OBiTALK Community

General Support => Installation and Set-Up (Devices) => Topic started by: RonR on March 29, 2011, 01:01:35 pm



Title: Using OBi Voice Gateways with SIP Providers
Post by: RonR on March 29, 2011, 01:01:35 pm
Ever wish the OBi supported using additional SIP providers for more outbound calling options?  Ever wish you could call people on other VoIP networks directly from your OBi using Sip Broker?  Ever wish you could call people directly from your OBi using their iNum number?  Well, you can, thanks to the Voice Gateways present in the OBi.  In this example, I'll show you how to add two additional SIP providers, calling via Sip Broker, and iNum calling.

NOTE:  You must have at least one OBi Voice Service (SPx/ITSPx) configured for SIP.  If you don't wish to configure a SIP provider on an SPx/ITSPx, simply set Service Providers -> ITSPx -> SIP -> ProxyServer to 127.0.0.1 and uncheck Voice Services -> SPx -> X_RegisterEnable.  Also, the SIP providers used in Voice Gateways must allow calling without SIP registration (many do, some don't).  Sip Broker and iNum calling do not use SIP registration.

The additional calling capability is added through the use of **3, **4, **6, and **7 dialing prefixes.  [**5 cannot be used as a dialing prefix because it's hard-coded into the OBi for use by Obihai.]  In this example, **3 and **4 will be used for additional SIP providers, **6 will be used for calling via Sp Broker, and **7 will be used for iNum calling.

To begin, you'll need to make a couple of additions to your PHONE Port DigitMap and PHONE Port OutboundCallRoute to add support for the new dialing prefixes:


Phone Port DigitMap:

|**1(Msp1)|**2(Msp2)|**3(Mvg3)|**4(Mvg4)|**6(Mvg6)|**7(Mvg7)|**8(Mli)|**9(Mpp)|

PHONE Port OutboundCallRoute:

{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{(<**3:>(Mvg3)):vg3},{(<**4:>(Mvg4)):vg4},
{(<**6:>(Mvg6)):vg6},{(<**7:>(Mvg7)):vg7},
{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}


This creates the following associations:

**3 -> Voice Gateway3 (VG3)
**4 -> Voice Gateway4 (VG4)
**6 -> Voice Gateway6 (VG6)
**7 -> Voice Gateway7 (VG7)


Now, let's set up additional VoIP providers on Voice Gateway3 and Voice Gateway4.

Voice Gateway3 will be used with Whistlephone:

Name : Whistlephone
AccessNumber : SPx(proxy.whistlephone.com)
DigitMap : (Mste)
AuthUserID : your_whistlephone_user_id
AuthPassword : your_whistlephone_password


Voice Gateway4 will be used with IdeaSIP:

Name : IdeaSIP
AccessNumber : SPx(proxy.ideasip.com)
DigitMap : (Mste)
AuthUserID : your_ideasip_user_id
AuthPassword : your_ideasip_password


Next, let's configure Voice Gateway6 for calling via Sip Broker:

Name : Sip Broker
AccessNumber : SPx(sipbroker.com)
DigitMap : (<*>[x*][x*].|*[x*][x*].)


Next, let's configure Voice Gateway7 for iNum calling:

Name : iNum
AccessNumber : SPx(sip.inum.net)
DigitMap : (<8835100>xxxxxxxx|8835100xxxxxxxx)


And finally, let's configure the User Defined DigitMap referenced in Voice Gateway3 and Voice Gateway4:

Label : ste
DigitMap : (1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|<1aaa>xxxxxxx|011xx.|(Mipd)|[^*]@@.'@'@@.)

where aaa is your local area code.


Following a reboot, the OBi should be ready to use its new capabilities.

Dialing **3 + number should place a PSTN call using Whistlephone.

Dialing **4 + number should place a PSTN call using IdeaSIP.

Dialing **6 + SIP code + number should place a VoIP call via Sip Broker.

For example:

Dialing **6 011 188888 should connect you with the Sip Broker test announcement.
Dialing **6 010 123456 should connect you with the Voxalot number 123456.
Dialing **6 747 17471234567 should connect you with the Gizmo5 number 1-747-123-4567.

For more details on Sip Broker, please visit : http://www.sipbroker.com/

Dialing **7 + number should place a VoIP call to an iNum number.

NOTE: Use only the last 8 digits of the iNum number (8835100 will be prepended for you).

For example:

Dialing **7 00000091 should connect you with the iNum echo test.
Dialing **7 04123456 should connect you with the Voxalot number 123456.
Dialing **7 71234567 should connect you with the Gizmo5 number 1-747-123-4567.

For more details on iNum, please visit : http://www.inum.net/


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: srhuston on March 29, 2011, 02:02:34 pm
I'm looking around their website, but maybe one of the gurus here knows; does Sipgate "allow calling without SIP registration"?  If so my plans for 911 capability with my OBi110 just got easier; two GV accounts setup as SP1/SP2 and Sipgate in one of the voice gateways with my E911 data registered with them (and the associated monthly fee).


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on March 29, 2011, 02:08:43 pm
two GV accounts setup as SP1/SP2 and Sipgate in one of the voice gateways with my E911 data registered with them (and the associated monthly fee).
You missed a major restriction the OBi developers placed on using Voice Gateways with SIP:

NOTE:  You must have at least one SIP provider already configured on the OBi using SPx/ITSPx.

Google Voice is not a SIP provider.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: oleg on March 29, 2011, 02:20:27 pm
Hi RonR,

Quote
**5 cannot be used as a dialing prefix because it's hard-coded into the OBi for use by Obihai.

Could you point out where is that said / documented?


BTW, I am glad your philosophy is turning toward mine  ;)

My general philosophy is to leave the PHONE Port DigitMap and OutboundCallRoute unmodified.  



Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on March 29, 2011, 02:45:52 pm
Quote
**5 cannot be used as a dialing prefix because it's hard-coded into the OBi for use by Obihai.

Could you point out where is that said / documented?
Dialing **5 + number unconditionally sends number to the OBiTALK server.  You can try it yourself (no changes are necessary).  I tried to override it in the DigitMap/OutboundCallRoute, but was unsuccessful.

BTW, I am glad your philosophy is turning toward mine  ;)

My general philosophy is to leave the PHONE Port DigitMap and OutboundCallRoute unmodified. 
I knew someone would call me out on that the instant I wrote it.   :)

I also said "it's generally a good idea to leave them both unmodified unless you fully understand the ramifications of changing them."  I do.   :)

I'm sure you understand that these 'modifications' are totally non-controversial and maintain the 1-to-1 philosophy of the default settings.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: yhfung on March 29, 2011, 03:45:08 pm
RonR,

Thank you for providing these useful examples such that we can know how to use the gateway for making outbound calls without registration.

Is it possible replace SPx by PP?

