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dji
Registration Date: 01.01.1970
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SIP Inbound calls & SIP URI destination issues |
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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
This post has been edited 3 time(s), it was last edited by Dia on 26.05.2010 at 00:20.
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25.05.2010 21:50 |
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dji
Registration Date: 01.01.1970
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25.05.2010 22:35 |
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Dia
Premium Account
Registration Date: 03.03.2006
Posts: 1443
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RE: SIP Inbound calls & SIP URI destination issues |
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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.
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26.05.2010 00:18 |
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dji
Registration Date: 01.01.1970
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26.05.2010 00:23 |
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Dia
Premium Account
Registration Date: 03.03.2006
Posts: 1443
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26.05.2010 00:33 |
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Dia
Premium Account
Registration Date: 03.03.2006
Posts: 1443
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26.05.2010 00:48 |
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dji
Registration Date: 01.01.1970
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26.05.2010 00:53 |
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Dia
Premium Account
Registration Date: 03.03.2006
Posts: 1443
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26.05.2010 01:35 |
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dji
Registration Date: 01.01.1970
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RE: SIP Inbound calls & SIP URI destination issues |
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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. |
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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!"
This post has been edited 2 time(s), it was last edited by dji on 26.05.2010 at 14:27.
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26.05.2010 02:02 |
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Dia
Premium Account
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Posts: 1443
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26.05.2010 21:37 |
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