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multipl3x
New Contributor II

I cant get SIP phones with private IP to work

I've been doing mostly SIP provider to FXS or PRI but today I tried to register both a softphone and a physical sip phone to the adtran and place calls but it does not work. The phone registers and the call completes but I get one way audio and I don't get ringback.

I have media-gateway ip primary on both interfaces, WAN with public IP and LAN with RFC1918.

I have tried to set up NAT like in this thread: Configure 908e as Router with DHCP.

but it made no difference.

what am i doing wrong???

0 Kudos
2 Replies
jayh
Honored Contributor
Honored Contributor

Re: I cant get SIP phones with private IP to work

Are the phones registering to the Adtran or passing through the 908e and registering to the provider directly? If this is the case, try:

ip sip

ip sip proxy

ip sip proxy transparent

Some versions of firmware omit the "ip", so if that errors, try:

sip

sip proxy

sip proxy transparent


You won't need any specific configuration on the 908e for each phone, it will fix up the SIP and the phone will register directly with the provider.

If you want the SIP phones to register to the Adtran, then the configuration is substantially more complex and may not be supported by your provider without some tweaking.

multipl3x
New Contributor II

Re: I cant get SIP phones with private IP to work

I suppose I am putting the cart before the horse so to speak. I have started verifying my NAT configuration is correct, as it is not what I am used to. Eventually I was able to make a call. Here is the configuration in case any one encounters the same frustration I had:

LAB-ADTRAN-908e#sh run int ethernet 0/1

Building configuration...

!

!

interface eth 0/1

  ip address  xxx.xxx.xxx.xxxx  255.255.255.224

  access-policy REMOTE_ACCESS

  ip access-group WAN_SEC_IN in

  media-gateway ip primary

  no shutdown

!

end

LAB-ADTRAN-908e#sh run int ethernet 0/2

Building configuration...

!

!

interface eth 0/2

  description ADTRAN LAN

  ip address  172.16.8.1  255.255.255.224

  access-policy NAT_INSIDE

  media-gateway ip primary

  no shutdown

!

end

LAB-ADTRAN-908e#sh run | b ip polic

ip policy-class NAT_INSIDE

  nat source list ALLOW-ALL interface eth 0/1 overload

  allow list ALLOW-ALL

  allow list self

!

ip policy-class REMOTE_ACCESS

  allow list TRUSTED_REMOTE_ACCESS

  allow list self

LAB-ADTRAN-908e#sh run | inc ip sip

ip sip

ip sip udp 5060

ip sip tcp 5060

ip sip default-call-routing switchboard

ip sip registrar

ip sip registrar realm 172.16.8.1

ip sip proxy

ip sip proxy transparent

==========================================================================================

The outstanding question:


==========================================================================================

I can see what this configuration is doing. It is a proxy. I have never done with before, and would prefer not to. I would prefer the Adtran 908 serves as a PBX, where internal LAN extensions are translated to share the same outgoing number. I am coming from CME world. In CME world this is what I commonly do:

voice translation-rule 1

rule 1 /.*/ /xxxxxxxxxx/ <= ASSIGNED NUMBER FROM METASWITCH GROUP

!

voice translation-rule 2

rule 1 /.*/ /299/ <=HUNT PILOT NUMBER

voice translation-profile META-TRUNK1-INBOUND

translate called 2

!

voice translation-profile META-TRUNK1-OUTBOUND

translate calling 1

dial-peer voice 100 voip

description OUTBOUND TO META GLOBAL DIAL PEER

translation-profile outgoing META-TRUNK1-OUTBOUND

preference 2

destination-pattern [1-9]..[1-9]......

session protocol sipv2

session target ipv4:xxx.xxx.xxx.xxx <=IP ADDRESS OF METASWITCH SBC

session transport udp

voice-class codec 1

voice-class sip bind control source-interface FastEthernet0/0

voice-class sip bind media source-interface FastEthernet0/0

dtmf-relay rtp-nte sip-notify

no vad

dial-peer voice 2000 voip

description INBOUND FROM META - TRUNK 2

translation-profile incoming META-TRUNK2-INBOUND

session protocol sipv2

session target ipv4:xxx.xxx.xxx.xxx <=IP ADDRESS OF METASWITCH SBC

session transport udp

incoming called-number xxxxxxxxxx <= ASSIGNED NUMBER FROM METASWITCH GROUP

voice-class codec 1

voice-class sip bind control source-interface FastEthernet0/0

voice-class sip bind media source-interface FastEthernet0/0

dtmf-relay rtp-nte sip-notify

no vad