(I have tried it with PP and SP1(GV), it does not work. It works with SP2( my own Asterisk server), as a result, we may request OBihai to allow us to make Voice Gateway calls via the PP)

YH


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: srhuston on March 29, 2011, 03:53:23 pm
You missed a major restriction the OBi developers placed on using Voice Gateways with SIP:

NOTE:  You must have at least one SIP provider already configured on the OBi using SPx/ITSPx.

Google Voice is not a SIP provider.


I know GV isn't a SIP provider.  And I wouldn't say I missed it, more like I didn't read much more into it than thinking "Oh this would be nice" when I read your initial post and wanted to get a quick clarification before I left work to head home.

This just puts me back in the idea of a second device to connect to my sipgate account all the time and plug into the LINE port of my 110.  No big deal :>


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: oleg on March 29, 2011, 04:02:33 pm
RonR,
I think my approach is more consistent and better fit to non-trivial routing (like in your setup above).
But I would not proceed with "philosophy" discussion  :)

I tried "**52008xxxxx" (with real number of another OBi110) - voice response was "the number you dialed star-star-five-two-zero-zero-eight has been sent to the server". Just that, not entire number. The records in syslog show the same (action started right after **52008) and than following:
Code:
3/29/2011 6:39:28 PM <7> OBHSN_SIGNUP:ssl connected
3/29/2011 6:39:28 PM <7> BASESSL:get common name:root.pnn.obitalk.com
3/29/2011 6:39:29 PM <7> OBHSN_SIGNUP:buflen=10087
3/29/2011 6:39:29 PM <7> OBHSN_SIGNUP:buflen=5048
3/29/2011 6:39:29 PM <7> OBHSN_SIGNUP:Sending cfg : 5048
Can you explain this?
I guess it may be one of features implemented for the troubleshooting, but it would be good if OBi mentioned it in the admin guide.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on March 29, 2011, 04:28:43 pm
Can you explain this?
I guess it may be one of features implemented for the troubleshooting, but it would be good if OBi mentioned it in the admin guide.
Obihai uses this during device registration with the OBiTALK Web Portal.  I suspect they're capturing/validating/whatever the OBi's phone number/serial number/mac address/who knows for a bit of security with the Circle of Trust business.  I haven't spent any time trying to figure it out.  It appears to be hard-coded into the OBi, so I let it go as such.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: murzik on March 29, 2011, 04:48:44 pm
Can you actually place a call through sipbroker or voxalot gateways?


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: oleg on March 29, 2011, 04:51:27 pm
**N is not the only dialing prefix. I used to have #N (e.g. '#2' rather than '**2') - it worked with Linksys adapters and works with OBi. Phone Port DigitMap and OutboundCallRoute have to be changed accordingly (just replace '**' with '#').

Someone may find it more convenient...


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on March 29, 2011, 05:16:40 pm
**N is not the only dialing prefix. I used to have #N (e.g. '#2' rather than '**2') - it worked with Linksys adapters and works with OBi. Phone Port DigitMap and OutboundCallRoute have to be changed accordingly (just replace '**' with '#').

Someone may find it more convenient...

Of course.  That's one of the nice things about the PHONE Port DigitMap/OutboundCallRoute processing scheme - you can throw the current one's out the window and make up a totally new dialing syntax if you wish.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on March 29, 2011, 07:26:06 pm
Can you actually place a call through sipbroker or voxalot gateways?
If you mean a free call to a PSTN phone, the answer is no.  You can, however, place free calls to others whose service provider peers with Sip Broker.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: murzik on March 30, 2011, 06:22:27 am
Can you actually place a call through sipbroker or voxalot gateways?
If you mean a free call to a PSTN phone, the answer is no.  You can, however, place free calls to others whose service provider peers with Sip Broker.

What I meant if sipborker and voxalot are work for you, because what I get is just one way audio.
Especially calling any peers through sipbroker.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: oleg on March 30, 2011, 07:12:14 am
I believe the reason is that OBi110 build 2101 does NOT use STUN for outgoing SIP calls via alternative providers (only use STUN for the provider configured in corresponding SPx service). In other words OBi110 sends your local IP:port in RTP stream information.

For most of us OBi adapter is located on local network behind NAT. Some destinations are smart enough to recognize local IP sent by OBi, disregard it and try to send RTP just in opposite direction (to the same IP/port where your voice stream comes from). Often this works, sometimes not...

Build 1892 used STUN in the same situation. I've suggested developers to re-think STUN usage decision, waiting for response...

P.S. Please let me know if someone needs more details / explanations.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on March 30, 2011, 09:58:14 am
What I meant if sipborker and voxalot are work for you, because what I get is just one way audio.
Especially calling any peers through sipbroker.
I agree with oleg that it's a NAT/RTP issue.  I haven't had any problems so far, but I also forward all applicable SIP and RTP ports to the appropriate devices as well as use a STUN server whenever possible.  VoIP audio is a problem waiting to happen when NAT is involved with SIP/RTP.

A good test to see if your router is possibly the culprit is to simply bypass it, if possible.  Connect the OBi directly to your Cable/DSL modem and see if your audio issues go away.  If they do, your router is definitely a suspect.  Make sure you reboot both devices after connecting them directly to each other.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: JohnLennon on March 30, 2011, 01:46:13 pm
Great solution!


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: oleg on March 30, 2011, 07:25:09 pm
A good test to see if your router is possibly the culprit is to simply bypass it, if possible.  Connect the OBi directly to your Cable/DSL modem and see if your audio issues go away.  If they do, your router is definitely a suspect.

Bypassing router you eliminate NAT and thus the need in STUN detection. Audio issues may go away. But how it makes "router a suspect" ??? That only proves that router has NAT inside and that router separates local network from the internet. Anybody doubts?  ;)
Normally it is responsibility of SIP device to detect internet address / port and send them to another SIP device. STUN protocol serves that purpose. Routers usually do not care about SIP protocol.

There are a few rare exceptions though:
- some NAT implementations may be practically impossible to traverse even using STUN.
- some rare routers (VOIP routers, SIP proxies) may recognize SIP protocol and substitute local address / port with internet address / port.



Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on March 30, 2011, 08:05:04 pm
Audio issues may go away. But how it makes "router a suspect" ???
The very first paragraph of the article linked below does a much better job of describing the problem than I ever could:

"The very first thing to note is that SIP was NOT designed to work with NAT. There are subsequent standards, hacks, workaround, kludges etc. to try and make it work but the original SIP designers somehow deemed it beneath them or put it in the too hard basket to bother coming up with a proper solution (there is not one instance of the string “NAT” in the whole SIP RFC)."

http://sipsorcery.wordpress.com/2009/08/05/nat-rtp-and-audio-problems/

If taking your router of the loop eliminates your audio issues, there's an excellent chance your router has a NAT implementation that's not SIP/RTP friendly.  I've got a number of older routers in the store room that fit that description.  Newer routers tend to do a lot better job, but they're still not all created equal and depending on the service provider's SIP implementation, one router may work in a particular situation  but not in another.  IOW, it's a crap shoot.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: oleg on March 30, 2011, 08:31:47 pm
Indeed, SIP was not designed to work with NAT. But most contemporary SIP implementations (including all Linksys ATAs) use STUN and successfully traverse most NATs.

