Making VoIP calls via OBiTalk using SIP URI method
yhfung:
I have already tried several attempts for making calls usng SIP URIs via the OBiTalk and failed to do so.
The format SIP may be in the form of
i) <ip_address>:<port_number>
ii) <hostname>:<port_number>
If port_number = 5060, it can be omitted.
This can be easily done by Linksys ATAs via iptel.org or Asterisk.
Please make OBi110 to be able to make calls via the OBiTalk using SIP URI method.
YH
MichiganTelephone:
+1
While I don't need this feature today, I have made use of it in the past with Sipura devices (using a SIP URI in a speed dial) so I can understand why folks would want it.
obi-support2:
yhfung,
IPv4 style dialing from the PHONE and calling SIP URL (via speed dial, or digit map) is one of the features added in the upcoming 1.2 release for the OBi110. Please look for it in 1-2 weeks.
If you really want to try out right now, you can do this (assume SP1 is using SIP Protocol, not GV):
- Setup a speed dial number as, say SP1(1234@myitsp.com:5061) or SP1(123@192.168.15.123), etc.
- Add the same URL in the SP1 DigitMap, like this ('1234@myitsp.com:5061'|'123@192.168.15.123'|....)
- Dial the speed dial number
- Note the use of ' ' to enclose the entire URL to avoid being interpreted as
special digit map syntax.
In other words, calling URL is kind of "supported" already in ver 1.1, except that the
DigitMap and CallRoute does not allow the characters to go through; hence the call fails.
--------
What is added in ver 1.2 is a simple syntax to allow all alphanumeric strings in digit map, so you
don't have to add them one by one in the digit map. And by default this new rule is added
to the SP1/SP2 digit map: [^*]@@.
which means to allow any length alphanumeric string (except # key) that does not start with a *
digit.
MichiganTelephone:
I have the beta firmware so decided to give this a test. I put the following in Speed Dial 8:
SP2(904@mouselike.org:5060) (tried it with and without the port number)
When I call 8# it rings once but then I get no audio. My call history does show that the call connected, however:
Terminal ID PHONE1 SP2
Peer Name
Peer Number SP2(904@mouselike.org:5060) 904@mouselike.org:5060
Direction Outbound Outbound
14:32:52 New Call
14:32:53 Call Connected
14:33:03 End Call
I can call this same address if I go through our Asterisk/FreePBX server and it works. I tried a few other sip addresses with mixed results (200901@login.zipdx.com, the VoIP user group conference bridge, rejected the call outright, possibly because it didn't see valid caller ID?). One that did work was this:
SP2(4153767253@podlinez.net)
Unfortunately this is just one way audio (CNN News) so I don't know if my audio was bidirectional or not. I don't understand why the first one (an echo/SIP test in the UK) wouldn't work, though.
obi-support2:
Ad hoc URL dialing, versus using a full blown SP1/SP2 trunk, has some limitations.
1. All NAT traversal parameters on the underlying SP trunk are suppressed;
i.e., no stun, no ice, and uses only local IP addresses.
2. It does not register with the server implied in the URL (obviously); so if the URL is a
real service, they may not accept the INVITE for that
3. The FROM userid/domain still follows the SP settings; again the server
might not accept the INVITE because of this.
4. If the called URL requires a different userid/password than the one on the current SP
for authentication, the call will fail
In my opinion, this ad hoc URL dialing is only useful for calling other devices in the same network, or asterisk extensions, etc. Unless the server is forgiving w/ respect to 1-4 above....
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