CSipSimple + Sip2Sip + Obi202 + Google Voice
azrobert:
said by ianobi:
Quote
Voice Services > SP1 Service > X_InboundCallRoute:
Delete: ph,ph2
Insert: {sp3(userid@sip2sip.info;ui=$1),ph,ph2}
then
Edit: drop the ui=$1 sip2sip does not like it. Further research needed.
I am also researching why ui=$1 doesn't work with SIP2SIP.
I did come up with 2 solutions that provide CallerID.
Solution 1
Get a free account at PBXes.org.
Define an extension.
Under Device Option change the Dial option to "SIP/userid@sip2sip.info"
Define an InboundRoute to route all calls to this extension.
Change OBi X_InboundCallRoute to {spx(userid-exten@pbxes.org;ui=$1),ph}
Downside: Adds another service provider.
Solution 2
Change OBi X_InboundCallRoute to the default.
Get a free IPKall phone number.
Config IPKall to SIP URI = userid@xx.xx.xx.xx:50xx (your home external IP address and OBi port#)
Add the IPKall number to your GoogleVoice Settings.
After you verify the IPkall number to GoogleVoice change the IPKall definition to URI = userid@sip2sip.info
In GoogleVoice Settings forward all calls to GoogleChat and IPKall
Downside: IPKall sometimes does not provide the correct CallerID.
Edit: You can skip pointing IPKall to the OBi and verify it on your Android handset.
ianobi:
@ azrobert
Yes, it's odd that sip2sip does not seem to allow the ui=$1. I chose sip2sip originally because (a) its free and (b) has a very good debug facility. I may try another provider.
I would like to keep the number of accounts / providers to a minimum. On that topic I have a few more comments, but I'll go back to the original thread:
http://www.obitalk.com/forum/index.php?topic=4647.0
threehappypenguins:
It worked!!! At least the calling TO my Android part!!! I was confused when you said to set up another sip2sip account. That threw me off, but I just assumed that I wasn't picturing the whole process properly, and just went with what I thought you were saying. So then I used my original sip2sip account, and disregarded the second one I created. That, and getting rid of the ;ui=$1 did the trick. I was able to call the Google Voice number (borrowing my in-law's account for testing purposes), and it rang on the Obi and the Android at the same time, like you said it would. Only the caller ID that shows is always the one that I set up to be my CSipSimple caller ID. Also, I checked Status > Call History and saw it successfully being forked to the sp3 and to the Sip2Sip account from there.
When I tried to implement the rest of your suggestions, there were several issues:
1. Speed dial will not work. I get the reorder tone (fast busy signal) when I dial 10#. I am not sure if this has to do with anything, but Trevor from AcroVoice knows my set up for Google Voice and set up the DigitMap and OutboundCallRoute for me so that when I dial a local number I only need 7 digits (we only have one area code out here in Nova Scotia), and when I dial long distance, I dial 1+area code+7 digits. And the primary line was (and normally will be) SP1, but for testing purposes since I don't have AcroVoice set up yet, I changed it and set up the Primary Line to SP2.
Under Physical Interfaces > PHONE1 Port
DigitMap: (!**5S0|[2-9]11S0|1[2-9]xx[2-9]xxxxxx|[2-9]xxxxxx|0|011xx.|*xx|***|#)
OutboundCallRoute: {(<#:>):li},{***:aa2},{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{([2-9]11S0|[2-9]xxxxxxS0|011xx.|0|*xx):sp1},{(1[2-9]xx[2-9]xxxxxx):sp2}
2. Adding the prefix and suffix in CSipSimple was confusing. I Googled how to do this, and I think I figured it out. What I did was:
Settings > Filters > Sip2Sip > Add Filter / rewrite rule
Changed Can't Call to Rewrite
Changed Starts with to All
Changed Replace match by to Prefix by (and then Suffix with when making the separate suffix rule)
The other thing that confused me is the Prefix rule that you wanted me to create. What exactly am I putting in there? You said:
Quote
suffix all with @happypenguin.com:5062 or fixed IP address:5062
I tried putting in my Obi's ip address (is that what you want? The Obi202 address that I made static?). So for example: 192.168.x.xx:5062
As for the mentioning of @happypenguin.com, do you mean I am putting in my User ID for sip2sip? So if my user ID for sip2sip is represented by userid, then it would be @userid.com? I am confused.
