Extension 500 is registered just fine. 200 OK Maybe you should start with a GUI version of Asterisk.
Try calling out via bandwidth with SIP verbose on and post your results. Call the other phone and post verbose. You do have logic in extensions.conf do you not? Thanks, Steve Totaro On Fri, Jul 10, 2009 at 10:58 AM, Ott Rose <sixfourimp...@hotmail.com>wrote: > Carrier is bandwidth.com > > we are running Asterisk 1.6.1.1 > > i ran sip set debug on from the CLI > > Once i did a module reload it started displaying all the debuging info. > Here is some of the debug info > > > --- (13 headers 0 lines) --- > Scheduling destruction of SIP dialog ' > 5fafc3b57e3928877141d12f58c9f...@127.0.0.2' in 32000 ms (Method: REGISTER) > [Jul 10 06:53:39] NOTICE[5641]: chan_sip.c:16397 handle_response_register: > Outbound Registration: Expiry for dynamic is 120 sec (Scheduling > reregistration in 105 s) > > <--- SIP read from UDP://127.0.0.1:5060 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 127.0.0.1:5060 > ;branch=z9hG4bK613dcb99;received=127.0.0.1;rport=5060 > From: <sip:5...@dynamic>;tag=as51c22cdd > To: <sip:5...@dynamic>;tag=as51c22cdd > Call-ID: 7e1e2c4c702c5b1619fef39612192...@127.0.0.2 > CSeq: 117 REGISTER > Server: Asterisk PBX 1.6.1.1 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces, timer > Expires: 120 > Contact: <sip:5...@127.0.0.1 <sip%3a...@127.0.0.1>>;expires=120 > Date: Fri, 10 Jul 2009 10:53:39 GMT > Content-Length: 0 > > ------------------------------ > Date: Fri, 10 Jul 2009 09:42:31 -0400 > > From: stot...@asteriskhelpdesk.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] setting up phones > > Who is the carrier? What flavor of Asterisk are you using? > > Regardless, the phones should register and be able to call each other and > other Asterisk apps if you have them in the dialplan. > > If you go to the Asterisk CLI and turn on SIP debugging, do you get > anything at all? > > also, change registersip to yes. > > Thanks, > Steve > > On Fri, Jul 10, 2009 at 9:32 AM, Ott Rose <sixfourimp...@hotmail.com>wrote: > > Here is my physical network. > > We have a Adtran router that is plugged into the Asterisk server and into > the circuit provided by my tel co. > > the other nic in the Asterisk box is plugged into your lan switch > > the phones are plugged into the lan switch > > > I can ping the phones from the Asterisk server. > > ------------------------------ > Date: Thu, 9 Jul 2009 17:42:43 -0400 > From: stot...@asteriskhelpdesk.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] setting up phones > > > > On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose <sixfourimp...@hotmail.com>wrote: > > I followed it the best I could. the phones say no service. I haven't got > to setting up the SIP trunk yet I was told I could get the extensions to > work so I could test between the two phones i have. I have to nics in my > server. one is connect to the phone router the other to a network switch. > which ip should it point to? I am guess the one connected to the switch. > That is the one i can access the GUI from. Below are my users.conf setting. > Notice all the spaces. I didn't put them in there they are like that in the > conf > > > Either you did not explain your network topology very well or that is your > problem. > > Unless you are trying to segregate your VoIP traffic, plug everything into > the switch. > > If using DHCP, get the IP and try pinging the phones from the Asterisk box. > > I bet it is just a network issue. > > -- > Thanks, > Steve Totaro > +18887771888 (Toll Free) > +12409381212 (Cell) > +12024369784 (Skype) > > ------------------------------ > Windows Live™: Keep your life in sync. Check it > out.<http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009> > >
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