What do you get from "sip show peers" in CLI? Do you have your ip address in sip.conf?
_____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Thursday, July 09, 2009 4:12 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones 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 [501] username = 501 transfer = yes mailbox = 501 call-limit = 100 type = peer fullname = 501 registersip = no host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 501 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 501 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 disallow = all allow = ulaw,gsm macaddress = 00085d10927f autoprov = yes label = 501 linenumber = 1 LINEKEYS = 1 [500] username = 500 transfer = yes mailbox = 500 call-limit = 100 type = peer fullname = 500 registersip = no host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 500 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 500 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 macaddress = 00085d1095aa autoprov = yes label = 500 linenumber = 1 LINEKEYS = 1 _____ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Thu, 9 Jul 2009 14:03:50 -0500 Subject: Re: [asterisk-users] setting up phones It should be pretty simple. Follow the instructions on this page http://www.voiptalk.org/products/aastra-setup.html put the username from sip.conf into the first 4 fields, the secret into the password field and your asterisk ip into the fields that say voiptalk.org users.conf [207] username=207 transfer=yes mailbox=207 call-limit=2 fullname=mickey mouse registersip=no host=dynamic callgroup=1 context=default cid_number=207 hasvoicemail=yes vmsecret=1234 email=u...@yourpbx.com threewaycalling=yes hasdirectory=yes callwaiting=yes hasmanager=yes managerread=system,call,log,verbose,command,agent,user,config managerwrite=system,call,log,verbose,command,agent,user,config hasagent=yes hassip=yes hasiax=no secret=xxxxx nat=yes canreinvite=no dtmfmode=rfc2833 insecure=no pickupgroup=1 macaddress=00000000001170 autoprov=yes label=207 linenumber=1 disallow=all allow=ulaw,gsm sip.conf [207] type=peer context=phones host=dynamic fromuser=207 call-limit=3 secret=xxxxx canreinvite=yes directrtpsetup=no nat=yes qualify=yes register => 207:xx...@yourpbx.com/207 defaultip=1.2.3.4 mailbox=207 disallow=all allow=alaw _____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Thursday, July 09, 2009 1:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] setting up phones Can someone tell me how to setup a Aastra 75i phone? I have been trying to set it up and have pointed it to our asterisk server and selected http for download. What is the path? I have created two extension in asterisk for testing. I can't even get the phones to call each other. _____ Lauren found her dream laptop. Find the <http://www.microsoft.com/windows/choosepc/?ocid=ftp_val_wl_290> PC that's right for you. _____ Windows LiveT: 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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