Try another experiment - replace OBi110 build 2101 with build 1892, make sure STUN enabled and configured. There is an excellent chance that audio issues go away with the same router (unless router belongs to first of two exceptions I mentioned above). According to your logic that proves OBi110 build 2101 "is definitely a suspect". Here I agree with you  :)


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: obi-support2 on March 31, 2011, 05:12:34 pm
It is correct that release 1.2.2101 removes the NAT support for SIP Gateway calls and URL calls. This includes using local IP address/port for Contact and RTP in the outbound INVITE, as well as not using STUN and ICE, regardless settings of SP1/SP2 service.

The logic for this decision is to make SIP Gateway Calls and URL calls less dependent on the behavior of the main service on SP1 and SP2. We expect most uses of SIP Gateway and URL Calls are in cases where:
1. Called URL is on the same subnet
2. Called URL, if it is an ITSP, would have infrastructure (like an outbound proxy) to
   handle NAT traversal.

We recognize that there are still valid cases where NAT support may be required. I have put in a request to see if an option can be added in a future release.

Meanwhile, port forwarding on your router would be a good work around if have access to it.
You will need to port forward the SIP Port on SP1/SP2, and also the entire RTP Port Range in the underlying ITSP Profile.





Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on March 31, 2011, 05:51:50 pm
We recognize that there are still valid cases where NAT support may be required. I have put in a request to see if an option can be added in a future release.
obi-support2,

Might I suggest you carve out your low-level SIP support routines such that they are generally callable from not only SP1/SP2 but also directly from the Voice Gateways, SIP URI's and IP Dialing, so they don't rely on having a SIP provider already configured on SP1/SP2.  It's a real shame you can't have Google Voice on both SP1 and SP2 and still use Voice Gateways, SIP URI's and IP Dialing for SIP calling, when the code is just sitting there.

I would also suggest you use a notation of SIP(URI) instead of SP1/SP2 for Voice Gateways, Speed Dials, etc. and that IP Dialing assume the same.  None of these should be tied to SP1/SP2 or PrimaryLine.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: oleg on March 31, 2011, 06:06:37 pm
Hi obi-support2,

Thank you for the response. Unfortunately port forwarding on the router does not help. I have dd-wrt Linux Kernel 2.4 based router which probably does the best for SIP, in particular it always assigns the same internet port number for outgoing packets as local port number (and of course I have SIP and RTP ports on router mapped to OBi device). Said that, two problems remain and I can confirm them (either running tcpdump or syslog on both sides):
- "Contact" header in "200 OK" was sent with local network IP (local port was equal to internet port), which caused ATA on other side to send undeliverable ACK (sent to my local IP). As result OBi was not able to establish connection.
- INVITE contains local IP for RTP data and ATA on other side may direct RTP stream to unreachable local IP. That resulted in one way audio. Many providers can recognize local IP and direct RTP stream "symmetrically" (than it works), but this is not the case for many ATA devices.

Please (please!!!!) consider employing STUN for outgoing calls. If you have doubts - add it conditionally, based on settings, that's fine. It will greatly improve connectivity and therefore usability of OBi device.

P.S. I second RonR's suggestion to separate SIP URI calling from SP1/SP2. Fully functional SIP URI calling independent from SP1/SP2 would be just great. The device would support up to 3 simultaneous registrations on SP1/SP2/OBiTALK and quite a big set of SIP URI destinations without registration. That's the dream device!

Regards,
---oleg


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: MichiganTelephone on April 09, 2011, 05:37:26 am
I realize this thread is a bit dated but I just wanted to add three points:

First, I also experienced one-way audio on inum calls, but not on Sipbroker calls.  I have no idea why it should be different between the two, but it is.  I didn't make any effort to resolve that because I never use inum anyway.  My firmware is 1.2.1 (Build: 2103).

Second, on Sipbroker calls, it was sending "unknown" for the Caller ID.  I found that if you put a regular phone number (I do suggest including the country code, but I think it will take any number) in the AuthUserID field of the Voice Gateway6 settings, it will send that as the Caller ID number, or at least that worked on a couple of test calls that I tried.

Third, when using Sipbroker I could not use the # key to indicate the end of the number because it appears the device considered the # part of the number dialed.  But if I add timeout values (such as S4) to the DigitMap then the # key works as it should:

(<*>[x*][x*].S4|*[x*][x*].S4)


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: N7AS on April 09, 2011, 03:38:14 pm
RonR,

In your origional post, you can use Voice Gateways with SIP providers...
Can this be done if I have 2 GV accounts sp1 and sp2? sp1 is primary. I have quite a few GV accounts that I would like to add to the OBi110 if it's possible.



Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on April 09, 2011, 04:31:36 pm
Quote
NOTE:  You must have at least one SIP provider already configured on the OBi using SPx/ITSPx.

It's a real shame, seeing as how all the code to handle SIP providers is sitting there, but you can only use SIP providers on Voice Gateways (and with considerable limitations) through an existing SP1/SP2 Service that's configured with a VoIP provider.

Google Voice accounts cannot be added though Voice Gateways.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: GizmoChicken on April 25, 2011, 05:50:15 am
You missed a major restriction the OBi developers placed on using Voice Gateways with SIP:
NOTE:  You must have at least one SIP provider already configured on the OBi using SPx/ITSPx.

Any thoughts on WHY the OBi developers placed this restriction on using Voice Gateways? 

As I've posted elsewhere:
I don't understand the need for the "SP1" and "SP2" preceding the target service.  As posted by another, this seems to imply that either "SP1" or "SP2" must be a SIP service.  That is, gateways won't work if BOTH "SP1" or "SP2" are attached to Google Voice accounts.  I can't understand why this limitation has been imposed. 

I suppose I could sort of understand why a voice gateway may require setting a "SIP signalling protocol" and why it might be easier to do that by referring to "ITSP Profile A" or "ITSP Profile B" rather that specifying the protocol (and/or other parameters) for each voice gateway.  But IF that's the case, why not adjust your code so that something like the following configurations could be used: ITSP-A(sip.xyz.com:5081) or ITSP-B(192.168.15.118).

Am I way off base?


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on April 25, 2011, 08:15:46 am
You missed a major restriction the OBi developers placed on using Voice Gateways with SIP:
NOTE:  You must have at least one SIP provider already configured on the OBi using SPx/ITSPx.

Any thoughts on WHY the OBi developers placed this restriction on using Voice Gateways?

Using the OBi Voice Gateways with SIP providers was an after-thought.  The SIP support code in the OBi was apparently not written in a manner for general use, and rather than reorganizing it such that it could be, SIP support through Voice Gateways was quickly shoe-horned in as a piggy-back operation on SP1/SP2, with a lot of limitations as a result.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: JamesC on April 25, 2011, 11:51:52 am
Hi All,

Quick question here. If a voice gateway uses the sp1/sp2 interfaces to call out, would the sp1/sp2 affect the quality?

