RTP resource unavailable or SDP negotiation failed

[bou401]

running Total Access TA908e 

Sip to PRI incoming - No Issues.

PRI to SIP cannot call out

RTP resource unavailable or SDP negotiation failed 

tried to move a DID to voice user and add coverage external to my cell and get the same error when called

sh RTP resources indicates 60 channels available ( could this still be Bad ? )

Thanks in advance

[bou401]

config...

hostname "Company Name"
enable password encrypted 41489bd142f1c82398c5390cfc2710b1128c
!
!
clock timezone -6-Central-Time
!
ip subnet-zero
ip classless
ip default-gateway 10.10.30.1
ip routing
no ipv6 unicast-routing
!
!
domain-proxy
!
!
no auto-config
auto-config authname adtran encrypted password 4543041fd4c975a3129cddcc45523a95ebcb
!
event-history on
no logging forwarding
no logging email
!
service password-encryption
!
username "admin" password encrypted "2b2c88f09b8ae8c2141cf745d72e2e28cc99"
username "Provideradmin" password encrypted "4060e8f842ca46fecbc344f3d594879a50e5ef676da6129b8a6f2c714266e1c9c943"
!
!
ip firewall
no ip firewall alg msn
no ip firewall alg mszone
no ip firewall alg h323
!
!
!
!
!
!
!
!
no dot11ap access-point-control
!
!
!
!
!
!
!
ip dhcp pool "Private"
network 10.10.10.0 255.255.255.0
netbios-node-type h-node
default-router 10.10.10.1
!
!
!
!
!
!
!
!
!
!
!
qos map ConfigWizardQoSMap 20
match dscp 46
match ip rtp 5060 10048 all
priority percent 10
set dscp 46
!
!
!
!
interface eth 0/1
description Connection to Co Address
ip address 10.10.30.27 255.255.255.0
no shutdown
media-gateway ip primary
!
!
interface eth 0/2
no ip address
shutdown
!
!
!
interface gigabit-eth 0/1
no ip address
shutdown
!
!
!
!
interface t1 0/1
shutdown
!
interface t1 0/2
shutdown
!
interface t1 0/3
shutdown
!
interface t1 0/4
description PRI to Co PBX
tdm-group 1 timeslots 1-24 speed 64
no shutdown
!
!
interface pri 1
description PRI to Co PBX
role network b-channel-restarts enable
isdn name-delivery setup
connect t1 0/4 tdm-group 1
no shutdown
!
!
interface fxs 0/1
shutdown
!
interface fxs 0/2
shutdown
!
interface fxs 0/3
shutdown
!
interface fxs 0/4
shutdown
!
interface fxs 0/5
shutdown
!
interface fxs 0/6
shutdown
!
interface fxs 0/7
shutdown
!
interface fxs 0/8
shutdown
!
!
interface fxo 0/0
shutdown
!
!
isdn-group 2
connect pri 1
!
!
isdn-group 3
!
!
!
!
timing-source t1 0/4
!
!
!
!
ip access-list standard Provider
permit xx.xxx.xxx.0 0.0.0.255
permit xx.xxx.196.0 0.0.0.255
permit host xx.xxx.106.252
!
ip access-list standard Provider-NOC
permit xx.xxx.196.0 0.0.3.255
permit xx.xxx.xxx.0 0.0.7.255
permit xx.xxx.184.0 0.0.3.255
permit host xx.xxx.254.195
!
ip access-list standard wizard-ics
remark Internet Connection Sharing
permit any
!
!
ip access-list extended Management
permit ip any host 10.10.30.27 log
!
ip access-list extended self
remark Traffic to Total Access
permit ip any any log
!
ip access-list extended web-acl-10
remark IPVPN
permit udp any host 10.10.30.27 eq isakmp log
!
ip access-list extended web-acl-11
remark IpVPN
!
ip access-list extended web-acl-4
remark Admin access
permit tcp any any eq www log
permit icmp any any echo log
permit udp any any eq snmp
permit 89 any any
permit tcp xx.xxx.196.0 0.0.0.255 any eq telnet log
permit tcp xx.xxx.xxx.0 0.0.0.255 any eq telnet log
