Using CSipSimple With OBi
duzlu_it:
Has anyone been able to get CSipSimple to work with an Obi110 over 3G/4G without the need for Sip2Sip? I've been working on it for days, but have the same problems as other's have mentioned - no audio when the line is answered. I've used DMZ, STUN, ICE, port forwarding, port triggering, symmetric RTP, turned on and off the firewall API for SIP in my router, etc. etc. etc. It seems like there is a problem negotiating and/or opening the RTP ports when the client is on a 3G/4G internet connection. I would really like to get this to work, as the latency is quite noticeable when going through Sip2Sip.
jjjooonnn:
I'm trying to use CSipSimple for poor quality wifi sources eg: College, Starbucks, McD's... since the OBi app isn't exactly up to par.
I know nothing!~
How do I use Mcot in all this? Is it: {(Mcot:username@sip2sip.info) or just {(username@sip2sip.info) and is it in all the spots that say (Mcot)?
Looking in the SIP settings in OBi's interface, where can I find sip2sip's URI? In sip2sip account info is it XCAP Root?
And finally: Is the UserAgentPort the port that connects to OBi?
Thank you, for any help!
Quote from: ianobi on November 25, 2012, 03:26:42 am
{(Mcot)>(<**7**1:>(Msp1)),(Mcot)>(<**1:>(Msp1)):sp1},{(Mcot)>(<**7**2:>(Msp2)),(Mcot)>(<**2:>(Msp2)):sp2},{(Mcot)>(<**7**8:>(Mli)),(Mcot)>(<**8:>(Mli)):li},{(Mcot)>(<**7**9:>(Mpp)),(Mcot)>(<**9:>(Mpp)):pp},{(Mcot)>(<**7:>(**0)),(Mcot)>**0:aa},{(Mcot)>(<**7:>(***)),(Mcot)>***:aa2},{(Mcot)>(<**7:>(Mli)),(Mcot)>(Mli):li},{(Mcot)>(<**7:>(0)),(Mcot)>0:ph},{ph}
Mcot has to contain your sip2sip user name.
Voice Services -> SP2 Service -> X_InboundCallRoute (SP2 must be configured for SIP):
At the OBi end I used sp2 for incoming calls, my UserAgentPort is 5071.
ianobi:
Quote
How do I use Mcot in all this? Is it: {(Mcot:username@sip2sip.info) or just {(username@sip2sip.info) and is it in all the spots that say (Mcot)?
cot is a User Defined Digit Map. If your sip2sip account is 12345678@sip2sip.info, then put it in cot like so:
User Settings > User Defined Digit Maps > User Defined Digit MapX >
Label: cot
DigitMap: (12345678|87654321|11223344)
My cot happens to have three Caller IDs in it. Using this method means you only have to change cot if you add or change Caller IDs, rather than change every reference of Mcot in the InboundCallRoute. cot has to contain your sip2sip user name.
Quote
Looking in the SIP settings in OBi's interface, where can I find sip2sip's URI?
For incoming calls your OBi does not need to know the sip2sip URI. It will route calls based on the “username”, which it sees as CallerID. “12345678” in the example above.
For calls to be forwarded from say an incoming call on SP1 something like this would be needed:
Voice Services -> SP1 Service -> X_InboundCallRoute:
{sp2(userid@sip2sip.info),ph}
In my examples I am using sp2 for incoming and outgoing calls from and to sip2sip.
Quote
Is the UserAgentPort the port that connects to OBi?
Each spX has its own UserAgentPort. For example an OBi110 at default uses sp1 – port 5060, sp2 - port 5061. Many of us change these to avoid sip scanners. In my example I used sp2 and changed the UserAgentPort to 5071. When connecting from CSipSimple to the OBi calls would be routed using @my.ddns.com:5071. The @my.ddns.com reaches my router and the port 5071 tells the router to send the call to sp2 on my OBi.
Most people find that the hardest part of this setup is configuring the CSipSimple filter rules. Good luck!
jjjooonnn:
@ianoboi
Thank you for that! -I finally got it... somewhat working...
All the calls I make from CSipSimple... ring my OBi houseline!
I am receiving calls on CSipSimple (rings my house, cell(forwarded from google voice's page), and CSipSimple. I think its a problem with the ports ??? I'm glad I can get calls!
Here are my settings, I tried to move all the ports to 5071 like you mentioned, but sip2sip doesn't connect unless there are all set to 5060, these have it working for now (except ring all calls placed only ring my OBi houseline)
On CSipSimple I'm using a DDNS with the filter rules (using @ddnsaddress:5060)
ITSP Profile B SIP settings (everything else default)
ProxyServer proxy.sipthor.net
ProxyServerPort 5060
ProxyServerTransport UDP
RegistrarServer sip2sip.info
RegistrarServerPort 5060
UserAgentDomain sip2sip.info
OutboundProxy proxy.sipthor.net
OutboundProxyPort 5060
RegistrationPeriod 600
Under SP2 Service I have the Mcot lines you orginally posted (with cot defined in User Settings as my sip2sip username w/o @sipp2sip.info)
So close! -yet so far way!
ianobi:
jjjooonnn,
There was some confusion in the first few posts of this thread. I may need to go back and sort it out! There is no need for a sip2sip account on the OBi. Only set up sip2sip as an account on the CSipSimple app. If using sp2, then set up the OBi as follows:
Service Providers -> ITSP Profile B -> SIP -> ProxyServer : 127.0.0.1
Service Providers -> ITSP Profile B -> SIP -> X_SpoofCallerID : checked
Voice Services -> SP2 Service -> AuthUserName : (any userid - this is CallerID sent on outgoing calls)
Voice Services -> SP2 Service -> X_RegisterEnable : (unchecked)
Voice Services -> SP2 Service -> X_ServProvProfile : B
With this set up we don’t need another sip2sip account on the OBi as outgoing calls from sp2 will go to the android sip2sip account just fine as it is.
Remember each spX needs a separate UserAgentPort. For example, you might use 5070 for sp1 and 5071 for sp2. Now if using 5071, then this should work:
On CSipSimple use a DDNS with the filter rules (using@ddnsaddress:5071).
You may need to port forward 5071 in your router.
You are almost there! Let us know how you get on.
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