Making VoIP calls via OBiTalk using SIP URI method
MichiganTelephone:
obi-support2, thanks for clarifying that.
murzik:
Quote from: obi-support2 on March 18, 2011, 02:21:35 pm
Ad hoc URL dialing, versus using a full blown SP1/SP2 trunk, has some limitations.
4. If the called URL requires a different userid/password than the one on the current SP
for authentication, the call will fail
I believe that authentication is fixed in the latest beta. So if you dial through gateway than user id and gateway password is in play.
I am afraid that NAT traversal for gateways or SIP URI dialing may become a problem.
I am experiencing one way audio using gateway to voxalot , so far I am unable to overcome that issue.
obi-support2:
What I was referring to here is totally ad hoc URL calling, not even using a pre-configured gateway (VG).
Yes, a pre-configured gateway in VG1-8, is somewhat more sophisticated than ad hoc url dialing, in the sense that it does not have the problems described in items 3 and 4 in my last post.
Thank you for mentioning this :)
oleg:
I also was playing with SIP URI dialing and succeeded to achieve my goals. Here is how...
I have current release (SoftwareVersion 1.1.0 (Build: 1892)). SP1 configured for GV, SP2 for SIP.
My first objective – call SIP URI via speed dials. I configured speed dials as following:
sp2(1234@sip.providerA.com)
sp2(56789@sip.providerB.com)
sp2(peter@ata.myfriend.com:5061)
sp2(john@somebody.dyndns.com:5062)
Note that ISTP B and SP2 have to be configured for SIP.
First two examples - SIP providers different to one configured in SP2 Service.
Second two examples - PAP2T ATAs. All peers do not require user/password. I do not know (yet) how to supply credentials.
I did NOT add the same URL in the DigitMap (as obi-support2 suggested in the post http://www.obitalk.com/forum/index.php?topic=381.msg2259#msg2259 above). I did NOT use quotes.
This works with current Obi110 software.
Further, it DOES use STUN server configured for sp2. I’ve learned it the hard way – at some point the STUN server was not available and I spent hours trying to explain 40 seconds delay between dialing and connecting – finally I run tcpdump on router and observed STUN requests with no response. I switched to another STUN server – fixed the problem and tcpdump shown STUN request/response.
Here are some suggestions to Obi development team:
- STUN requests and failures might be traced into syslog (I had it running on level 7)
- ability to configure several STUN servers might be useful (if one STUN server does not respond – ask another).
My second objective – use alternative SIP provider (beside one configured in sp2) to dial arbitrary number from handset.
I have configured Phone port as following:
- added element to DigitMap "**3(Msp2)" with no quote marks
- added to OutboundCallRoute "{(<**3:>(Msp2)<:@sip.alternativeprovider.com>):sp2}"
Now I have primary line sp1 (with no prefix), should dial prefix **2 to use sp2 (SIP provider) and dial prefix **3 to use alternative SIP provider (specified in OutboundCallRoute).
Once again:
- alternative SIP in my case does not require user/password
- I did not add URI to DigitMaps
- I did not use quotes (')
- OBi does use STUN (that’s really good)
- That works in current release (build 1892)
- I hope it will not stop working in upcoming release 1.2 obi-support2 mentioned above.
Regards
---oleg
yhfung:
Without the GV, SP1, SP1, just using OBiTalk, are you able to do so?
YH
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