I have the below settings on my Obi110
sp1 : Google Voice (for free in/out US calls)
sp2 : sipgate (For free backup incoming calls)
vg4 : sp2(callcentric.com)

I've always heard about great quality of Callcentric, but would the fact that it is run through sipgate be a quality factor? Not saying sipgate does not have great quality, just wondering about the relation.

Thanks


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on April 25, 2011, 12:20:47 pm
Hi All,

Quick question here. If a voice gateway uses the sp1/sp2 interfaces to call out, would the sp1/sp2 affect the quality?

I have the below settings on my Obi110
sp1 : Google Voice (for free in/out US calls)
sp2 : sipgate (For free backup incoming calls)
vg4 : sp2(callcentric.com)

I've always heard about great quality of Callcentric, but would the fact that it is run through sipgate be a quality factor? Not saying sipgate does not have great quality, just wondering about the relation.

Thanks


It's not actually being run through Sipgate.  It's being piggy-backed on the support for SIP that's configured on SP2, but the actual communications is with CallCentric and not Sipgate.

There are some considerations that might affect CallCentric operation:

Note that when using a SP trunk to access a (SIP) gateway, the device will:
- Not use the outbound proxy, ICE, or STUN regardless the settings on the SP trunk.
- Use only the device’s local address as the SIP Contact, and ignore any natted address discovered by the device.
- Use the gateway’s SIP URL to form the FROM header of the outbound INVITE.
- Use the gateway’s AuthUserID and AuthPassword for authentication.
- Apply the symmetric RTP concept.

If I were you, I'd temporarily configure SP2 for CallCentric instead of Sipgate and find out what, if any, trade-offs there are in using the Voice Gateway instead.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: GizmoChicken on April 25, 2011, 01:13:08 pm
Using the OBi Voice Gateways with SIP providers was an after-thought.  The SIP support code in the OBi was apparently not written in a manner for general use, and rather than reorganizing it such that it could be, SIP support through Voice Gateways was quickly shoe-horned in as a piggy-back operation on SP1/SP2, with a lot of limitations as a result.

Thanks RonR.  I hope OBi will eventually reorganize the code to remove these limitations.  Or better still, I hope OBi will just allow for more SP accounts.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: JamesC on April 26, 2011, 10:25:23 am
Thank you RonR!

Using Callcentric through vg4 has provided superb quality these pass few days. I can't tell the difference in quality configuring it as SP1 or VG4. Now I could start filling up the other voice gateway slots!
The weird thing is that while localphone and callcentric works well as a VG, I never got voxalot or iNum to work. The Rx stream is always 0. Would it be something to do with disabling G711 codecs?

Thanks

James


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on April 26, 2011, 11:01:15 am
I never got voxalot or iNum to work. The Rx stream is always 0. Would it be something to do with disabling G711 codecs?
I don't have any problem with either Voxalot or iNum and G711U.  Voxalot uses a blend of Asterisk and OpenSER (I think that's the other half) and has always been a bit prone to RTP issues for many.  I'm glad to hear LocalPhone and CallCentric do well on Voice Gateways.


Title: Re: Using OBi Voice Gateways with SIP Providers (with GV as both SP1 and SP2)
Post by: GizmoChicken on April 28, 2011, 07:43:47 am
Using the OBi Voice Gateways with SIP providers was an after-thought.  The SIP support code in the OBi was apparently not written in a manner for general use, and rather than reorganizing it such that it could be, SIP support through Voice Gateways was quickly shoe-horned in as a piggy-back operation on SP1/SP2, with a lot of limitations as a result.

Thanks RonR.  I hope OBi will eventually reorganize the code to remove these limitations.  Or better still, I hope OBi will just allow for more SP accounts.

Just to reiterate the situation, when SP1 and SP2 are BOTH configured to work with Google Voice (GV) accounts, we are NOT able to use the outgoing SIP "Voice Gateways" feature.

In the "Feature Requests" section, I requested OBi to eliminate this limitation so as to allow using the outgoing SIP "Voice Gateways" feature even when SP1 and SP2 are BOTH configured to work with Google Voice (GV) accounts.  If you support (or oppose) this request, please comment here:

http://www.obitalk.com/forum/index.php?topic=751.0 (http://www.obitalk.com/forum/index.php?topic=751.0)




Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: daibaan on May 09, 2011, 11:35:30 am
Anybody tried to configure onesuite VoIP service on voice gateway?

I was able to configure onesuite on SP1 (GV on SP2) and use it to make outgoing call.  I have also successfully configure sipgate, callentric, callwithus, whistlephone on VG and callout via SP1.  However, when I configure onesuite on a VG, outgoing call simply fails, Obi complains that server does not respond.

Anyone got an idea?


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on May 09, 2011, 11:44:59 am
Does OneSuite support calling without SIP registration?  If OneSuite requires SIP registration, you won't be able to use it on a Voice Gateway.  You will also need a SIP provider provisioned on either SP1 or SP2 in order to be able to use a SIP provider on a Voice Gateway.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: daibaan on May 09, 2011, 11:55:41 am
I am not sure if onesuite requires registration or not, but I have always enable registration on onesute connection, and that is how X_RegisterEnable in ITSP Profile A (used by SP1) is set.

However, you just reminded me that since I AM using onesuite as SP1 - with registration enable, that may have interfere  with the onesuite VG call using the same SP1 config? maybe I should change SP1 to sipgate  (which is actually the ultimate goal, because I want to use sipgate to provide a backup dial-in number) first and try the onesuite VG again. will get to it tonight

Thanks RonR


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: daibaan on May 09, 2011, 08:49:39 pm
Problem solved with 1.2.1 (Build: 2283)

First, I have verified onesuite.com does not require registration.
Second, one thing a little strange about onesuite.com is the username format, the username for user "sam" is in the form of "sam-voip.onesuite.com" with proxy "voip.onesuite.com", so the fully qualified name would be "sam-voip.onesuite.com@voip.onesuite.com", this cause some SIP software/broker to be confused.

However, without changing any config, once the new 2283 firmware is loaded, the onesuite.com VG works like a charm, that is good enough for me I am happy now


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: daibaan on May 09, 2011, 10:35:52 pm
Firmware Release 1.2.1(2283) has been posted. One enhancement includes is the
addition of options to support NAT traversal for SIP Gateway and URL calls
on SP1/2: You can now append the following URL parameters to each speed dial and
SIP Gateway VG1-4 access number, separated by ';',
   - ui=userid[:password]   //ui=user-info, password is optional
   - op=[m][n]          //option flags, i=ice, m=symmetric-rtp,
                                    //n=use-natted-address, s=stun

Thanks for the info, 2 questions regarding these enhancement:

Does the "ui" option overlaps the "AuthUserID/AuthPassword" options for VG? or vise versa?
Why is this options limited to VG1-4? can these options not be used for VG5-8?


