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Anonymous
Not applicable

SIP to PRI

Looking for help with setting up our TA908 to provide SIP - PRI. Have the SIP side responding to Options pkts from my SBC and Binding is clean in my Metaswitch. The PRI is up to my Exfo test set clean.

I can complete and inbound call from cell phone to Exfo but I get no audio. an outbound call from Exfo dies at the PRI - SIP handoff. I know I have to be missing something between the two services within the 908. Thanks for the help...

hostname "TA908"

no enable password

!

!

ip subnet-zero

ip classless

ip routing

!

!

!

!

no auto-config

!

event-history on

no logging forwarding

no logging email

!

no service password-encryption

!

!

no ip firewall alg msn

no ip firewall alg mszone

no ip firewall alg h323

!

!

!

!

!

no dot11ap access-point-control

!

!

!

!

!

!

!

!

!

!

!

!

!

interface eth 0/1

  ip address  172.XX.XX.X  255.255.255.248

  media-gateway ip primary

  no shutdown

!

!

!

!

interface t1 0/1

  shutdown

!

interface t1 0/2

  description PRI to PBX

  tdm-group 1 timeslots 1-24 speed 64

  no shutdown

!

!

interface pri 1

  description PRI to PBX

  isdn name-delivery setup

  connect t1 0/2 tdm-group 1

  role network b-channel-restarts disable

  no shutdown

!

!

interface fxs 0/1

  no shutdown

!

interface fxs 0/2

  no shutdown

!

interface fxs 0/3

  no shutdown

!

interface fxs 0/4

  no shutdown

!

interface fxs 0/5

  no shutdown

!

interface fxs 0/6

  no shutdown

!

interface fxs 0/7

  no shutdown

!

interface fxs 0/8

  no shutdown

!

!

isdn-group 1

  connect pri 1

!

!

!

!

timing-source t1 0/2

!

!

!

!

!

!

!

ip route 0.0.0.0 0.0.0.0 172.XX.XX.X

!

no ip tftp server

no ip tftp server overwrite

no ip http server

no ip http secure-server

no ip snmp agent

no ip ftp server

no ip scp server

no ip sntp server

!

!

!

!

!

!

ip sip

ip sip udp 5060

no ip sip tcp

!

!

!

voice feature-mode network

voice forward-mode network

!

!

!

!

!

!

!

!

!

!

!

voice codec-list trunk

  codec g729

  codec g711ulaw

!

!

!

voice trunk T01 type sip

  outbound-proxy primary 172.XX.X.X

  codec-group trunk

!

voice trunk T02 type isdn

  description "PRI to PBX"

  resource-selection linear ascending

  connect isdn-group 1

  rtp delay-mode adaptive

!

!

voice grouped-trunk SIP

  trunk T01

  accept $ cost 0

!

!

voice grouped-trunk PRI

  trunk T02

  accept $ cost 0

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

line con 0

  no login

!

line telnet 0 4

  login

  no shutdown

line ssh 0 4

  login local-userlist

  no shutdown

!

!

!

!

!

end

TA908#

Labels (2)
0 Kudos
4 Replies
tszala
New Contributor

Re: SIP to PRI

Can you do a packet capture?  I've seen RTP Audio issues are generally a IP / NAT issue.   Config appears to be ok, not sure which generation you're running, but I always run my PRI's off T-1 interface 3 or 4. 

ftwrobert
New Contributor III

Re: SIP to PRI

The config only shows two t1 interfaces, which makes me think this is a non- e TA900 unit. T1 0/2 would be the only available interface to run a PRI.     

ftwrobert
New Contributor III

Re: SIP to PRI

I'd be interested in seeing the configuration you've got on the Metaswitch, in order to get a full grasp of how things are programmed. It looks like you're connected on the same, local subnet? (172.xxx.xxx.xxx)

Here's the TA900 configuration I use, well, the important parts.

voice trunk T00 type sip

description "SBC1 - SIP to PRI/POTS"

sip-server primary <metaswitch_IP>

registrar primary <metaswitch_IP>

register <sip_username> auth-name <sip_username> password <sip_password>

authentication username <sip_username> password <sip_password>

interface t1 <interface>

tdm-group 1 timeslots 1-24 speed 64

codec-group G711u

no shutdown

exit

interface pri 1

isdn name-delivery setup

connect t1 <interface> tdm-group 1

digits-transferred <0,3,4,7,all>

no shutdown

exit

isdn-group 1

connect pri 1

exit

voice trunk T01 type isdn

description “PBX - SIP to PRI”

resource-selection linear descending

connect isdn-group 1

rtp delay-mode adaptive

exit

voice grouped-trunk PRI

trunk T01

accept $

exit

write

exit

Anonymous
Not applicable

Re: SIP to PRI

ftwrobert, thanks for the reply....we built a /29 network for testing this service. The contact IP 172.22.24.X and outside interface of my Perimeta SBC 172.22.1.X which in turn points to the inside CFS 172.20.1.XXX.

  Name                                            901TEST

  Usage                                           Subscriber

  LearnsContactDetails                            False

  DelegatedManagementGroup                        Ems.0.default/MVConfigDBConn/DelegatedManagementGroupContainer/DelegatedManagementGroup.0 //default

  UseDNForIdentification                          True

  SIPAuthenticationRequired                       False

  SIPDomainName                                   sbc.imon.net

  IPAddressMatchRequired                          False

  ContactIPAddress                                172.22.24.X

  ContactIPPort                                   5060

  SupportedIncomingTrunkGroupParameterType        None

  TrunkGroupParameterTypeOnOutgoingMessages       None

  ProxyIPAddress                                  172.20.1.XXX

  ProxyIPPort                                     5060

  TransportProtocol                               UDP

  MediaGatewayModel                               ../MediaGatewayModelContainer/MediaGatewayModel.133 //Remote Media Gateway Model "SIP Trunk Business"

  NetworkNode                                     Use default

    NetworkNodeDefaultValue                         None

    NetworkNodeSpecificValue                        None

  SIPBindingLocation                              None

  ESAProtectionDomain                             None

  Trusted                                         True

  UseCallerNameProvidedBySIPDevice                False

  PlayAnnouncementsWhenErrorConditionsOccur       True

  UseStaticNATMapping                             False

  MaximumCallAppearances                          10

  MaximumConcurrentHighBandwidthCallAppearancesAllowed 0

  PollPeerDevice                                  True

  PollingInterval                                 30

  ConcurrentNumberOfCallAppearancesInUse          0

  ConcurrentNumberOfHighBandwidthCallAppearancesInUse 0

  DeactivationMode                                Normal

  LastCallFailure                                 Last call failure

  LastCallFailureCause                            Subscriber not contactable

  LastCallFailureTimestamp                        7/1/15 10:45:09 AM

  LastCallFailureLogCorrelator                    0429 01d8 5fd4 1c9b