Using CSipSimple With OBi
hwittenb:
I tried a new Acrobits test today using my OBi202. I called an unused SP1 setup as RonR outlined with the proxy at 127.0.0.1 with an incoming call routing to bridge the call to SP2 setup as voip. The bridging worked very well. The transmission of dtmf after the call completed worked fine better than the OBi110 test. My guess is the OBi202 has more horsepower than the OBi110.
I also tried calling SP2 and then bridging the call out on SP2. The call connects and bridges but the RTP sound packets don't start up. Similiar to what happens with the OBi110.
ianobi:
@ hwittenb.
I can reproduce your findings on my OBi110. If I use a “registered” account on sp2, then all direct calls through trunks and to the OBi phone work fine, but calls bridged through the auto attendant drop out within ten seconds. Unchecking X_RegisterEnable fixes the problem, but then of course you can only make outgoing calls on that service. Using the RonR method seems the best answer:
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)
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.
It is possible to set up an account in CSipSimple to call without registration. I have done this and made calls both ways with no third parties involved, but only using wifi. I set up a dynamic dns provider on my android phone, but the app tells me I am behind a proxy so it will not work. As things stand we have:
Android phone <> sip2sip <> OBi
The only problems with this setup seems to be that sip2sip does not pass Caller ID and through calls from android phone to OBi need to go out from OBi on a different trunk to the one they come in on. I don't see any dtmf problems.
Plenty to think about ::)
QBZappy:
Ian,
sip to sip calls should pass CID. That is one of the features of sip.
ianobi:
QBZ,
I agree! Problem is the actual provider sip2sip really does not like us using ui=$1 to pass userid on. I would be happy to be wrong :)
QBZappy:
Ian,
Quote from: ianobi on December 03, 2012, 08:29:26 am
using ui=$1 to pass userid on.
This code is unique to OBi products. It did not work very well when I tested it with Freephoneline. It showed CID only after 7 rings, making it useless with this SP. I think the CID should be delivered using a more conventional sip method, perhaps using sip uri. I believe that you have tested that without success. Just a stab in the dark, I remember that "X_UseRefer" setting carries the CID in the header in particular use cases. I'm not certain if this is relevant in your case.
Navigation
[0] Message Index
[#] Next page
[*] Previous page