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
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.
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.