Transfer/conference from CM to SIP station at Session Manager, line not released

[ejvl]

Hi,
We've the following configuration for one of our customers.
Avaya Midsize Enterprise with CM, SM and SMGR.
Avaya Dect, configured and registered on the Session Manager by SIP and routed by AAR.
The Dect is configured with EC500 with a 96xx station, so when the station rings, the dect is also ringing.

Situation:
When there is an incoming call, both stations ringing. Customer answers with dect station, talk to the caller and transfer the conversation to another station.
When there is a next incoming call, only the 96xx is ringing, not the dect. It looks like the dect is still in the first conversation, mabye with a conference?

Also when we transfer from a 96xx station to a dect station, the line is still active on the 96xx station, it looks like a converence in stead of a transfer or something....

Here is a trace of a test call where 1212 is the dect and 2107 the 96xx station. 1212 have a conversation, transfer to 2107, then 2107 transfer to 1212 and the line on 2107 is still active after transfer.

09:51:03 TRACE STARTED 11/18/2016 CM Release String cold-02.0.823.0-20199
09:52:30 Calling party station 2107 cid 0x1e27
09:52:30 Calling Number & Name 2107 xxxx
09:52:30 dial 1212
09:52:30 ring station 1212 cid 0x1e27
09:52:30 dial 1212
09:52:30 seize trunk-group 129 member 2 cid 0x1e28
09:52:30 Calling Number & Name NO-CPNumber NO-CPName
09:52:30 Setup digits 1212
09:52:30 Calling Number & Name 2107 xxxx
09:52:30 Proceed trunk-group 129 member 2 cid 0x1e28
09:52:31 Alert trunk-group 129 member 2 cid 0x1e28
09:52:36 transfer station 2107 cid 0x1e27
09:52:39 active station 1212 cid 0x1e27
09:52:55 conf/tran hold station 1212 cid 0x1e27
time data
09:52:55 active station 1212 cid 0x1e2c
09:52:55 G711A ssff ps:20
rgn:12 [10.116.254.237]:28240
rgn:12 [10.116.254.221]:2064
09:52:55 xoip options: fax:T38 modemT ttyff uid:0x5000c
xoip ip: [10.116.254.221]:2064
09:52:58 dial 2107
09:52:58 ring station 2107 cid 0x1e2c
09:52:58 G711A ssff ps:20
rgn:12 [10.116.252.72]:2422
rgn:12 [10.116.254.221]:2082
09:52:58 SIP>INVITE sip:81214@xxxx SIP/2.0
09:52:58 Call-ID: 04caec4e2b6e61c9d856d33a9000
09:52:58 SIP<SIP/2.0 100 Trying
09:52:58 Call-ID: 04caec4e2b6e61c9d856d33a9000
09:52:58 SIP<INVITE sip:81214@xxxx SIP/2.0
09:52:58 Call-ID: 04caec4e2b6e61c9d856d33a9000
09:52:58 SIP>SIP/2.0 100 Trying
09:52:58 Call-ID: 04caec4e2b6e61c9d856d33a9000
09:52:58 SIP>SIP/2.0 180 Ringing
09:52:58 Call-ID: 04caec4e2b6e61c9d856d33a9000
09:52:58 SIP<SIP/2.0 180 Ringing
09:52:58 Call-ID: 04caec4e2b6e61c9d856d33a9000
09:52:58 SIP>PRACK sip:xxxx;transport=tcp SIP/2.0
09:52:58 Call-ID: 04caec4e2b6e61c9d856d33a9000
09:52:58 SIP<PRACK sip:xxxx;transport=tcp SIP/2.0
09:52:58 Call-ID: 04caec4e2b6e61c9d856d33a9000
09:52:58 SIP>SIP/2.0 200 OK
09:52:58 Call-ID: 04caec4e2b6e61c9d856d33a9000
09:52:58 SIP<SIP/2.0 200 OK
09:52:58 Call-ID: 04caec4e2b6e61c9d856d33a9000
09:53:02 SIP>SIP/2.0 200 OK
09:53:02 Call-ID: 04caec4e2b6e61c9d856d33a9000
09:53:02 SIP<SIP/2.0 200 OK
09:53:02 Call-ID: 04caec4e2b6e61c9d856d33a9000
09:53:02 SIP>ACK sip:xxxx;transport=tcp SIP/2.0
09:53:02 Call-ID: 04caec4e2b6e61c9d856d33a9000
09:53:02 active station 2107 cid 0x1e2c
09:53:02 SIP<ACK sip:xxxx;transport=tcp SIP/2.0
09:53:02 Call-ID: 04caec4e2b6e61c9d856d33a9000
VOIP data from: [10.116.254.221]:2064
09:53:06 Jitter:6 0 0 0 0 0 0 0 0 0: Buff:18 WC:7 Avg:6
09:53:06 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
VOIP data from: [10.116.254.221]:2066
09:53:07 Jitter:1 0 0 0 0 0 0 0 0 0: Buff:12 WC:1 Avg:1
09:53:07 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
09:53:08 transfer station 1212 cid 0x1e2c
09:54:05 Calling party station 1214 cid 0x1e33
time data
09:54:05 Calling Number & Name 1214 ~~xxxx
09:54:05 dial 1212
09:54:05 ring station 1212 cid 0x1e33
09:54:05 dial 1212
09:54:05 seize trunk-group 129 member 4 cid 0x1e34
09:54:05 Setup digits 81214
09:54:05 Calling Number & Name NO-CPNumber NO-CPName
09:54:05 Setup digits 1212
09:54:05 Calling Number & Name 1214 ~xxxx42
09:54:05 Proceed trunk-group 129 member 4 cid 0x1e34
09:54:05 Alert trunk-group 129 member 4 cid 0x1e34
09:54:08 G711A ssff ps:20
rgn:12 [10.116.254.231]:20124
rgn:12 [10.116.254.221]:2062
09:54:08 xoip options: fax:T38 modemT ttyff uid:0x5000e
xoip ip: [10.116.254.221]:2062
time data
09:54:08 active station 1212 cid 0x1e33
09:54:08 G711A ssff ps:20
rgn:12 [10.116.254.231]:20124
rgn:12 [10.116.254.231]:20112
09:54:08 G711A ssff ps:20
rgn:12 [10.116.254.231]:20112
rgn:12 [10.116.254.231]:20124
09:54:09 transfer station 1214 cid 0x1e33
09:54:09 G711A ssff ps:20
rgn:12 [10.116.254.231]:20124
rgn:1 [10.100.0.101]:2068
09:54:09 xoip options: fax:T38 modemT ttyff uid:0x5000e
xoip ip: [10.100.0.101]:2068
VOIP data from: [10.100.0.101]:2068
09:54:19 Jitter:0 0 0 0 0 0 0 0 0 0: Buff:11 WC:0 Avg:0
09:54:19 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
09:54:08 active station 1212 cid 0x1e33
09:54:08 G711A ssff ps:20
rgn:12 [10.116.254.231]:20124
rgn:12 [10.116.254.231]:20112
09:54:08 G711A ssff ps:20
rgn:12 [10.116.254.231]:20112
rgn:12 [10.116.254.231]:20124
09:54:09 transfer station 1214 cid 0x1e33
09:54:09 G711A ssff ps:20
rgn:12 [10.116.254.231]:20124
rgn:1 [10.100.0.101]:2068
09:54:09 xoip options: fax:T38 modemT ttyff uid:0x5000e
xoip ip: [10.100.0.101]:2068
VOIP data from: [10.100.0.101]:2068
09:54:19 Jitter:0 0 0 0 0 0 0 0 0 0: Buff:11 WC:0 Avg:0
09:54:19 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0

