as I described previously.
CSipSimple is so not simple that it has a web site of its own including its own wiki:
https://code.google.com/p/csipsimple/
Another way to test a call from CSipSimple to your OBi and out to PSTN, is to set up a contact in the format **712345678901@alalalal.ddns.me.com:5061 and send it without any filters set up. You will need to use the CSipSimple dial pad to do this if your native key pad does not allow letters/symbols.
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