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New Contributor

PRI transfer to analog line failing

Hi! I have a 908 2nd gen that is handling a fiber PRI to another vendors IVR and another vendors PBX. the IVR accepts numbers passed through the PRI, and we only have 1 number from the PRI actually using this connection, the rest are set for analog FXS ports that go to the PBX, the IVR and PBX do not interact.

My issue right now, is when a call hits the IVR, goes through all its messages, it is suppose to transfer the call to a specific number that is handled through an FXS port. and if another call comes in, the ISP is handling call forward busy if someone is on that specific FXS port, which then rings a different fxs port. Unfortunately this seems to happen intermittently. with most calls dropping. the ISP shows their monitor as the call was ending as it was expected and getting an OK for the call with no issues.

I want to make sure my config for the 908 wont have any issues with the IVR asking to transfer the call to the fxs port numbers. the 900 takes in a full 10 digit request from the IVR and strips it down to 4 when delivering it to the analog ports.

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no dot11ap access-point-control

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interface eth 0/1

  description WAN to XXXXXXXXXXX NTU

  ip address  XXXXXXXXXXX  XXXXXXXXXXX

  media-gateway ip primary

  no shutdown

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interface t1 0/1

  description PRI

  tdm-group 1 timeslots 1-12,24 speed 64

  shutdown

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interface t1 0/2

  description PRI

  tdm-group 1 timeslots 1-12,24 speed 64

  no shutdown

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interface pri 1

  isdn name-delivery setup

  connect t1 0/2 tdm-group 1

  no shutdown

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interface fxs 0/1

  no shutdown

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interface fxs 0/2

  no shutdown

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interface fxs 0/3

  no shutdown

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interface fxs 0/4

  no shutdown

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interface fxs 0/5

  no shutdown

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interface fxs 0/6

  no shutdown

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interface fxs 0/7

  no shutdown

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interface fxs 0/8

  no shutdown

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isdn-group 1

  connect pri 1

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ip route XXXXXXXXXXX

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no tftp server

no tftp server overwrite

no http server

no http secure-server

no snmp agent

no ip ftp server

no ip scp server

no ip sntp server

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sip

sip udp XXXXXXXXXXX

no sip tcp

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voice feature-mode network

voice quality-stats history max-streams 100

voice forward-mode network

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voice trunk T01 type sip

  sip-server primary XXXXXXXXXXX

  domain "XXXXXXXXXXX"

  register XXXXXXXXXXX auth-name "XXXXXXXXXXX" password "XXXXXXXXXXX"

  register XXXXXXXXXXX auth-name "XXXXXXXXXXX" password "XXXXXXXXXXX"

  register XXXXXXXXXXX auth-name "XXXXXXXXXXX" password "XXXXXXXXXXX"

  register XXXXXXXXXXX auth-name "XXXXXXXXXXX" password "XXXXXXXXXXX"

  register XXXXXXXXXXX auth-name "XXXXXXXXXXX" password "XXXXXXXXXXX"

  register XXXXXXXXXXX auth-name "XXXXXXXXXXX" password "XXXXXXXXXXX"

  register XXXXXXXXXXX auth-name "XXXXXXXXXXX" password "XXXXXXXXXXX"

  register XXXXXXXXXXX auth-name "XXXXXXXXXXX" password "XXXXXXXXXXX"

  register XXXXXXXXXXX auth-name "XXXXXXXXXXX" password "XXXXXXXXXXX"

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voice trunk T51 type isdn

  description "to PBX"

  resource-selection linear ascending

  caller-id-override number-inbound XXXXXXXXXXX

  connect isdn-group 1

  match dnis "XXXXXX$" substitute "$"

  modem-passthrough detection-time 120

  rtp delay-mode fixed

  rtp packet-delay nominal 80

  rtp packet-delay fax 80

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voice grouped-trunk CPESIPTG

  trunk T01

  accept $ cost 0

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voice grouped-trunk PRI

  description "PRI"

  trunk T51

  accept $ cost 1

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voice user 1001

  connect fxs 0/1

  password "XXXXXXXXXXX"

  did "XXXXXXXXXXX"

  sip-identity XXXXXXXXXXX T01

  modem-passthrough

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voice user 1002

  connect fxs 0/2

  password "XXXXXXXXXXX"

  did "XXXXXXXXXXX"

  sip-identity XXXXXXXXXXX T01

  modem-passthrough

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voice user 1003

  connect fxs 0/3

  password "XXXX"

  did "XXXXXXXXXXX"

  sip-identity XXXXXXXXXXX T01

  modem-passthrough

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voice user 1004

  connect fxs 0/4

  password "XXXX"

  did "XXXXXXXXXXX"

  sip-identity XXXXXXXXXXX T01

  modem-passthrough

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voice user 1005

  connect fxs 0/5

  password "XXXXXXXXXXX"

  did "XXXXXXXXXXX"

  sip-identity XXXXXXXXXXX T01

  modem-passthrough

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voice user 1006

  connect fxs 0/6

  password "XXXXXXXXXXX"

  did "XXXXXXXXXXX"

  sip-identity XXXXXXXXXXX T01

  modem-passthrough

  no echo-cancellation

Labels (3)
0 Kudos
2 Replies
Honored Contributor
Honored Contributor

Re: PRI transfer to analog line failing

The problem may be with your provider's implementation of call-forward-busy. You could try building a ring-group and putting the FXS extensions within the ring group, then have the provider point the calls to the ring group. This sounds like a series-completion problem within your provider.

Highlighted
New Contributor

Re: PRI transfer to analog line failing

Finally found the issue.

voice trunk T51 type isdn

  description "to PBX"

  resource-selection linear ascending

  caller-id-override number-inbound XXXXXXXXXXX

the reason we had to add this is the IVR is too old to send any sip commands, because of this it could not send a number to charge for the call. we added the caller-id-override to work around the fact the IVR cannot send us a charge number, but we ended up using the first line in their analog port numbers for the override.

So, with it set this way, if the first number was available, it would charge itself for the call and the call would drop. if it was busy, the transfers would work fine. but any time the first number was not busy, any transfers would fail.

To fix this we changed the caller-id-override to the IVR number that isnt on one of the analog ports, no number on their system will ever call this number, so it will never charge itself, which means the calls will not fail.