3. The port forwarding threw me off too. I don't know how to do port forwarding, nor do I understand it. I mean, I think I understand why you want me to do port forwarding. Is it so that I can make a call from my Android to my Obi while my Android is connected to a separate Wifi than the home network my Obi202 is connected to?
I went to Router Configuration > Port Forwarding and then stared blankly at it for a while. I considered for a moment to choose Both under Protocol, and then type 5062 under StartingPort and 5062 under EndingPort thinking that this would cover all bases... but then I looked at ServerIPAddress and thought "What Server? What do I put in here?" It had a spot in it to put in the last one or two number(s) of an IP address. Plus, I didn't know what RuleDescription was for, and whether it was just simply a label for the port forward thingy that I am supposed to create or if it was actually some sort of code or something. So I just left the port forwarding issue alone for now. I am not sure if that affected anything since currently I am testing while connected to my home network on my android.
4. I did not understand you with all that Mcot stuff. You said that a cot is a User defined Digit Map. To tell you the truth... I don't even know what a Digit Map is either. All I know about it is if you put some stuff in the DigitMap part of PHONE1 Port, then it tells the phone what will happen when I dial certain digits or amounts of digits. So... it is a map for dialing. That is to the extent of which I understand. So when you say that your cot has 3 caller ID's in it, I have no idea what you are talking about... so I took a stab at what you were saying, and changed the first Mcot to say (assuming that my sip2sip user name is represented by userid):
Muserid
So I ented under X_InboundCallRoute under SP3:
{(Muserid)>(<**7**1:>(Msp1)),(Mcot)>(<**1:>(Msp1)):sp1},{(Mcot)>(<**7**2:>(Msp2)),(Mcot)>(<**2:>(Msp2)):sp2},{(Mcot)>(<**7**3:>(Msp3)),(Mcot)>(<**3:>(Msp3)):sp3},{(Mcot)>(<**7**4:>(Msp4)),(Mcot)>(<**4:>(Msp4)):sp4},{(Mcot)>(<**7**9:>(Mpp)),(Mcot)>(<**9:>(Mpp)):pp},{(Mcot)>(<**7:>(**0)),(Mcot)>**0:aa},{(Mcot)>(<**7:>(***)),(Mcot)>***:aa2},{(Mcot)>(<**7:>(Msp2)),(Mcot)>(Msp2):sp2},{(Mcot)>(<**7:>(0)),(Mcot)>0:ph,ph2},{ph,ph2}
That didn't work (404/user not found was the error when I dialed with the native dialer, choosing Sip2Sip), and I didn't understand what you were talking about with the Caller ID's, so I changed every time it said "cot" to "userid" in the whole big line of confusingness that you pasted :-\ and that did not work either.
ianobi:
OK, there was a lot of jargon, numbers and general OBi type stuff to deal with. Be patient, we will get there. It's late here in my time zone, but tomorrow I will try to be less confusing.
I could not live with myself if I turned threehappypenguins into threesadpenguins :)
ianobi:
Firstly, an apology. This subject has been changing as we have been posting, so there has been some confusion. Secondly, please bear with me. On this forum we are all amateur OBi users like yourself just helping each other out, we do not work for Obihai. This is a big subject for someone new to OBi – you are diving into the deep end. Hey, penguins are good swimmers, you should be ok :)
The principles:
Calls from OBi to CSipSimple via sip2sip, leave OBi as:
aaaaaaaa@sip2sip.info
Where aaaaaaaa is your sip2sip username.
sp3(aaaaaaaa@sip2sip.info) tells the OBi to send the call out via sp3.
Calls from CSipSimple to OBi via sip2sip, leave CSipSimple as:
**7xxxxxxxxxxx@yourpublicipaddress:5062
Where **7 is for routing in OBi. xxxxxxxxxxx is the number dialled. @yourpublicipaddress is your router public WAN address. (This may be in format xx.xxx.xxx.xxx. If this is likely to change you may consider using a free dynamic dns provider to get a fixed address something like @penguin.ddns.com). 5062 is the port number associated with sp3 in the OBi, so it tells the router to send the call there.