permit tcp xx.xxx.196.0 0.0.0.255 any eq ssh log
permit tcp xx.xxx.xxx.0 0.0.0.255 any eq ssh log
deny tcp any any eq ssh log
deny tcp any any eq telnet log
permit tcp xx.xxx.xxx.0 0.0.0.255 any eq https log
deny tcp any any eq https log
!
ip access-list extended web-acl-5
remark RTP
permit udp any any range 16384 32767
!
ip access-list extended web-acl-6
remark SIP
permit udp host xxx.xxx.xxx.130 any eq 5060
permit udp host xx.xxx.xxx.13 any eq 5060
!
!
!
!
ip policy-class Private
allow list self self stateless
!
ip policy-class Public
allow list web-acl-4 self
allow list web-acl-5 self
allow list web-acl-6 self
!
!
!
ip route 0.0.0.0 0.0.0.0 10.10.30.1
!
no tftp server
no tftp server overwrite
http server
http secure-server
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 quality-stats history max-streams 100
voice transfer-mode local
voice forward-mode local
voice call-appearance-mode single
voice modem-passthrough-mode inbound
voice conferencing-mode local
!
!
!
!
!
!
!
!
voice dial-plan 1 local NXX-XXXX
voice dial-plan 2 local xxx-NXX-XXXX
voice dial-plan 3 long-distance 1-XXX-XXX-XXXX
voice dial-plan 4 extensions npanxx2081
!!
!
!
voice codec-list "G711 - G729"
default
codec g711ulaw
codec g729
!
!
voice trunk T04 type isdn
description "PRI to Co PBX"
resource-selection circular descending
caller-id-override number-inbound npanxx2072 if-no-cpn
connect isdn-group 2
no early-cut-through
modem-passthrough detection-time 1
t38
t38 redundancy high-speed 1
t38 redundancy low-speed 1
no echo-cancellation
rtp delay-mode adaptive
rtp packet-delay maximum 200
rtp qos dscp 46
rtp dtmf-relay inband
codec-list "G711 - G729"
!
voice trunk T11 type sip
description "SIP trunk to Provider_Perimeta"
sip-server primary xxx.xxx.xxx.130
registrar primary xxx.xxx.xxx.130
registrar threshold absolute 5
domain "xxx.xxx.xxx.130"
dial-string source to
sip-keep-alive options 30
register npanxx2072 auth-name "npanxx2072" password encrypted "3f37822a38c8358a37ead2dbd1042cd80022"
codec-list "G711 - G729" both
grammar from host sip-server
grammar supported 100rel
authentication username "npanxx2072" password encrypted "252d94463e23a70cbca2834f6a8f51cf3304"
!
!
voice grouped-trunk "SIP TO Provider"
description "Provider SIP Trunk"
trunk T11
accept NXX-NXX-XXXX cost 0
accept 1-NXX-NXX-XXXX cost 0
accept 1-800-NXX-XXXX cost 0
accept 1-888-NXX-XXXX cost 0
accept 1-877-NXX-XXXX cost 0
accept 1-866-NXX-XXXX cost 0
accept 1-855-NXX-XXXX cost 0
accept 011-$ cost 0
accept 0-NXX-NXX-XXXX cost 0
accept NXX-XXXX cost 0
accept 411 cost 0
accept 611 cost 0
accept 911 cost 0
reject NXX-976-XXXX
reject 1-900-NXX-XXXX
reject 1-976-NXX-XXXX
reject 976-XXXX
!
!
voice grouped-trunk "PRI TO SYSTEM"
description "Co PBX PRI Trunk"
trunk T04
accept npanxx2072 cost 0
accept npanxx2073 cost 0
!
!
!
!
!
!
voice user npanxx2081
no cos
password encrypted "1c18c8833690fd7eb7cd7af7485d29c9883c"
coverage external npanxx6291
sip-authentication password encrypted "2723ef1bace523e88dc1839aa58a7da49d2f"
codec-list "G711 - G729"
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
sip grammar from host domain
no sip grammar supported 100rel
!
!
!
!
!
!
ip rtp quality-monitoring
ip rtp quality-monitoring udp
ip rtp quality-monitoring sip
!

