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no ring back tone with IX messaging

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youthtelco

IS-IT--Management
Apr 28, 2022
12
0
1
US
My call scenario is this:

A caller from outside is dialing number, which is coming to SIP trunk of SBC -> Avaya -> IX messaging (play Auto Attendance announcement, press 1, Send the phone number x1000) -> x1000 station on Avaya CM -> x1000 setting with CFW to Cell phone 999-200-5000

The issue is a initial caller cannot hear any ring back, like dead air, but at the same time a cell phone is ringing.
Then once a cell phone answers the call, A caller and a callee can hear each other.

Anyone give some idea about this issue? Every other call and cfw is working fine without IX messaging server. I feel that this appears only with IX messaging server.

09:01:05 SIP<INVITE sip:1000@google.com SIP/2.0
09:01:05 Call-ID: 3be95048fe554c3c3280f75e75a16977
09:01:05 SIP>SIP/2.0 181 Call Is Being Forwarded <---------------------------------------------- from here, dead air
09:01:05 Call-ID: 3be95048fe554c3c3280f75e75a16977
09:01:05 Calling party trunk-group 910 member 98 cid 0x871
09:01:05 Calling Number & Name +19992005000 NO-CPName
09:01:05 term station 1000 cid 0x871
09:01:05 call-forwarding 1999
09:01:05 term trunk-group 997 cid 0x871
09:01:05 call-forwarding 19992005000
09:01:05 route-pattern 2000 preference 1 location 1 cid 0x871
09:01:05 seize trunk-group 997 member 81 cid 0x871
09:01:05 Calling Number & Name +19993334000 NO-CPName
09:01:05 G711MU ss:eek:ff ps:20 rgn:242 [192.168.64.26]:35370 rgn:2 [192.169.64.21]:2126
09:01:05 xoip options: fax:Relay modem:eek:ff tty:US uid:0x500350 xoip ip: [192.169.64.21]:2126
09:01:05 INVITE going via Sig Grp 910, ignoring called number routing
09:01:05 SIP>INVITE sip:19992005000@google.com;user=phone SIP/2.0
09:01:05 Call-ID: 33ba4aaaa3cb41edac1d0c29df7964
09:01:05 Setup digits 19992005000
09:01:05 Calling Number & Name +19993334000 NO-CPName
09:01:05 SIP<SIP/2.0 100 Trying
09:01:05 Call-ID: 33ba4aaaa3cb41edac1d0c29df7964
09:01:05 Proceed trunk-group 997 member 81 cid 0x871
09:01:06 SIP<PRACK sip:+19993334000@192.168.64.5;transport=tcp;asm=6;
09:01:06 SIP<x-b2b-refer-hand-send-hold SIP/2.0
09:01:06 Call-ID: 3be95048fe554c3c3280f75e75a16977
09:01:06 SIP>SIP/2.0 200 OK
09:01:06 Call-ID: 3be95048fe554c3c3280f75e75a16977
09:01:06 SIP>SIP/2.0 183 Session Progress
09:01:06 Call-ID: 3be95048fe554c3c3280f75e75a16977
09:01:06 SIP<PRACK sip:+19993334000@192.168.64.5;transport=tcp;asm=6;
09:01:06 SIP<x-b2b-refer-hand-send-hold SIP/2.0
09:01:06 Call-ID: 3be95048fe554c3c3280f75e75a16977
09:01:06 SIP>SIP/2.0 200 OK
09:01:06 Call-ID: 3be95048fe554c3c3280f75e75a16977
09:01:08 SIP<SIP/2.0 180 Ringing
09:01:08 Call-ID: 33ba4aaaa3cb41edac1d0c29df7964
09:01:08 SIP>UPDATE sip:+19993334000@192.168.64.26:5060;transport=tcp
09:01:08 SIP>;gsid=e0b1b510-a3c9-11ed-90a5-000c292857e3;asm=3 SIP/2.
09:01:08 SIP>0
09:01:08 Call-ID: 3be95048fe554c3c3280f75e75a16977
09:01:08 SIP>SIP/2.0 180 Ringing
09:01:08 Call-ID: 3be95048fe554c3c3280f75e75a16977
09:01:08 Alert trunk-group 997 member 81 cid 0x871
09:01:08 G711MU ss:eek:ff ps:20 rgn:242 [192.168.64.26]:39644 rgn:2 [192.169.64.21]:2080
09:01:08 xoip options: fax:Relay modem:eek:ff tty:US uid:0x50014b xoip ip: [192.169.64.21]:2080
09:01:08 Alerting party uses public-unknown-numbering
09:01:08 SIP<SIP/2.0 200 OK
09:01:08 Call-ID: 3be95048fe554c3c3280f75e75a16977
09:01:08 SIP>SIP/2.0 180 Ringing
09:01:08 Call-ID: 3be95048fe554c3c3280f75e75a16977
09:01:08 SIP<PRACK sip:19992005000@192.168.64.5;transport=tcp;asm=6;x
09:01:08 SIP<-b2b-refer-hand-send-hold SIP/2.0
09:01:08 Call-ID: 3be95048fe554c3c3280f75e75a16977
09:01:08 SIP>SIP/2.0 200 OK <-----------------------------------------------------------------------------Call connect, sounds OK
09:01:08 Call-ID: 3be95048fe554c3c3280f75e75a16977 VOIP data from: [192.169.64.21]:2080
09:01:17 Jitter:4 0 0 0 0 0 0 0 0 0: Buff:26 WC:19 Avg:4
09:01:17 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0 VOIP data from: [192.169.64.21]:2126
09:01:18 Jitter:0 0 0 0 0 0 0 0 0 0: Buff:11 WC:1 Avg:0
09:01:18 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
09:01:24 SIP<CANCEL sip:1000@google.com SIP/2.0 09:01:24 Call-ID: 3be95048fe554c3c3280f75e75a16977
09:01:24 SIP>SIP/2.0 200 OK
09:01:24 Call-ID: 3be95048fe554c3c3280f75e75a16977
09:01:24 SIP>SIP/2.0 487 Request Terminated
09:01:24 Call-ID: 3be95048fe554c3c3280f75e75a16977
09:01:24 SIP>CANCEL sip:19992005000@google.com;user=phone SIP/2.0
09:01:24 Call-ID: 33ba4aaaa3cb41edac1d0c29df7964
09:01:24 idle trunk-group 910 member 98 cid 0x871
09:01:37 Calling party trunk-group 997 member 83 cid 0x88a
 
Is extension 1000 an individual that is trying to forward to their cell phone? My suggestion is to create a coverage path for station 1000 that routes to a vdn. The vector will have a first step with wait-time 2 sec hearing ringback. Second step would be a route to statement 19992005000 so it calls that number. You can also use ix messaging to send the call off pbx using the feature find me but that requires the advanced licensing.
 
My system has off pbx mapping, so that I use that feature. it works!!! Coverage path would be need to make the case as many as people use. So off pbx is a perfect solution.
I really appreciated it.
 
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