[asterisk-users] setting up phones
Steve Totaro
stotaro at totarotechnologies.com
Fri Jul 10 11:42:43 CDT 2009
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 <sixfourimpala at 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 => 500:500 at 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 => 501:501 at 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: stotaro at first-notification.com
>
> To: asterisk-users at 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 <sixfourimpala at 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 '1e56b3345aabbe4373a46fb973476e72 at 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:asterisk at 10.0.0.52>;tag=as66b3ded8
> To: <sip:500 at 10.0.0.52>;tag=as66b3ded8
> Call-ID: 56cf67f56e3cba6602965f6317656c1a at 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:asterisk at 10.0.0.52>
> Accept: application/sdp
> Content-Length: 0
>
>
> ------------------------------
> From: danny at debsinc.com
> To: asterisk-users at 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-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Ott Rose
> *Sent:* Friday, July 10, 2009 10:33 AM
> *To:* asterisk-users at 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: danny at debsinc.com
> To: asterisk-users at 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:secret1 at 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:secret2 at 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-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Ott Rose
> *Sent:* Friday, July 10, 2009 8:33 AM
> *To:* asterisk-users at 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: stotaro at asteriskhelpdesk.com
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] setting up phones
>
> On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose <sixfourimpala at 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)
> ------------------------------
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>
>
>
> --
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>
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--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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