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: obi-support2 on May 10, 2011, 08:40:25 am
Firmware Release 1.2.1(2283) has been posted. One enhancement includes is the
addition of options to support NAT traversal for SIP Gateway and URL calls
on SP1/2: You can now append the following URL parameters to each speed dial and
SIP Gateway VG1-8 access number, separated by ';',
   - ui=userid[:password]  
         user-info, password is optional
   - op=[ i ][ m ][ n ][ s ]
         option flags, i=ice, m=symmetric-rtp,
         n=use-natted-address, s=stun

 Examples:
   SpeedDial = sp2(1234@sip.inum.net;ui=1000:xyz;op=sm)
   VG1-8 AccessNumber = SP1(sip.inum.net;user=1000;op=imns)
   Note that if userid or password is specified in VG1-8 AccessNumber,
   it overwrites the settings in    AuthUserID, and AuthPassword in the VG.
   If stun disabled in the main SP, then stun client is not enabled,
   and may lead to crash when doing URL dialing and stun specified.



Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: JonG on May 29, 2011, 03:24:36 am
Thank you Obi  for adding this feature. 
The Obi now does more than ATAs costing five times as much, and so I have ordered my first OBi box. 
Lots of frustrated Linksys users will soon join too.

Obihai must be a great place to work.
You included GoogleVoice when customers asked, and now you put this user request into a firmware upgrade in just a month!

Now please update the Admin Manual.






Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: Riyas on June 26, 2011, 11:52:04 am
Hi !!! First of all, thanks for this wonderful tip. I've configured it... It's working great... Thanks a lot

My settings:

SP1 => Pbxes.org
SP2 => Google talk
VG3 => Free (French internet provider)

Everything is working fine now. But my internet provider announced that he will restrict the sip account only from my IP address (I have a static IP address). I know that if I use my "free" account with SP1 or SP2, it will be my IP address. But in VG3, will it go through Pbxes.org IP address ? In that case, it won't work for me. I don't know if it's clear ?

Thanks


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on June 26, 2011, 12:36:43 pm
SP1 will be communicating directly with PBXes using SIP.

and

VG3 will be communicating directly with whoever is configured at Voice Gateway3 -> AccessNumber using SIP.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: neilio on August 03, 2011, 07:46:36 pm
Not sure what I'm doing wrong, but when I add the two new strings in the Phone Port section

DigitMap
|**1(Msp1)|**2(Msp2)|**3(Mvg3)|**4(Mvg4)|**6(Mvg6)|**7(Mvg7)|**8(Mli)|**9(Mpp)|

and

OutboundCallRoute
{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{(<**3:>(Mvg3)):vg3},{(<**4:>(Mvg4)):vg4},{(<**6:>(Mvg6)):vg6},{(<**7:>(Mvg7)):vg7},{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}

I get red exclamation points beside both of those fields, and dialing **# stops working.

Have I missed something obvious?


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on August 03, 2011, 07:55:01 pm
Phone Port DigitMap:

|**1(Msp1)|**2(Msp2)|**3(Mvg3)|**4(Mvg4)|**6(Mvg6)|**7(Mvg7)|**8(Mli)|**9(Mpp)|

PHONE Port OutboundCallRoute:

{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{(<**3:>(Mvg3)):vg3},{(<**4:>(Mvg4)):vg4},
{(<**6:>(Mvg6)):vg6},{(<**7:>(Mvg7)):vg7},
{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}

The bolded rules are additions to what's already there (you sandwich them in).


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: neilio on August 03, 2011, 08:08:07 pm
Phone Port DigitMap:

|**1(Msp1)|**2(Msp2)|**3(Mvg3)|**4(Mvg4)|**6(Mvg6)|**7(Mvg7)|**8(Mli)|**9(Mpp)|

PHONE Port OutboundCallRoute:

{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{(<**3:>(Mvg3)):vg3},{(<**4:>(Mvg4)):vg4},
{(<**6:>(Mvg6)):vg6},{(<**7:>(Mvg7)):vg7},
{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}

The bolded rules are additions to what's already there (you sandwich them in).


That's what I have currently in those textfields - I simply copied and pasted right from the original post, making sure there were no phantom linebreaks. Or am I missing the point?


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on August 03, 2011, 08:16:41 pm
You start with the default values and blend this into them at the appropriate places:

|**3(Mvg3)|**4(Mvg4)|**6(Mvg6)|**7(Mvg7)|

and

{(<**3:>(Mvg3)):vg3},{(<**4:>(Mvg4)):vg4},{(<**6:>(Mvg6)):vg6},{(<**7:>(Mvg7)):vg7}

These are ADDITIONS to the default values, not replacements for the default values.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on August 03, 2011, 08:20:35 pm
The net results should be:

([1-9]x?*(Mpli)|[1-9]|[1-9][0-9]|911|**0|***|#|**1(Msp1)|**2(Msp2)|
**3(Mvg3)|**4(Mvg4)|**6(Mvg6)|**7(Mvg7)|**8(Mli)|**9(Mpp)|(Mpli))

and

{([1-9]x?*(Mpli)):pp},{(<#:>|911):li},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},
{(<**3:>(Mvg3)):vg3},{(<**4:>(Mvg4)):vg4},{(<**6:>(Mvg6)):vg6},{(<**7:>(Mvg7)):vg7},
{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: neilio on August 03, 2011, 08:31:12 pm
Thanks again, RonR. Dialing works again, but when I try to dial out on **3 now I get a "no service available" error.

Here's what I have configured:

http://neil.io/9292

And here's what I have in the User Defined DigitMaps:

http://neil.io/92o2

Strange that I get the red ! everywhere. I must have missed a setting.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on August 03, 2011, 08:36:19 pm
http://neil.io/9292

SPx is supposed to SP1 or SP2, which has to configured for SIP.

Strange that I get the red ! everywhere. I must have missed a setting.

I can't help you with that.  I don't use the OBiTALK Web Portal.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: Obvdobi on October 21, 2011, 03:42:35 pm

Next, let's configure Voice Gateway6 for calling via Sip Broker:

Name : Sip Broker
AccessNumber : SPx(sipbroker.com)
DigitMap : (<*>[x*][x*].|*[x*][x*].)


Next, let's configure Voice Gateway7 for iNum calling:

Name : iNum
AccessNumber : SPx(sip.inum.net)
DigitMap : (<8835100>xxxxxxxx|8835100xxxxxxxx)


Following a reboot, the OBi should be ready to use its new capabilities.


Dialing **6 + SIP code + number should place a VoIP call via Sip Broker.

For example:

Dialing **6 011 188888 should connect you with the Sip Broker test announcement.
Dialing **6 010 123456 should connect you with the Voxalot number 123456.
Dialing **6 747 17471234567 should connect you with the Gizmo5 number 1-747-123-4567.

For more details on Sip Broker, please visit : http://www.sipbroker.com/

Dialing **7 + number should place a VoIP call to an iNum number.

NOTE: Use only the last 8 digits of the iNum number (8835100 will be prepended for you).

For example:

Dialing **7 00000091 should connect you with the iNum echo test.
Dialing **7 04123456 should connect you with the Voxalot number 123456.
Dialing **7 71234567 should connect you with the Gizmo5 number 1-747-123-4567.

