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....

[MichiganTelephone]

obi-support2, thanks for clarifying that.

[murzik]

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

[QBZappy]

oleg

Hello and welcome. Just to confirm. Are you the oleg "firmware developer"?

I think the OBi has been noticed in Russia.

[oleg]

Without the GV, SP1, SP1, just using OBiTalk, are you able to do so?

YH

I did not verify but I am pretty sure sp1 as GV is irrelevant. For SIP URI calling it's important to have at least one of services configured for SIP and refer that trunk in Speed dial and / or OutboundCallRoute. And it does not have to register with any provider (uncheck X_RegisterEnable if don't need to register).

Just to confirm. Are you the oleg "firmware developer"?
I think the OBi has been noticed in Russia.

I have diploma in electronic engineering and experience in firmware development. Currently – I am software engineer (Linux, server side). But I am in US

[jerome]

Hello, I  have a version 1.3, I don't see where the SIP speed dial need to be done. can some post the step by step to perform this.

thank you
Jerome

[RonR]

Hello, I  have a version 1.3, I don't see where the SIP speed dial need to be done. can some post the step by step to perform this.

To make a SIP URI call from a Speed Dial, simply use:

User Settings -> Speed Dials -> n : SPx(18005551212@sip.tollfreegateway.com)

or

User Settings -> Speed Dials -> n : SPx(18005551212@76.10.223.207)

or

User Settings -> Speed Dials -> n : **?18005551212@sip.tollfreegateway.com

or

User Settings -> Speed Dials -> n : **?18005551212@76.10.223.207

where:

n is the Speed Dial number
SPx is SP1 or SP2, which must be configured for SIP.
**? is **1 or **2, which must be configured for SIP.

[jerome]

Thank you, I must missed something, could you post screen shot?

Thank you
Jerome

[RonR]

Thank you, I must missed something, could you post screen shot?

Scrren shot attqached.

[jerome]

ok thank you for the screen shot, that didn't work, obi says the number dialed SP2 .... is not a valid number.

anything else should be configured?

[RonR]

Is SP2 configured for SIP?

The SPx you use for SIP URI calls must be configured for SIP.

[jerome]

that may be the reason, how do you enable SIP on the SP2 ?

thank you
Jerome

[RonR]

that may be the reason, how do you enable SIP on the SP2 ?

If SP2 is not currently in use, this will suffice:

Service Providers -> ITSP Profile B -> SIP -> ProxyServer : 127.0.0.1
Voice Services -> SP2 Service -> AuthUserName : (put anything here)
Voice Services -> SP2 Service -> X_RegisterEnable : (unchecked)
Voice Services -> SP2 Service -> X_ServProvProfile : B