Single-Stage Dialing Through Any OBi Trunk Using SIP
RonR:
Quote from: Diana on March 09, 2012, 03:36:01 am
SIP Line: SP2 - setup with anveo; no DID
SIP Port: 5060 (set for 5060 even though Anveo uses 5010)
With Anveo on SP2/ITSPB, all SIP ports should be at Default settings with the exception of ProxyServerPort, which should be set to 5010.
Quote from: Diana on March 09, 2012, 03:36:01 am
SP1: GV
SP1 Port: 5064
With Google Voice on SP1/ITSPA, all SIP ports should be at Default settings since they are not used.
Quote from: Diana on March 09, 2012, 03:36:01 am
Questions:
Quote
User Settings -> User Defined DigitMaps -> User Defined Digit MapX
Label : p2p
DigitMap : (<n*:>(@@.)<:@192.168.1.140:sip_port>|<n*:>(@@.)<:@hostname:sip_port>)
(1) If I'm only using one OBi (OBi No: 200OBiNo), should I set n = 1?
(2) I'm not sure what info should be substitute here: "@hostname:sip_port>"; should it be @xxxxxx.dyndns.org:5064?
DigitMap : (<1*:>(@@.)<:@192.168.1.211:5060>|<1*:>(@@.)<:@hostname:5060>)?
This DigitMap is used to route calls to other OBi's. Since you don't have another OBi for this OBi to make calls through, this DigitMap can be left empty : ()
Quote from: Diana on March 09, 2012, 03:36:01 am
(3) Can I still use the OBiON App and a SIP Client on the three Android Devices?
Yes.
Quote from: Diana on March 09, 2012, 03:36:01 am
The other changes that were made are:
Quote
User Settings -> User Defined DigitMaps -> User Defined Digit Map4
Label : tsc
DigitMap : (200OBiNo|290xxxxxA|290xxxxxB|290xxxxxC|123456|123789)
This is a list of trusted OBi and SIP client userid's that are allowed to use single-stage dialing on this OBi. If no other OBi's or SIP clients are allowed to use single-stage dialing on this OBi, the DigitMap may be left empty: ()
Only SIP client userid's go in this list. OBiTALK numbers are not used anywhere in this SIP oriented configuration.
Momo:
Hi,
Thanks for your tutorial. My obi110 is already configured with the old method.
http://www.obitalk.com/forum/index.php?topic=1103.0
Now I would like to test this method. But I have 2 questions:
1) I don't understand how I should configure my sip client if I am calling from outside my lan
My OBI's lan address is 192.168.0.47. My wan IP address is like 82.2225.xxx.xx (I've just created a no-ip account.)
2) My OBITALK service inbounnd call route is acutally configured like this:
{(Mcot)>(0[6-8]xxxxxxxx|0033[6-8]xxxxxxxx|0091[2-6]xx.):li},{(Mcot)>(0[1-5,9]xxxxxxxx|0033[1-5,9]xxxxxxxx):tg3},{(Mcot)>(0091[7-9]xxxxxxxxxS0|0097xx.):vg3},{(Mcot)>(Mli):li},{(Mcot)>(<*1:>(Msp1)),(Mcot)>(<**1:>(Msp1)):sp1},{(Mcot)>(<*2:>(Msp2)),(Mcot)>(<**2:>(Msp2)):sp2},{(Mcot)>(<*8:>(Mli)),(Mcot)>(<**8:>(Mli)):li},{(Mcot)>(<*9:>(Mpp)),(Mcot)>(<**9:>(Mpp)):pp},{(Mcot):aa},{ph}
Can I configure it like this with the new method ?
Voice Services -> SPx Service -> X_InboundCallRoute (SPx must be configured for SIP) (OBi100/OBi110):
{(Mtsc)>(0[6-8]xxxxxxxx|0033[6-8]xxxxxxxx|0091[2-6]xx.):li},{(Mtsc)>(0[1-5,9]xxxxxxxx|0033[1-5,9]xxxxxxxx):tg3},{(Mtsc)>(0091[7-9]xxxxxxxxxS0|0097xx.):vg3},{(Mtsc)>(<*1:>(Msp1)),(Mtsc)>(<**1:>(Msp1)):sp1},{(Mtsc)>(<*2:>(Msp2)),(Mtsc)>(<**2:>(Msp2)):sp2},{(Mtsc)>(<*8:>(Mli)),(Mtsc)>(<**8:>(Mli)):li},{(Mtsc)>(<*9:>(Mpp)),(Mtsc)>(<**9:>(Mpp)):pp},{(Mtsc)>**0:aa},{(Mtsc)>***:aa2},{(Mtsc)>(Mp2p):spx},{(Mtsc)>(Mpli):pli},{(Mtsc)>0:ph},{(Mtsc):},{ph}
Thanks for your help.
Momo:
Hi,
I've tried to follow your method. But I don't know how to configure the sip client.
I have Csipsimple (Android).
My proxy is : myusername@no-ip.org:5061 (I am using SP2)
My username : SP2's username
My password : SP2's password.
I don't know how to configure the SIP to make calls without SIP registration.
Quote from: RonR on February 01, 2012, 05:54:15 pm
SIP clients should be configured to make calls without SIP registration. The SIP proxy should be the IP address or hostname of the OBi (with :5061 appended for SP2). If SIP clients are located outside the OBi's LAN, ports 5060 and 5061 will need to be forwarded to the OBi in the router.
Do you have any ideas ?
Thanks for your help.
Stewart:
Quote from: Momo on May 31, 2012, 01:54:52 am
I don't know how to configure the SIP to make calls without SIP registration.
See http://code.google.com/p/csipsimple/issues/detail?id=981
However, even though my CSipSimple can successfully make calls without registration via a provider, there seems to be a bug that prevents such a connection on the LAN. If myusername@no-ip.org resolves to a remote host, you should be ok.
Momo:
Thanks Stewart for your reply. I just want one clarification.
Is these informations are correct ?
Csipsimple username : SP2's username
Csipsimple password : SP2's password
Thanks for your help
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