12-07-2016 02:37 AM - edited 03-17-2019 08:52 AM
Hello Experts,
I have extension 8801 shared between SIP and SCCP phones. When a call is made, it rings the SCCP phone only and once we answer call and hang-up, 2nd call made to this extension always receive busy signal. In order to make it work again I have to reload router, but it works for only 1 call. Every next call gets the busy tone.
Configuration:
ephone-dn 15 octo-line
number 8801
label CHIEF ENG DAYROOM 8801
name CHIEF ENG DAYROOM 8801
shared-line sip
voice register dn 15
number 8801
name CHIEF ENG BEDROOM 801
shared-line max-calls 4
label CHIEF ENG BEDROOM 801
ephone 15
description 7931
mac-address 381C.1ABA.26CE
ephone-template 2
max-calls-per-button 2
type 7931
auto-line 1 answer-incoming
button 1:15 2s2 3s3 4s4
button 5s5 6s6 7s7 8s8
button 9s9 10s10 11s11 12s12
button 13s13 14s14 15s1 16s16
button 17s17 18s18 19s19 20s20
button 21s21 22s22 23s23 24s24
voice register pool 96
id mac BC67.1CDC.4FE6
type 3905
number 1 dn 15
presence call-list
dtmf-relay rtp-nte
username 8801 password 123789
codec g711ulaw
no vad
Dial-peer status:
dial-peer voice 20015 pots
destination-pattern 8801$
huntstop
progress_ind setup enable 3
port 50/0/15
dial-peer voice 40057 voip
destination-pattern 8801$
session target ipv4:10.87.200.180:5060
session protocol sipv2
dtmf-relay rtp-nte
digit collect kpml
codec g711ulaw bytes 160
no vad
after-hours-exempt FALSE
awaiting your expert advice :-)
Abhi
12-07-2016 07:35 AM
Hi Abhi,
Have you tried with no huntstop command on the dn configuration ?
HTH
Rajan
12-07-2016 08:29 AM
Thanks for your response Rajan!!
Well I have tried it with no huntstop as well but ended with no luck.
I have similar configuration on other shared line and all working fine except this.
Abhi
12-08-2016 06:45 AM
Can you provide the below debugs for the failed call along with the call details.
debug voip ccapi inout
debug ccsip messages
Thanks
Rajan
12-08-2016 06:59 AM
here you go...
File attached...
Disconnect cause shows user busy, though it is not the case.
SIP/2.0 486 Busy here
Via: SIP/2.0/UDP 10.87.200.109:5060;rport;branch=z9hG4bKPj59Em52oTvY-vqg1IeQ5ozKhWtt4QDN3X
From: "SENIOR ELECTRICIAN" <sip:4411@10.87.200.2>;tag=93f2c1d8-6a8f-4edd-84e6-9e00ca990c72
To: sip:8@10.87.200.2;tag=989B24D0-12E2
Date: Tue, 29 Nov 2016 09:16:46 GMT
Call-ID: b6155db2-2101-4f05-a99d-f905801f989f
CSeq: 581 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.M2
Reason: Q.850;cause=17
Content-Length: 0
12-09-2016 03:45 AM
Can you check what is this dial-peer as we are hitting this dial-peer as well. I see this is not the one you have mentioned before.
Dial-peer Tag=40134
HTH
Rajan
12-09-2016 11:11 PM
Hello Rajan,
It is possible that DP tag has changed after I recreated the Voice register dn. Previously it was tag 40134, but now it has changed to 40057.
I can see new tag in recent debug message.
Abhi
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