I don't see anything obvious right away other than have you confirmed that the phone is actually working? Can you get it to ring? With my Sipura adapters that use Linksys software I can view the call status in the "Info" section which if you have that panel might tell you if the adapter thinks a call is coming in. I just looked at my Info page with a call coming in and I can see the call state as Ringing and a bunch of other details.
Call Status Call 1 State: Ringing Call 1 Tone: Ring - Hold Call 1 Encoder: G711u Call 1 Decoder: G711u Call 1 FAX: No Call 1 Type: [L1]Inbound Call 1 Remote Hold: No Call 1 Callback: No Call 1 Peer Name: UNAVAILABLE Call 1 Peer Phone: 1XXXXXXXXX Call 1 Duration: Call 1 Packets Sent: 0 Call 1 Packets Recv: 0 Call 1 Bytes Sent: 0 Call 1 Bytes Recv: 0 Call 1 Decode Latency: 0 ms Call 1 Jitter: 0 ms Call 1 Round Trip Delay: 0 ms Call 1 Packets Lost: 0 Call 1 Packet Error: 0 Call 1 Mapped RTP Port: 16420 >> 0 [EMAIL PROTECTED] wrote: > They are on the same lan > > the adapter is registered > > sip show peers > Name/username Host Dyn Nat ACL Port Status > sippyskypeuser/sippyskype 192.168.2.76 5070 OK (1 ms) > 1000/1000 192.168.2.76 D 5061 OK (1 ms) > freephonie-out/0950607456 212.27.52.5 N 5060 OK (766 ms) > callcentric/17772962667 204.11.192.34 N 5080 OK (206 ms) > > the pap2t's IP is 192.168.2.205 > and the IP of the asterisk box is 192.168.2.76 > > sip show registry > Host Username Refresh State > Reg.Time > freephonie.net:5060 095060xxxx 1785 Registered > Wed, 09 Jul 2008 10:12:44 > callcentric.com:5080 177729xxxxx 46 Registered > Wed, 09 Jul 2008 10:13:29 > > I use line2 of my pap2t (line 1 is not enabled). Here is the conf : > http://emmanuelfavrenicolin.free.fr/Public/Divers/Snapshots1/20080709_pap2t.jpg > > > On 7/9/08, MFH <[EMAIL PROTECTED]> wrote: > >> Are asterisk and the phone on the same lan? I see you have nat=no. Do >> you see the phone adapter registered? >> >> Emmanuel Favre-Nicolin wrote: >> >>> Hi, >>> >>> I'm having a problem to receive inbound call from my sip provider. I used >>> to >>> be OK, I may I have change something (for example I switched from asterisk >>> >>> 1.4.20.1 to 1.4.21.1). Could that be a bug? (I doubt that and I guess it a >>> >>> configuration problem on my side!) >>> >>> I have basically a sip account and a linksys voip adapter with a phone on >>> it >>> (sip name 1000), configured in asterisk. Outbound call from the phone just >>> >>> work fine. Inbound call fail to ring my phone. When the inbound call occur >>> I >>> see on the asterisk command line : >>> >>> -- Executing [EMAIL PROTECTED]:1] >>> Dial("SIP/callcentric.com-081f1ac8", "SIP/1000") in new stack >>> >>> -- Called 1000 >>> >>> -- SIP/1000-081ed5e0 is ringing >>> >>> but my phone is not ringing >>> >>> in sip.conf: >>> >>> [1000] >>> type=friend >>> secret=blablabla >>> qualify=yes ; Qualify peer is not more than 2000 mS away >>> nat=no ; This phone is not natted >>> host=dynamic ; This device registers with us >>> canreinvite=no ; Asterisk by default tries to redirect >>> context=fromsoftphone >>> port=5061 ; Uncomment this line if Ekiga and Asterisk are on the same >>> host >>> >>> >>> in extensions.conf: >>> [from-callcentric] >>> exten => 17772962667,1,Dial(SIP/1000) >>> exten => 17772962667,n,Hangup() >>> >>> >>> The default extension I got for inbound call is 17772962667 that's why I >>> used >>> that extension. I tu >>> >>> _______________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >>> Register Now: http://www.astricon.net >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users