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g-man
New Contributor

SB.CALL 3 RTP resource unavailable or SDP negotiation failed. Call from

I am trying to pass calls between Provider and PBX via TA908. I have eth0/1 as public and eth0/2 as LAN. I can make outbound calls however when an inbound call arrives I am getting an error SB.CALL 3 RTP resource unavailable or SDP negotiation failed. Call from (xxxxxxxxxx) to (xxxxxxxxxx).

clock timezone -8

!

ip subnet-zero

ip classless

ip routing

ipv6 unicast-routing

!

!

name-server 209.18.47.61 209.18.47.62

!

!

no auto-config

!

!

service password-encryption

!

username

!

!

no ip firewall alg msn

no ip firewall alg mszone

no ip firewall alg h323

!

!

!

!

!

!

!

!

no dot11ap access-point-control

!

interface eth 0/1

  description (WAN)

  ip address  xxx.xxx.xxx.xxx  255.255.255.248

  no shutdown

  media-gateway ip primary

!

!

interface eth 0/2

  ip address   xxx.xxx.xxx.xxx  255.255.255.0

  no awcp

  no shutdown

  media-gateway ip primary

!

!

!

interface gigabit-eth 0/1

  no ip address

  shutdown

!

ip access-list standard mgmt-allow-list

  permit xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx

!

ip access-list standard sip-allow-list

  permit hostname provider.name.com

  permit xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx

!

!

!

!

!

!

ip route 0.0.0.0 0.0.0.0 xxx.xxx.xxx.xxx

!

no tftp server

no tftp server overwrite

http server

http secure-server

no snmp agent

no ip ftp server

no ip scp server

no ip sntp server

!

sip

sip udp 5060

no sip tcp

no sip tls

!

!

!

voice feature-mode network

voice forward-mode local

!

!

!

!

!

!

!

!

voice dial-plan 1 local NXX-XXXX

voice dial-plan 2 long-distance 1-NXX-XXX-XXXX

!

!

!

!

voice codec-list VOICE

  default

  codec g711ulaw

!

!

!

voice trunk T01 type sip

description "SIP"

  caller-id-override emergency-outbound xxxxxxxxxx

  sip-server primary xxx.xxx.xxx.xxx

  registrar primary xxx.xxx.xxx.xxx

  domain "public IP"

  register 5555555555 auth-name "xxx" password "xxx"

  codec-list VOICE both

  authentication username "xxx" password "xxx"

!

voice trunk T11 type sip

  description "PBX"

  sip-server primary PBX LAN ADDRESS

  grammar from host local

  transfer-mode network

!

!

voice grouped-trunk PROVIDER

  trunk T01

  accept NXX-XXX-XXX cost 0

  accept 1-NXX-NXX-XXXX cost 0

  accept 011-$ cost 0

  accept 411 cost 0

  accept 911 cost 0

!

!

voice grouped-trunk PBX

  trunk T11

  accept XXXXXXXXXX cost 0

  accept XXXXXXXXXX cost 0

!

!

sip access-class ip "sip-allow-list" in

!

line con 0

  no login

!

line telnet 0 4

  login local-userlist

  no shutdown

  ip access-class mgmt-allow-list in

line ssh 0 4

  login local-userlist

  no shutdown

  ip access-class mgmt-allow-list in

!END

ADTRAN, Inc. OS version R12.3.0.SA.E

  Mainline Version: ENM.16.126

  P4 Changelist: 633842

  Checksum: 30e472eeaf64e5d8d7d680446c0c1f79ec2114393ce495716255c5e039148abf

  Built on: Tue Jan 24 17:12:35 CST 2017

Hardware version M.1.5.0.0-0

Boot ROM version R10.9.3.B1

  Built on: Fri Apr 11 09:42:09 CDT 2014

  Compatibility Version: 1

Copyright (c) 1999-2017, ADTRAN, Inc.

Platform: Total Access 908e (3rd Gen), part number 4243908F1

Flash: 97120256 bytes  DRAM: 509648896 bytes

uptime is 0 days, 0 hours, 3 minutes, 22 seconds

slot 0, DSP 1

  DSP software version: G3.R1.4..2-R

  DSP hardware version: Freescale BSC9131

  Total channels: 60

Labels (1)
0 Kudos
1 Reply
jayh
Honored Contributor
Honored Contributor

Re: SB.CALL 3 RTP resource unavailable or SDP negotiation failed. Call from

Your firmware is ancient. I haven't read through all of the release notes to see if this is a known bug, but upgrade to the latest extended maintenance release.

Also, some configurations of SIP-to-SIP require that you have an SBC license now, and that's a possibility.

A "debug sip stack messages" capture on a failed call could be useful.