For more details on iNum, please visit : http://www.inum.net/


When I set it up for vg6 and vg7 with the instructions above, no test call can be completed for anything start with **6/**7.

What am i doing wrong? Setup in attached image.

 


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on October 21, 2011, 05:08:48 pm
When I set it up for vg6 and vg7 with the instructions above, no test call can be completed for anything start with **6/**7.

What am i doing wrong? Setup in attached image.

Do you have |**6(Mvg6)|**7(Mvg7)| in your PHONE Port DigitMap?

Do you have {(<**6:>(Mvg6)):vg6},{(<**7:>(Mvg7)):vg7} in your PHONE Port OutboundCallRoue?

Is SP2 configured for SIP and reporting 'Connected' if it's a real SIP provider or 'Registration not required' if it's a dummy SIP configuration?

What are you dialing and what is the response you're getting?

**6 011 188888 should reach the SIP Broker Test Announcement.

**7 0000 0091 should reach the Inum Echo Test.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: Obvdobi on October 21, 2011, 05:28:57 pm

Quote
Do you have |**6(Mvg6)|**7(Mvg7)| in your PHONE Port DigitMap?
yes

Quote
Do you have {(<**6:>(Mvg6)):vg6},{(<**7:>(Mvg7)):vg7} in your PHONE Port OutboundCallRoue?
yes
Quote
Is SP2 configured for SIP and reporting 'Connected' if it's a real SIP provider or 'Registration not required' if it's a dummy SIP configuration?
SP2 is shown as "Registered (server=64.xxx.xxx.xxx:5060; expire in 38s)"

Quote
What are you dialing and what is the response you're getting?

**6 011 188888 should reach the SIP Broker Test Announcement.

**7 0000 0091 should reach the Inum Echo Test.
I dialed both when I was at work dialing in through AA. Neither one work. The prompt is a male voice like "The number you dialed **6 011 188888 is not valid".  I noticed that after pressing **, the two "*" turned into 'P' on the display.

But when I get home and dialed directly through  my home phone, it works! :)  What's the difference? I guess the remote system's ** are not interpreted properly.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on October 21, 2011, 05:37:04 pm
I dialed both when I was at work dialing in through AA. Neither one work. The prompt is a male voice like "The number you dialed **6 011 188888 is not valid".  I noticed that after pressing **, the two "*" turned into 'P' on the display.

But when I get home and dialed directly through  my home phone, it works! :)  What's the difference? I guess the remote system's ** are not interpreted properly.

The difference is the Auto Attendant has it's own DigitMap and OutboundCallRoue.  If you want the Auto Attendant to behave similar to the PHONE Port, you must make corresponding changes there.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: Obvdobi on October 21, 2011, 05:41:25 pm
This makes perfect sense.  Thank you, Ron. You are so HELPFUL! I learned a lot from you in the last couple of days.


The difference is the Auto Attendant has it's own DigitMap and OutboundCallRoue.  If you want the Auto Attendant to behave similar to the PHONE Phone, you must make corresponding changes there.



Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: pooh-bah on January 17, 2012, 04:11:45 pm
Has anybody successfully used localphone on one of the gateways? I got callcentric to work fine, but localphone gives me a 407 error.

I've got GV on SP1 and Anveo on SP2. I'm trying to "route" (not sure that's the right word) the localphone calls through SP2.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on January 17, 2012, 04:18:01 pm
Has anybody successfully used localphone on one of the gateways? I got callcentric to work fine, but localphone gives me a 407 error.

I'd have to do a forum search to be sure, but I'm almost posiive Localphone works on Voice Gateways.

Are you sure you have the correct AuthUserID and AuthPassword entered in the Voice Gateway settings?


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: pooh-bah on January 17, 2012, 04:27:34 pm
I think I found my problem. Localphone's website seems to bury the AuthUserName and AuthPassword in an odd place on their website. I can't try the connection until later, but it should work now.

Stewart's post here helped: http://www.obitalk.com/forum/index.php?topic=1731.0


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on January 17, 2012, 04:34:45 pm
I've got GV on SP1 and Anveo on SP2. I'm trying to "route" (not sure that's the right word) the localphone calls through SP2.

I assume you're planning on using Localphone on a Voice Gateway to make outgoing calls?


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: pooh-bah on January 17, 2012, 08:23:09 pm
Yes, I am setting up localphone on gateway #3. I was using the wrong long/pass since I couldn't find the proper login/pass on the website.

It's all up and running, and now I'm searching for a way to have localphone send a different caller ID. This may not be possible, but I did get callcentric and anveo to send different caller ID's.

Thanks


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: Stewart on January 17, 2012, 09:07:31 pm
Localphone does not permit you to spoof caller ID on a SIP call.

If you want to send your GV number on a Localphone call, you can call from GV via Localphone calling card (requires two-stage dialing) or Local Numbers (to contacts you have set up on Localphone).

For a single contact, you can spoof the caller ID by forwarding your Localphone iNum to the contact, then sending the call to yourlpinum@sip.inum.net, with the desired caller ID.

Voxbeam (another Localphone brand) allows you to send an arbitrary caller ID with each call, though there are some gotchas.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: QBZappy on April 13, 2012, 08:07:55 am
OBihai,
As you can see by the number of views on this thread a fair number of people have read it. This specialized OBi config would be worthy of a "sticky". I can think of at least two more specialized OBi configs which are also worthy candidates developed by RonR.

RonR,
If we ask politely perhaps some of your setups can be put up as a "sticky". Can you suggest some setups you think are noteworthy. Some of these configs show the flexibility of the OBi device. I would think that this would be a good selling point for the device. I'm not sure if they require your permission, give you the credit or if it is already in the public domain.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: Shammi on April 20, 2012, 09:22:16 pm

Label : ste
DigitMap : (1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|<1aaa>xxxxxxx|011xx.|(Mipd)|[^*]@@.'@'@@.)

where aaa is your local area code.


Hi RonR,

I am using VG3 for LocalPhone to callIndia from Canada. LocalPhone requires the prefix 91 (India country code) before dialling all indian numbers. On the other hand, if I have to call other countries, I have to dial their country codes, e.g. for USA I have to add prefix ’1'

I am following the quoted method for it. My goal is to be able to dial Indian number on LocalPhone VG3 without the prefix 91. If I am correct, the 10 digit number would be dialled as: **3 xxx xxx xxxx

To achieve this can I change :
User defined Digit Map 3> DigitMap > (91xxxxxxxxxx|<91>[1-9]xxxxxxxxx|<91>xxxxxxxxxx|011xx.|(Mipd)|[^*]@@.'@'@@.)

If this is incorrect, please correct me.

