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--- RE: SIP Inbound calls & SIP URI destination issues (http://www1.pbxes.com/forum/threadid.php?threadid=1274817006)

SIP Inbound calls & SIP URI destination issues

I have been struggling with my setup.

I currently have 2 Trunks that I use both for inbound and outbound routing.

1) Cybersurf Internet Access - Local DID number - Trunk name = CyInAc
2) Virtual SIP DID - Trunk name = VoipMS

If I setup the following only using 1 inbound route for all calls, I can receive calls successfully on both CyInAc DID and VoipMS Virtual SIP DID.

But as soon as I make any of the following changes, I don't get the result required:

1) Keeping 1 inbound route, but choosing the SIP URI destination, where I input a SIP URI of a place that is known to receive inbound sip calls, the result is getting a busy signal when an inbound call is received on either CyInAc or VoipMS DIDs.

2) From above, I change SIP URI back to ringing an extension when an inbound call is received, but this time instead of keeping the inbound trunk name blank, I change it to Voip.ms . The result is also getting a busy signal when calls are received on the VoipMS Virtual SIP.

What I would like to accomplish in the end is having 2 Inbound routes. If someone calls the CyInAc DID, then calls will ring an extension, if someone calls the VoipMS Virtual SIP during regular hours, then the call goes to a Digital Receptionist, after hours then the call should be sent out to a SIP URI. But due to the above 2 issues, I cannot get this to work and cannot figure out why. Any help/responses would be appreciated.

djino

RE: SIP Inbound calls & SIP URI destination issues

Hi Diafora,

Even after you've made the changes to my account (changing the trunk names/etc), I still get a busy signal when I call 111111727111@toronto.voip.ms from another sip account (i.e. callcentric.com)

Even calling djino@pbxes.org gets a busy signal.

FYI: the above voip.ms account is the Voip.ms virtual DID, that gets forwarded to the Voip.ms account I have a trunk for.

Again note, this all works perfectly when I reduce it all to 1 inbound trunk that is left blank (for the inbound trunk name).

djino

RE: SIP Inbound calls & SIP URI destination issues

Do you have access to a real DID, not a virtual one, that can be called from the PSTN? If the routing for real DIDs works, then we can concentrate on routing calls to SIP URIs.

As an aside, please note the dial pattern 1NxxNxxxxxx permits access to the complete North American continent, which except the US & Canada includes countries like Bermuda, Barbados, Bahamas, Dominican Republic, Grenada, Jamaica, etc. These destinations are usually charged at higher rates than continental North America, and usually should be excluded from your North American Custom Dial Patterns.

RE: SIP Inbound calls & SIP URI destination issues

Diafora,

The CyInAc one is a real DID that can be accessed by PSTN. The Voip.ms DID, is only SIP.

They are both real DID. The Voip.ms is only Virtual in the sense that it can only be accessed via SIP. The CyInAc (cybersurf Internet Access) is a PSTN DID with my Local Gatineau, Quebec number (as you can see when you look at the username).

Yeah, you are right about the dial-pattern. Just not sure how I can configure a dial-plan that only includes US/Canada calling.

djino

RE: SIP Inbound calls & SIP URI destination issues

Please keep in mind that, not all ITSPs allow inbound SIP URI calls without any restrictions. PBXes happens to allow them and as such a call to djino-test@pbxes.org should end up on the extensions of Ring Group 1.

In addition, you have to ensure which is your correct VoipMS SIP URI. If the user part of your URI is 111111727111 then your complete URI might be 111111727111@voip.ms and not necessarily 111111727111@toronto.voip.ms

Trial and error will usually yield the correct answer.

RE: SIP Inbound calls & SIP URI destination issues

• Try a call to your local Gatineau number, and verify whether it rings on the extensions of Ring Group 1.
• The "NorthAmericanOnly" route has been updated as well, with the current NPA Nxx list.

RE: SIP Inbound calls & SIP URI destination issues

Zitat:

Originally posted by Diafora • Try a call to your local Gatineau number, and verify whether it rings on the extensions of Ring Group 1.

Just tried calling from my cell. Home line doesn't ring. Cell drops the call.

Zitat:

Originally posted by Diafora• The "NorthAmericanOnly" route has been updated as well, with the current NPA Nxx list.

Thank you very much appreciate it

djino

RE: SIP Inbound calls & SIP URI destination issues

After a few iterations the call to the Gatineau number ends up on a Voice Mailbox. I am not sure where it resides though.

• What kind of SIP UAs are the extensions 100 and 200 registering on?
• What kind of broadband router is connecting the above extensions to the Internet?

RE: SIP Inbound calls & SIP URI destination issues

Zitat:

Originally posted by Diafora After a few iterations the call to the Gatineau number ends up on a Voice Mailbox. I am not sure where it resides though.

Yes, I saw when you made that call. That is the correct outcome, as you heard it ring several times then a voicemail picked up.

The Gatineau DID is fine, I never had issue here. The only time this was an issue is when 1) I edit this Gatineau Inbound route for it and tell it to go to a SIP URI.. and tell it to send the call to an outside SIP Address. I get a busy signal here. Its as if the SIP URI destination option alone for pbxes does not function as it does not dial-out to another SIP Address that is placed there.

2) Besides that issue, the other main issue as described in the original post is when using callcentric.com SIP (with X-Lite application), I will dial 11111727111@toronto.voip.ms , this is a Virtual SIP that gets forwarded to my Voip.ms account (the 111727 account @ toronto.voip.ms), but pbxes does not answer the call when there is more than 1 inbound route.

If I have just 1 inbound route, the SIP inbound call rings the extension I have set. But if we create 2 inbound routes (one for the Gatineau DID, and another for the SIP DID, then pbxes can't seem to find the VoipMS inbound route, so it just returns a busy signal.

djino

EDIT: Actually 1) works now. I had to change the inbound route name from Voip.ms to 11111727111

Now when someone from outside calls 11111727111@toronto.voip.ms , it will ring the extensions listed for that inbound route.

2) This issue is still pending (not working). ie. If I were to tell it to forward an incoming call to a callcentric SIP, the SIP URI destination function doesn't appear to be working.

EDIT: Found a work around for #2.

djino
"I'm good to go now, thanks for your help!"

RE: SIP Inbound calls & SIP URI destination issues

I am glad that all your issues have been resolved.