OK. I see that. Is your user behind NAT somewhere else (remote) or local to the sipx server? Can you test with a hardphone or something like xlite? I've never had great success with (sipcommunicator myself). From the trace it "appears" the user is not registered before the invite comes in, what does the registration look like in the gui?
On Sun, Oct 10, 2010 at 12:06 PM, Alexander <[email protected]> wrote: > Hi Tony, > Thank you for the quick response. > > I read wiki and have done all necessary configuration in the GUI. > > 1. System > Servers > Services > SIP Trunking > > Incoming calls destination = (empty) > > 2. Users > User ID > > Aliases = 8127407501 > > 3. Inbound request DID = 8127407501 > > INVITE sip:[email protected]:5080 SIP/2.0 > Via: SIP/2.0/UDP > 81.24.124.244:5062 > ;rport=5062;branch=z9hG4bK-622830-280081472-637534336-1483217493;received=81.24.124.244 > From: > <sip:[email protected] > :5062>;tag=607830-280081472-637534336-1483217493 > To: <sip:[email protected]:5080> > Call-ID: [email protected] > CSeq: 1 INVITE > Contact: <sip:[email protected]:5062> > Max-Forwards: 70 > User-Agent: MERA MVTS3G v.4.0.1-31 > Cisco-Guid: 2272417113-3550613983-2423062562-2438472060 > Remote-Party-ID: > <sip:[email protected]:5062>;party=calling;privacy=off;screen=no > Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,REFER,REGISTER > Content-Type: application/sdp > Content-Length: 263 > > v=0 > o=- 1286714451 1286714451 IN IP4 81.24.xx.xx > s=- > c=IN IP4 81.24.xx.xx > t=0 0 > m=audio 17794 RTP/AVP 8 0 18 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > Maybe I did not consider additional options. > > I attached screenshot with my configuration. > > > -------------------------------- > BR, > Alexander Shvaryov > > > В Вск, 10/10/2010 в 09:45 -0400, Tony Graziano пишет: > > Uncheck that. > > > > > > The alias (DID number)goes in the user alias field. > > > > > > 812xxxxxxx > > > > > > Either you want ALL CALLS from the trunk to go to the destination you > > set in the SBC field OR you want them to go to unique users based on > > the DID in the INVITE. > > > > > > 812xxxxxxx > > > > > > Put the DID number in the user alias field. Read the wiki. > > > > > > On Sun, Oct 10, 2010 at 9:13 AM, Alexander <[email protected]> wrote: > > Hello All, > > > > I'm testing SipXecs v.4.3.2-5482fd7 2010-10-08T12:12:36 > > build24 > > > > My test is simple - PSTN numbers are assigned to individual > > users (DID), > > but I have some question. > > I turned up SIP trunk with my ITSP and checked "Use built-in > > SIP Trunk > > SBC". When I use "Incoming calls destination" (SBC > > configuration) - all > > "Ok", I can to receive inbound call on softphone (Incoming > > calls > > destination = alias of my number = 812xxxxxxx). But when I > > don't use > > "Incoming calls destination" (empty), I can't receive inbound > > call. > > Prompt saying that: "If empty, inbound calls are directly > > routed to the > > specified number in the inbound request and have to be > > redirected by > > aliases or dial plan rules." The number in the inbound request > > = alias > > of my number = 812xxxxxx. > > > > I attached merge logs and scheme of my network. > > > > Any help is appreciated. > > > > > > ------------------------- > > BR, > > Alexander Shvaryov > > > > > > > > _______________________________________________ > > sipx-users mailing list > > [email protected] > > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > > > > > -- > > ====================== > > Tony Graziano, Manager > > Telephone: 434.984.8430 > > sip: [email protected] > > Fax: 434.984.8431 > > > > Email: [email protected] > > > > LAN/Telephony/Security and Control Systems Helpdesk: > > Telephone: 434.984.8426 > > sip: [email protected] > > Fax: 434.984.8427 > > > > Helpdesk Contract Customers: > > http://www.myitdepartment.net/gethelp/ > > > > Why do mathematicians always confuse Halloween and Christmas? > > Because 31 Oct = 25 Dec. > > > > > > _______________________________________________ > > sipx-users mailing list > > [email protected] > > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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