I saw 127.0.0.2, never seen that before. Loopback that I have seen is 127.0.0.1.
I always just bind to 0.0.0.0 since I have never really seen a point to binding to a specific IP. I guess if you are dual homed and don't want remote phones to work, but then you could just block that stuff in IPTables or whatever firewall. Thanks, Steve T BTW, what GUI? That was part of what I was asking when I said "what flavor of Asterisk?" On Fri, Jul 10, 2009 at 11:51 AM, Danny Nicholas <da...@debsinc.com> wrote: > You are running asterisk as a local service (127.0.0.1 is localhost). > You need to use the address from ifconfig (192.168.X.X) in sip.conf > (bindaddr). This will make asterisk where your phones can “talk” to it and > register. > > > ------------------------------ > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ott Rose > *Sent:* Friday, July 10, 2009 10:33 AM > > *To:* asterisk-users@lists.digium.com > *Subject:* Re: [asterisk-users] setting up phones > > > > > so i filled in the info and now i get this when i run sip show peers > Name/username Host Dyn Nat ACL Port Status > 500/500 127.0.0.1 D 5060 OK (1 ms) > 501/501 127.0.0.1 D 5060 OK (1 ms) > 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 > offline] > > > I still cannot call the extensions and the phones say no service on there > screen > ------------------------------ > > From: da...@debsinc.com > To: asterisk-users@lists.digium.com > Date: Fri, 10 Jul 2009 08:40:49 -0500 > Subject: Re: [asterisk-users] setting up phones > > Let’s draw this out and let you fill in the blanks. Your asterisk server > has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has > ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. > > > > Sip.conf should look this > > > > [phone1] > > type=peer > > context=phones > > host=dynamic > > fromuser=phone1 > > secret=secret1 > > canreinvite=no > > directrtpsetup=no > > call-limit=3 > > nat=no > > qualify=yes > > register=no > > session-timers=accept > > session-expires=60 > > session-minse=120 > > session-refresher=uac > > register => phone1:secr...@foobar.com/phone1 > > defaultip=192.168.23.2 > > mailbox=1001 > > disallow=all > > allow=alaw > > [phone2] > > type=peer > > context=phones > > host=dynamic > > fromuser=phone2 > > secret=secret2 > > canreinvite=no > > directrtpsetup=no > > call-limit=3 > > nat=no > > qualify=yes > > register=no > > session-timers=accept > > session-expires=60 > > session-minse=120 > > session-refresher=uac > > register => phone2:secr...@foobar.com/phone2 > > defaultip=192.168.23.3 > > mailbox=1002 > > disallow=all > > allow=alaw > > > > assuming your phones are set up to contact 192.168.23.1 with username > phone1/phone2 and proper secret, all should register and you should be good > to go. > ------------------------------ > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ott Rose > *Sent:* Friday, July 10, 2009 8:33 AM > *To:* asterisk-users@lists.digium.com > *Subject:* Re: [asterisk-users] setting up phones > > > > 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> > > > ------------------------------ > > Hotmail® has ever-growing storage! Don’t worry about storage limits. Check > it > out.<http://windowslive.com/Tutorial/Hotmail/Storage?ocid=TXT_TAGLM_WL_HM_Tutorial_Storage_062009> > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype)
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users