Phone 1 has 500 in all of it's id's and connects to server 10.0.0.52?
_____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 11:05 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Great i changed it to my ip here is the debug and sip show peers. phones still say no service i get a dial tone when i pick it up and a busy signal when i call the other extension. Name/username Host Dyn Nat ACL Port Status 500/500 10.0.0.52 D 5060 OK (1 ms) 501/501 10.0.0.52 D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] --- (12 headers 0 lines) --- Really destroying SIP dialog '1e56b3345aabbe4373a46fb973476...@10.0.0.52' Method: OPTIONS linux-zswk*CLI> <--- SIP read from UDP://10.0.0.52:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.52:5060;branch=z9hG4bK1210f27b;received=10.0.0.52;rport=5060 From: "asterisk" <sip:aster...@10.0.0.52>;tag=as66b3ded8 To: <sip:5...@10.0.0.52>;tag=as66b3ded8 Call-ID: 56cf67f56e3cba6602965f6317656...@10.0.0.52 CSeq: 102 OPTIONS Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:aster...@10.0.0.52> Accept: application/sdp Content-Length: 0 _____ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 10:51:18 -0500 Subject: Re: [asterisk-users] setting up phones 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 LiveT: Keep your life in sync. Check <http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009> it out. _____ HotmailR has ever-growing storage! Don't worry about storage limits. Check <http://windowslive.com/Tutorial/Hotmail/Storage?ocid=TXT_TAGLM_WL_HM_Tutori al_Storage_062009> it out. _____ Windows LiveT SkyDriveT: Get 25 GB of free online storage. Get it on your BlackBerry or iPhone. <http://windowslive.com/online/skydrive?ocid=TXT_TAGLM_WL_SD_25GB_062009>
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