I have two more questions:

1) Can I call call through the VG 3 by using my cell phone that is one of the trusted number on the obi?
2) Can there be a setting that when I call India, I don't need to dial country code prefix 91 (like above); still be able to call other countries using the same vg3. I mean, can vg3 differentiate when I call India dialling **3 + 10 digit Indian number, at the same time next call I can make to USA dialling **3 + 1 + 10 digit number.

thanks in advance

Shammi


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: Shammi on April 20, 2012, 09:26:11 pm

Label : ste
DigitMap : (1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|<1aaa>xxxxxxx|011xx.|(Mipd)|[^*]@@.'@'@@.)

where aaa is your local area code.


Hi RonR,

I am using VG3 for LocalPhone to callIndia from Canada. LocalPhone requires the prefix 91 (India country code) before dialling all indian numbers. On the other hand, if I have to call other countries, I have to dial their country codes, e.g. for USA I have to add prefix ’1'

I am following the quoted method for it. My goal is to be able to dial Indian number on LocalPhone VG3 without the prefix 91. If I am correct, the 10 digit number would be dialled as: **3 xxx xxx xxxx

To achieve this can I change :
User defined Digit Map 3> DigitMap > (91xxxxxxxxxx|<91>[1-9]xxxxxxxxx|<91>xxxxxxxxxx|011xx.|(Mipd)|[^*]@@.'@'@@.)

If this is incorrect, please correct me.

I have two more questions:

1) Can I call call through the VG 3 by using my cell phone that is one of the trusted number on the obi?

2) Can there be a setting that when I call India, I don't need to dial country code prefix 91 (like above); still be able to call other countries using the same vg3. I mean, can vg3 differentiate when I call India dialling **3 + 10 digit Indian number, at the same time next call I can make to USA dialling **3 + 1 + 10 digit number. If it is not possible, I would prefer the freedom of not adding 91 prefix for indian numbers, and probably make another account for other countries.

thanks in advance

Shammi


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on April 20, 2012, 10:03:23 pm
If you want to use VG3 for calling India only and have 91 automatically prepended to 10-digit numbers:

Voice Gateway3:
Name : LocalPhone
AccessNumber : SPx(LocalPone Proxy Server)
DigitMap : (<91>xxxxxxxxxx|91xxxxxxxxxx)
AuthUserID : LocalPhone_UserID
AuthPassword : LocalPhone_Password


If you want to use VG4 for general international calling where you dial country code + number:

Voice Gateway4:
Name : LocalPhone
AccessNumber : SPx(LocalPone Proxy Server)
DigitMap : (xx.)
AuthUserID : LocalPhone_UserID
AuthPassword : LocalPhone_Password


If you're configuring your OBi with the OBiTALK Web Portal and Circle-of-Trust, the only option for single-stage dialing from an OBiON App is through the PrimaryLine (which cannot be VG3).

Using Single-Stage Dialing Through Any OBi Trunk (http://www.obitalk.com/forum/index.php?topic=1103.0), OBiON Apps can use any trunk they wish and can have any trunk as their PrimaryLine.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: Shammi on April 21, 2012, 09:56:54 am
Wow, thanks for simplifying it. You are always at the top of it. Thanks again
Shammi


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: Shammi on April 21, 2012, 10:52:51 am
If you want to use VG3 for calling India only and have 91 automatically prepended to 10-digit numbers:

Voice Gateway3:
Name : LocalPhone
AccessNumber : SPx(LocalPone Proxy Server)
DigitMap : (<91>xxxxxxxxxx)
AuthUserID : LocalPhone_UserID
AuthPassword : LocalPhone_Password


Hi RonR,
I was not successful, it says, "there is no call route available". I triple checked the ID, password and server information. However, the one you suggested for **4 works fine. Thanks. Shammi


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: RonR on April 21, 2012, 11:15:32 am
Shammi,

Sorry 'bout that.  The DigitMap should be : (<91>xxxxxxxxxx|91xxxxxxxxxx)


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: Shammi on April 21, 2012, 11:36:18 am
Thanks, RonR. Worked perfect.  :)


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: SIMPLE_desires on May 17, 2013, 02:30:08 pm
I've been using the ObiTalk.com portal to manage the device.  Will these new settings be manageable from the portal or do I need disable portal to manage manually?

Or can I manage manually and set portal to READ ONLY?



Label : ste
DigitMap : (1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|<1aaa>xxxxxxx|011xx.|(Mipd)|[^*]@@.'@'@@.)


Please consider adding extra direction to find "user digit maps".  Albeit frazzled I did eventually find the setting by opening the "user settings" tree.

Not being familiar entirely with obi_110 I was able to follow the directions otherwise.

[ i ]
I would like MORE VoiceGateways.

(a)
 Without buying another device can I have more somehow?  I use them to spoof outbound CallerID to match other devices/lines I own.

(b)
Can I have more by daisy chaining OBIs somehow?


[ ii ]
For iNum and SIP broker I'm using the enhanced firmware syntax:

Code:
SP1(sip.inum.net;op=mns)
SP1(sipbroker.com;op=mns)

For SipBroker someone with a high post count suggested changing digitmap to

Code:
(<*>[x*][x*].S4|*[x*][x*].S4)

[ iii ]
I used VG8 but had to change PHY: Phone Port: DigitMap & OutboundCallRoute

The latter was NOT of the form: {(<**N:>(MvgN)):vgN} |N=single digit.  It had "li" and some stuff I neglected to retain.

Will using VG8 this way deprive me of other LI functionality?







Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: heny on June 25, 2013, 05:13:17 pm
This post is very helpful for beginners to setup voice gateway, thanks!

I have a few questions:

1. the post using vg3,4,6,7 as example, I assume I can follow same logic and setup vg1,2,5,8? (for example **0 to route to vg1 etc)
2. the phone port digital map on my obitalk is slightly different than the first post, it is: ([1-9]x?*(Mpli)|[1-9]S9|[1-9][0-9]S9|911|**0|***|#|**1(Msp1)|**2(Msp2)|**9(Mpp)|(Mpli)), where can I find the explanation on each item?
3. one of the posts mentioned "The records in syslog", where can I find the syslog? does it contain all the call logs as well?

thanks



Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: ipse on January 06, 2014, 03:47:03 pm
Reviving an old topic, but I have a question: when using a VG (my case for 1800 dialing -for free) does Obi use the DTMF settings for SPx?

My setup: SP2 =Anveo, VG3=CallWithUs and VG3 is set to use SP2.
I am having problems passing DTMF tones AFTER I get connected (conf bridge) and I'm guessing if I make changes to SP2 DTMF settings it won't matter anyways.

I have another VG6 set up with tollfree and that one seems to be working much better (less than 5% errors) - so I'm afraid this just boils down to CWU performance.
Anyone else having trouble with CWU and DTMF? Could this be specific to the use of a VG instead of SP?

/EDIT I have the same issue with IPKall and toll free calls. Like I said, tollfreegateway works almost perfect - they all hand off SP2.

/EDIT2 I take back what I posted before: making DTMF changes for ITSP B (Anveo) DOES improve DTMF sending...I have it set to Inband now and digit success rate is close to 100%. I would not have guessed that params from SPx are actually used by VGs as a "fake" SIP account can also be used.
I toggle between Inband and RFC2833 and I see the difference immediately.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: ob1joef on June 15, 2019, 03:11:12 pm
OBi200 and Voice Gateways : Issues??