[geo]

Hello bou401 and thanks for posting.

In short, if you are successful SIP to PRI, then your DSP is good.  I would enable these debugs, make an outbound call, and open a ticket with Support, unless you want to scrub some of the sensitive info:

debug sip stack message

debug sip cldu

debug isdn l2-for

debug voice verbose

We need to throw all these debugs at it in order to see where and why the outbound calls are failing.  

Was this working previously and just stopped?

Thanks!

Geo

[bou401]

Thanks for the follow-up. Unfortunately contacting support is not an option in this case. No Service contract on this unit. Attached is the basic information pertaining to this config and debug to voice user which attempts to call out but fails

This unit is currently waiting to be deployed and not currently in service. 

 Registered sip binding (voice Trunk T11) currently registered to Sip Server SBC

Incoming calls to PRI are fine

Outgoing calls fail

Ive created a voice user to avoid having the end user continuously testing outgoing calls for us.

this voice user has call coverage to an outside number which also fails with same result as PRI

see attached for basic config and debugs

[geo]

Hello bou401,

It is not quite the same test doing it that way.  If you can get the customer to let you do another test call with the actual call flow, with the debugs I suggested, I could better assist you.

Thanks,

Geo

[bou401]

failed Call from PRI. Debugs attached

[geo]

Hello bou401,

Something odd is going on for sure.  You might need to add:

debug sip cldu 

and make the test call again.  The switchboard routing looks add (unusually high score for a good match) and there might be more going on behind the scenes here.

Thanks!

Geo

[bou401]

These were the debugs running at the time of the test call

debug sip stack message

debug sip cldu

debug isdn l2-for

debug voice verbose

[geo]

You are right I noticed that now.  What version of AOS voice is device running on?

Geo

[bou401]

"T900E3A-R14-4-0-E.biz"

Platform: Total Access 908e (3rd Gen), part number 4243908F5

[bou401]

btw. Ive also loaded the same config on another TA908e at our offices and have the same outcome.

[geo]

Hello bou401,

Clearly there is something odd going on.  Without a service plan, troubleshooting is very limited on the forum.  There is some redacted info (for obvious security concerns) that makes this process harder.  Also, there are a couple of configuration options that are not best practice or supported:

1) change voice call-appearance-mode single to multiple

2) remove the alpha numeric dial-plan entries which are invalid

Here is what I would do if I was in your place, I would start with a barebone config.  Do not use the web gui to configure, use the CLI.  Don't change any voice config settings to anything other than default to start.  Add your config for ethernet and T1 PRI.  Add your voice trunks with ONLY the necessary config.  Then add your voice grouped-trunks with ONLY the necessary accepts. For example, these are not all needed, that is what the web gui will spit out by default and it is mainly for use on the NV 7100:

accept NXX-NXX-XXXX cost 0        <<----this overlaps with
accept 1-NXX-NXX-XXXX cost 0
accept 1-800-NXX-XXXX cost 0
accept 1-888-NXX-XXXX cost 0
accept 1-877-NXX-XXXX cost 0
accept 1-866-NXX-XXXX cost 0
accept 1-855-NXX-XXXX cost 0
accept 011-$ cost 0
accept 0-NXX-NXX-XXXX cost 0
accept NXX-XXXX cost 0             <<----this
accept 411 cost 0
accept 611 cost 0
accept 911 cost 0
reject NXX-976-XXXX
reject 1-900-NXX-XXXX
reject 1-976-NXX-XXXX
reject 976-XXXX

Just use accept 1-NXX-NXX-XXXX cost 0 to start.  You can add more specific config options later if necessary.  Test SIP to PRI and PRI to SIP in your office where you tested before.  Once you can call in both directions, try the same config for the customer, verify calling in both directions, then add other config options if needed.

Regards,

Geo  

[bou401]

Ive made some changes to my access list because natting was involved and i see call going out of the adtran to sip. However im getting a SIP/2.0 403 From: URI not recognized message on the voice server side. Can this be changed as to how the adtran sends the Invite

Thanks

[bou401]

I figured out my issues. Thanks anyway