Modification and Gateway usage question.

Started by neftv, April 06, 2011, 04:11:23 PM

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neftv

As previously suggested in my post from March 29 I have the following. (was/is working good)

Modified ITSPA DigitMap:

(1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|<**2>011xx.|xx.|(Mipd)|[^*]@@.)

Modified PHONE Port OutboundCallRoute:

{([1-9]x?*(Mpli)):pp},{911:sp1},{(<#:>):li},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},
{(<**2:>(Msp2)):sp2},{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}


What I am looking at doing...
SP1 is my primary domestic Voip provider and for 911.

I now want to free up my SP2 if I can.  I want to take Google Voice for a spin on this Device.

I like to use Gateway or Trunk to do the following.
Have two different Voip providers for two different countries.
when dialing international "01161"  (Australia) it goes to PennyTel I guess in position Voice Gateway1
When dialing international "01130"  (Greece)  it goes to another provider in positional Voice Gateway2.  Not sure if these should be trunk or Gateway.

Or maybe if I just direct Australian international calls to PennyTel and the rest of the international calls to another provider that would work if it makes programming easier.




RonR

As long as a VoIP provider (PennyTel for example) allows outgoing calls without SIP registration, there shouldn't be a problem using them through a Voice Gateway.  Once that's done, it shouldn't be a problem routing specific country codes to the appropriate provider.  For example, using Voice Gateway3 and Voice Gateway4:


Modified PHONE Port OutboundCallRoute:

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


Modified ITSPA DigitMap:

(1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|<**3>01161xx.|<**4>01130xx.|xx.|(Mipd)|[^*]@@.)


For completeness and uniformity, I'd follow this cookbook for starters:

http://www.obitalk.com/forum/index.php?topic=526.0

neftv

Thanks for the info. I will look at the site you gave. This is very new to me.
I did make the changes you suggested and I tried the gateway 4 provider but I seem to get a fast busy.  And it seems the call never even makes it to the provider so something is not happy in the Obi.  Perhaps the deal with my main Voip provider not allowing outgoing calls without SIP registration is the issue? 

RonR

It would be best to configure VoIP providers on Voice Gateways and get them working per the instructions at the link I referenced before trying to lash them into your ITSPA DigitMap.

neftv

I followed the example in the link just for sip broker. When I do that sip test call I get a message saying "there is no service to complete your call"    I think its the adapter telling me this.
I have QuantumVoice for SIP 1 and PennyTel for SIP 2.    thoughts?

RonR

Did you add the additional rules (or at least the **6 rules) to the PHONE Port DigitMAP and OutboundCallRoute?

neftv

Yes I believe so. I inserted the extra parameters. This is what I have
Digimap
([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))
Outbound call route
{([1-9]x?*(Mpli)):pp},{911:sp1},{(<#:>):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}

RonR

Look right at first glance.

And Voice Gateway6 is configured?:

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

where SPx is SP1 or SP2 where you have a VoIP provider configured.


Sip Broker is alive and well working here.

neftv

#8
Ok I am making some headway here.
I forgot put SPx as SP1 where my domestic provider is.

Now when I dial **6 011 188888  I get silence then it hangs up after a few seconds and goes back to dial tone.

I decided to put my smartvoip provider in position 4 and that worked.  

Don't know if I was suppose to get a welcome message on SIP broker as I thought I would.

update...
I moved Pennytel to position 3 and that works they have a echo test.  

I guess I am on my way.
I not sure why I don't receive back from Sipbroker.

RonR

You were probably getting one-way audio on the **6 011 188888 call.  The current implementation of SIP calling through Voice Gateways isn't as robust as it could be when it comes to the audio (RTP) side of things.

neftv

Is that something they going to improve on?

let me ask you when a provider gets put in a gateway position  then that provider is only for outgoing calls correct?  Incoming calls wont come through?

RonR

Quote from: neftv on April 06, 2011, 08:21:15 PM
Is that something they going to improve on?

Good question.  I don't know the answer, but I hope so.

Quote from: neftv on April 06, 2011, 08:21:15 PM
let me ask you when a provider gets put in a gateway position  then that provider is only for outgoing calls correct?  Incoming calls wont come through?

Correct.

neftv

would having a stun server put in on my main voip provider help any with the RTP issue with sipbroker?

RonR

Unfortunately it won't (that's one of the shortcomings).

From the OBi Device Administration Guide:

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.

neftv

That is interesting today I had to reboot the Obi100 because on the one gateway provider that was working yesterday I tried to use it today and I was getting a voice prompt saying the I got no response from service provider.  When I rebooted the Obi it worked fine.  Is this part of that issue your talking about in the previous post?  Does Obi tech support read these?

RonR

Quote from: neftv on April 07, 2011, 10:26:17 AMWhen I rebooted the Obi it worked fine.  Is this part of that issue your talking about in the previous post?
It's hard to tell.  SIP/RTP through NAT routers is a bit fragile under the best of conditions.  The additional restrictions stated above certainly don't help the situation.


Quote from: neftv on April 07, 2011, 10:26:17 AMDoes Obi tech support read these?
Sometimes it's clear they do.  Other times, you have to wonder.   :)