any idea how to fix?
I think when customer transfer a call, station does not a transfer but a conference or something...
Thanks!

 

[kyle555]

Maybe this:

display system-parameters features                              Page   7 of  20
                        FEATURE-RELATED SYSTEM PARAMETERS

CONFERENCE/TRANSFER
               Abort Transfer? n                 No Dial Tone Conferencing? n
        [highlight #FCE94F]Transfer Upon Hang-Up? n[/highlight]       Select Line Appearance Conferencing? n
             Abort Conference? n                                    Unhold? y
   No Hold Conference Timeout: 60
 No LED Wink during soft hold? n

Also, if you display off-pbx station-mapping of the extension with EC500, maybe there's a max calls allowed of 1 that you can increase.

 

[ejvl]

Thanks!
Our configuration is:
CONFERENCE/TRANSFER
Abort Transfer? n No Dial Tone Conferencing? n
Transfer Upon Hang-Up? y Select Line Appearance Conferencing? n
Abort Conference Upon Hang-Up? n Unhold? n
No Hold Conference Timeout: 60

There is no option for max calls allowed in EC500 configuration.

 

[kyle555]

OK. You should never be transferring to the SIP station from the 96xx for which their is an EC500 mapping. You can use an extend key for that.

What model of SIP phones do you have? Maybe there's some set-specific configuration that can happen to help out. Seeing a traceSM of that flow from the sip phone would be helpful too.

 

[ejvl]

Customer doesn't transfer to their own station.
For example:
User 1000 with 9608 phone and extension 1001 for their sip (dect) phone.
User 2000 with 9608 phone and extension 2001 for their sip (dect) phone.

User 1000 received an incoming call, 1000 and 1001 are ringing. User answer with 1001 dect, talk to the caller and transfer the call to user 2000, 2000 answer the call, 1001 transfer and disconnect.
Then there is a new call for user 1000, 9608 phone is ringing but dect 1001 is not ringing! It looks like 1001 is still in a conversation (maybe conference?) with 2000.

Another example:
User 1000 received an incoming call, 1000 and 1001 are ringing. User answer with 1000, talk to the caller and transfer the call to user 2000. User 2000 answer with dect 2001. User 1000 transfer the call and disconnect.
But the first line of the 9608 phone is still active. When he push the line button, he is in conference, when he disconnect the line, the whole conversation is gong, also by 2001.

The SIP stations are 3720 and 3725 Avaya IP Dect stations, registered with Avaya Base Stations and registered in Session Manager.

A Trace SM is available on next Monday, I'll post it soon as possible!

 

[kyle555]

any other sip phones you can try with like a 96xx sip just to see if its anything specific to the 3700?

 

[gwebster]

I thought that Avaya DECT R4 used H.323/X-Mobile protocol to connect CM via H.323 trunks, not SIP.