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-   -   SIP Trunk call to IVR IGC Removed. (http://support.avaya.com/forums/showthread.php?t=5879)

adrianchurchill 01-06-2015 05:13 AM

SIP Trunk call to IVR IGC Removed.
 
I have an IVR system connected via QSIG. When I try and call into the IVR from a SIP trunk the call appears to answer but there is no audio. Calls from ISDN trunks and internal stations work fine.

I think this is something to do with the IGC removed entry in the trace below:

Any suggestions appreciated.

12:53:14 SIP<INVITE sip:387520@10.72.2.21;user=phone SIP/2.0
12:53:14 Call-ID: c49392310620e0024d9971526e22e311
12:53:14 active trunk-group 61 member 125 cid 0x22ed
12:53:14 dial 4584 route:UDP|AAR
12:53:14 term trunk-group 45 cid 0x22ed
12:53:14 xoip options: fax:Relay modem:off tty:UK (igc)
xoip ip: [10.72.2.17]:25436
12:53:14 xoip options: fax:Relay modem:off tty:UK (igc)
xoip ip: [10.72.2.32]:23416
12:53:14 G711A ss:off ps:20 - IGC created
rgn:1 [10.72.2.17]:25436 (PN1)
rgn:1 [10.72.2.32]:23416 (PN2)
12:53:14 dial 4584 route:UDP|AAR
12:53:14 route-pattern 45 preference 1 location 3/ALL cid 0x22ed
12:53:14 seize trunk-group 45 member 9 cid 0x22ed
time data
12:53:14 Setup digits 4584
12:53:14 Calling Number & Name NO-CPNumber Anonymous
12:53:14 Proceed trunk-group 45 member 9 cid 0x22ed
12:53:14 SIP>SIP/2.0 180 Ringing
12:53:14 Call-ID: c49392310620e0024d9971526e22e311
12:53:14 Alert trunk-group 45 member 9 cid 0x22ed
12:53:14 IGC removed
rgn:1 [10.72.2.17]:25436 (PN1)
rgn:1 [10.72.2.32]:23416 (PN2)
12:53:14 G711A ss:off ps:20
rgn:1 [10.2.200.20]:44754
rgn:1 [10.72.2.17]:25452
12:53:14 xoip options: fax:T38 modem:off tty:UK uid:0x50523
xoip ip: [10.72.2.17]:25452
12:53:14 SIP<PRACK sip:10.72.2.21;user=phone;transport=tcp SIP/2.0
12:53:14 Call-ID: c49392310620e0024d9971526e22e311
time data
12:53:14 SIP>SIP/2.0 200 OK
12:53:14 Call-ID: c49392310620e0024d9971526e22e311
12:53:16 SIP>SIP/2.0 200 OK
12:53:16 Call-ID: c49392310620e0024d9971526e22e311
12:53:16 active trunk-group 45 member 9 cid 0x22ed
12:53:16 SIP<ACK sip:10.72.2.21;transport=tcp SIP/2.0
12:53:16 Call-ID: c49392310620e0024d9971526e22e311
VOIP data from: [10.72.2.17]:25452
12:53:26 Jitter:0 0 0 0 0 0 0 0 0 0: Buff:11 WC:0 Avg:0
12:53:26 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
VOIP data from: [10.72.2.17]:25452
12:53:34 Jitter:0 0 0 0 0 0 0 0 0 0: Buff:11 WC:0 Avg:0
12:53:34 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
12:53:34 SIP<BYE sip:10.72.2.21;transport=tcp SIP/2.0
12:53:34 Call-ID: c49392310620e0024d9971526e22e311
12:53:34 SIP>SIP/2.0 200 OK

schaf10 01-26-2015 12:35 PM

Can you check that the attached data (DNIS or caller ID) of the call is not being over written by the QSIG trunk? I think your going to have to look at the IVR logs on this one. I think your routing is ok, but your IVR does not know what to do with the call.

Check the public or private numbering is administered for all trunks.


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