On arrival at your router, the router will remove the @yourpublicaddress and use 5062 to route the call to sp3 InboundCallRoute. OBi will remove the **7 and route the call according to the InboundCallRoute rules.
CSipSimple:
Assumes a sip2sip account set up already.
Settings > User Interface > Dialer integration : check
Settings > User Interface > Call logs integration : check
Settings > Filters > sip2sip > Add filter / rewrite rule > Rewrite > All > Prefix by : **7
Settings > Filters > sip2sip > Add filter / rewrite rule > Rewrite > All > Suffix with : @yourpublicipaddress:5062
Yes, I agree filters are a bit hard to get the hang of!
Now, using the native android dial pad (not the CSipSimple dial pad), when you select a contact or input digits, you will be given a choice to send using mobile or sip2sip.
Router:
Obihai recommends that you port forward the following ports:
Allow Outgoing:
TCP Ports: 6800, 5222, 5223
UDP Ports: 5060 to 5063, 10000 to 11000, 16600 to 16998, 19305
Allow Incoming on UDP Port: 10000
I’m sorry I’m not much of a router expert. Under “Open Port” mine has fields “Start Port”, “End Port” , “Local Host”. “Local Host” is the OBi address such as 192.168.x.xx
OBi202:
We do not need the sip2sip account set up on the OBi. Please remove those settings and replace with the following:
Service Providers -> ITSP Profile C -> SIP -> ProxyServer : 127.0.0.1
Service Providers -> ITSP Profile C -> SIP -> X_SpoofCallerID : checked
Voice Services -> SP3 Service -> Enable : (checked)
Voice Services -> SP3 Service -> AuthUserName : (any letters or numbers)
Voice Services -> SP3 Service -> X_RegisterEnable : (unchecked)
Voice Services -> SP3 Service -> X_ServProvProfile : C
Voice Services -> SP3 Service -> X_UserAgentPort : 5062
Voice Services -> SP3 Service -> CallerIDName : Whatever
Voice Services -> SP3 Service -> MaxSessions : 4
Voice Services -> SP3 Service -> X_InboundCallRoute (Assumes Primary Line is set to sp2):
{(Mcot)>(<**7**1:>(Msp1)),(Mcot)>(<**1:>(Msp1)):sp1},{(Mcot)>(<**7**2:>(Msp2)),(Mcot)>(<**2:>(Msp2)):sp2},{(Mcot)>(<**7**3:>(Msp3)),(Mcot)>(<**3:>(Msp3)):sp3},{(Mcot)>(<**7**4:>(Msp4)),(Mcot)>(<**4:>(Msp4)):sp4},{(Mcot)>(<**7**9:>(Mpp)),(Mcot)>(<**9:>(Mpp)):pp},{(Mcot)>(<**7:>(**0)),(Mcot)>**0:aa},{(Mcot)>(<**7:>(***)),(Mcot)>***:aa2},{(Mcot)>(<**7:>(Msp2)),(Mcot)>(Msp2):sp2},{(Mcot)>(<**7:>(0)),(Mcot)>0:ph,ph2},{ph,ph2}
User Settings > User Defined Digit Maps > User Defined Digit Map2 > Label : cot
User Settings > User Defined Digit Maps > User Defined Digit Map2 > DigitMap : (aaaaaaaa)
Where aaaaaaaa is your sip2sip username.
No need to change any “Mcot”s, OBi will substitute your sip2sip username by using the cot DigitMap.
Dialling from your android phone native key pad, selecting the sip2sip option to send:
**1xxxxxxxxxxx > Acrovoice on sp1
xxxxxxxxxxx > GV on sp2 (no need for ** code as this is your Primary Line)
**0 > Obi auto attendant
0 > Rings both OBI phones
I’m not sure why your speed dial did not work. No digit maps etc are involved with this format:
sp3(aaaaaaaa@sip2sip.info)
It is a lot of information all in one go. Feel free to come back for more.
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