Getting SIP to work on Obi 202
ianobi:
You are correct. 5061 is the UserAgentPort for sp2 so the call is directed to sp2.
It might be interesting for you to direct the call to a spare sp maybe sp3 or sp4, after setting it up for sip as sugested by RonR. I presume default UserAgentPorts for an OBi202 range from 5060 to 5063. You might have to look again at port forwarding.
You can set UserAgentPort to other values, I use 5070 and 5071 in my OBi110 to try to avoid sip scanners.
rsriram22:
keep rocking @qbzappy.. your solution worked for me.. i had same issue as @hortoristic (sip forwarding from CC to my obi, call comes through,including caller id and then no voice after that from both directions)
..i fwded both ITSP profiles port sets in my router (just in case)
Quote from: QBZappy on September 28, 2012, 12:46:48 pm
On the SPx account, you may need to forward these ports as well.
ITSP Profile X
General
SIP
RTP <------------ Port Forward this range. Note that these should be different for every SPx account.
EVRMINC:
Quote from: QBZappy on September 28, 2012, 03:14:51 pm
Quote from: Hortoristic on September 28, 2012, 02:45:24 pm
Port forwarding as: 5060-5062 works but 5060-5062, 17000-17098 won't save on router.
SIP
These could be set up as separate rules
UDP 5060-5062
UDP 17000-17098
SIP setting on the router might be your sip alg which might need to be disabled. Uncheck it. It might be enough to fix everything.
QBzappy:
I am an OBiPLUS Beta user and use two OBi202s for two separate busineses. All is basically ok except I am trying to figure out if I can change the port assignments on the OBi202 to use ports other than 5060-5062 and 17000-17098. Why? I have a Fonality phone on the same LAN that uses 5060-5061, 10000-20000.
Any insight is appreciated, thanks.
QBZappy:
I think this is what you are trying to control:
Voice Services
SP1 Service->X_UserAgentPort = Whatever
ITSP Profile X
General
SIP
RTP->LocalPortMin = Whatever
->LocalPortMax = Whatever
lk96:
I recently got a DID from Voxbeam and thought it would be straightforward.
The symptoms I experience are identical to the original posting in this thread.
What I have so far:
1. UDP port forward is configured in the Dlink DIR-655 router I'm using for the appropriate range of ports.
2. SIP ALG is disabled on the router
3. I use SP4 for this service as such I forward incoming SIP traffic to port 5063.
Appropriate rules is in the Dlink router.
As such all suggestions posted in this thread are taken care of.
my URI looks like: 1234567890@name.dyndns.org:5063
Incoming calls ring the phone attached to the 202 at the correct service (SP4). But once I answer no audio
is there in either direction.
Checking the call status in progress I see one strange thing: the Rx audio codec is empty.
The Tx audio codec is G711u. The Rx stats keep incrementing. But the Tx stats are 0.
Anybody has any ideas/thoughts of what may be the problem ?
thanks
L.
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