Bind to 0.0.0.0 put your phones on DHCP if they are not already and reboot.
reload asterisk. turn on sip debugging call 501 from 500 post debug info. i bet it rings. On Fri, Jul 10, 2009 at 12:38 PM, Ott Rose <sixfourimp...@hotmail.com>wrote: > Here is my conf files. > > sip.conf > [general] > context=default > port=5060 ; UDP port for Asterisk > bindaddr=10.0.0.52; If we want to specify only an IP (if a computer has > three different IPs) 0.0.0.0 means any IP > srvlookup=yes ; Enable DNS SRV server > > > > > [500] > type=peer > context=phones > host=dynamic > fromuser=500 > secret=500 > 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 => 500:5...@10.0.0.52/500 > defaultip=10.0.0.60 > mailbox=1001 > disallow=all > allow=alaw > > [501] > type=peer > context=phones > host=dynamic > fromuser=501 > secret=501 > 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 => 501:5...@10.0.0.52/501 > defaultip=10.0.0.46 > mailbox=1002 > disallow=all > allow=alaw > > ====================================================================== > users.conf > [501] > username = 501 > transfer = yes > mailbox = 501 > call-limit = 100 > type = peer > fullname = 501 > registersip = yes > > 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 = yes > > 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 > disallow = all > allow = ulaw,gsm > > > > ------------------------------ > Date: Fri, 10 Jul 2009 12:16:50 -0400 > From: stot...@first-notification.com > > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] setting up phones > > Change the address in sip.conf, not the phone. > > On Fri, Jul 10, 2009 at 12:04 PM, Ott Rose <sixfourimp...@hotmail.com>wrote: > > 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 <http://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 <http://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> > > ------------------------------ > Windows Live™ SkyDrive™: 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> > > _______________________________________________ > -- 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) > > ------------------------------ > 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)
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