Once upon a time I had an OBi110 and I successfully installed  working Voice Gateways on it.
I followed the instructions on  the OBi web forum for an OBi110  (above).

I repeated those instructions in that  ...
I tried replicating the VGs on an OBi200 and keep getting "No service configured" error.

I've gone over my work many times for syntax errors, etc.

I am trying to have a callcentric dial out
I tried @srv.callcentric.com  that I've successfully used with a SP3102
and tried stun.callcentric.com, too.   

But I'm failing before it can get to calling out to Callcentric.  So I'm lost.

Anybody know of issues with Voice Gateways moving from an OBi110 and an OBi200?

thanks in advance
Joe F


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: drgeoff on June 15, 2019, 04:25:22 pm
@ob1joef

Please do not post the same content more than once or in separate topics. http://www.obitalk.com/forum/index.php?topic=16015.msg99604#msg99604


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: ob1joef on June 26, 2019, 02:55:52 pm
Using my OBi200 with the latest fw 3.2.2 (Build: 5921EX) on HV 1.4
I've followed the directions to the letter at the top of this post to implement VG
 and always get:

"No service configured error"
"Please log into OBiTalk and configure your device"

SP1-4 work fine using Google Voice and Callcentric
but trying to use a Voice Gateway with a working Callcentric account
gets me this error.

Original HW & PTS warranty ended Jul 2018 so I paid for a year of Premium Tech Support
but have not heard from them in 10 days.

Any more suggestions?
Can anyone confirm that THEY are using Voice Gateway on an OBi200???

thanks,
Joe f


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: azrobert on June 26, 2019, 03:37:47 pm
I'm on the same fw and voice gateways work for me.

Are you VoipHomeUser on the DSLReports forum? Did you try my suggestion here:
https://www.dslreports.com/forum/r32423568-

This will narrow the cause of the problem. If the speed dial works then it's a routing problem. If it fails then it's a VG definition problem.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: zaelkaleem on December 17, 2019, 09:36:30 pm
Ever wish the OBi supported using additional SIP providers for more outbound calling options?  Ever wish you could call people on other VoIP networks directly from your OBi using Sip Broker?  Ever wish you could call people directly from your OBi using their iNum number?  Well, you can, thanks to the Voice Gateways present in the OBi.  In this example, I'll show you how to add two additional SIP providers, calling via Sip Broker, and iNum calling.

NOTE:  You must have at least one OBi Voice Service (SPx/ITSPx) configured for SIP.  If you don't wish to configure a SIP provider on an SPx/ITSPx, simply set Service Providers -> ITSPx -> SIP -> ProxyServer to 127.0.0.1 and uncheck Voice Services -> SPx -> X_RegisterEnable.  Also, the SIP providers used in Voice Gateways must allow calling without SIP registration (many do, some don't).  Sip Broker and iNum calling do not use SIP registration.

The additional calling capability is added through the use of **3, **4, **6, and **7 dialing prefixes.  [**5 cannot be used as a dialing prefix because it's hard-coded into the OBi for use by Obihai.]  In this example, **3 and **4 will be used for additional SIP providers, **6 will be used for calling via Sp Broker, and **7 will be used for iNum calling.

To begin, you'll need to make a couple of additions to your PHONE Port DigitMap and PHONE Port OutboundCallRoute to add support for the new dialing prefixes:


Phone Port DigitMap:

|**1(Msp1)|**2(Msp2)|**3(Mvg3)|**4(Mvg4)|**6(Mvg6)|**7(Mvg7)|**8(Mli)|**9(Mpp)|

PHONE Port OutboundCallRoute:

{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{(<**3:>(Mvg3)):vg3},{(<**4:>(Mvg4)):vg4},
{(<**6:>(Mvg6)):vg6},{(<**7:>(Mvg7)):vg7},
{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}


This creates the following associations:

**3 -> Voice Gateway3 (VG3)
**4 -> Voice Gateway4 (VG4)
**6 -> Voice Gateway6 (VG6)
**7 -> Voice Gateway7 (VG7)


Now, let's set up additional VoIP providers on Voice Gateway3 and Voice Gateway4.

Voice Gateway3 will be used with Whistlephone:

Name : Whistlephone
AccessNumber : SPx(proxy.whistlephone.com)
DigitMap : (Mste)
AuthUserID : your_whistlephone_user_id
AuthPassword : your_whistlephone_password


Voice Gateway4 will be used with IdeaSIP:

Name : IdeaSIP
AccessNumber : SPx(proxy.ideasip.com)
DigitMap : (Mste)
AuthUserID : your_ideasip_user_id
AuthPassword : your_ideasip_password


Next, let's configure Voice Gateway6 for calling via Sip Broker:

Name : Sip Broker
AccessNumber : SPx(sipbroker.com)
DigitMap : (<*>[x*][x*].|*[x*][x*].)


Next, let's configure Voice Gateway7 for iNum calling:

Name : iNum
AccessNumber : SPx(sip.inum.net)
DigitMap : (<8835100>xxxxxxxx|8835100xxxxxxxx)


And finally, let's configure the User Defined DigitMap referenced in Voice Gateway3 and Voice Gateway4:

Label : ste
DigitMap : (1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|<1aaa>xxxxxxx|011xx.|(Mipd)|[^*]@@.'@'@@.)

where aaa is your local area code.


Following a reboot, the OBi should be ready to use its new capabilities.

Dialing **3 + number should place a PSTN call using Whistlephone.

Dialing **4 + number should place a PSTN call using IdeaSIP.

Dialing **6 + SIP code + number should place a VoIP call via Sip Broker.

For example:

Dialing **6 011 188888 should connect you with the Sip Broker test announcement.
Dialing **6 010 123456 should connect you with the Voxalot number 123456.
Dialing **6 747 17471234567 should connect you with the Gizmo5 number 1-747-123-4567.

For more details on Nox (https://downloadnox.onl/) Vidmate (https://vidmate.vet/) VLC (https://vlc.onl/)] Sip Broker, please visit : http://www.sipbroker.com/

Dialing **7 + number should place a VoIP call to an iNum number.

NOTE: Use only the last 8 digits of the iNum number (8835100 will be prepended for you).

For example:

Dialing **7 00000091  should connect you with the iNum echo test.
Dialing **7 04123456 should connect you with the Voxalot number 123456.
Dialing **7 71234567 should connect you with the Gizmo5 number 1-747-123-4567.

For more details on iNum, please visit : http://www.inum.net/
what I get is just one way audio.
Especially calling any peers through sipbroker.


Title: Re: Using OBi Voice Gateways with SIP Providers
Post by: drgeoff on December 17, 2020, 08:13:06 am
Digit Maps only affect what called number is sent to what ITSP.  If you call reaches the desired far end number via the desired ITSP. then a digit map has no effect whatsoever on audio quality or lack thereof.