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493
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15
Helpful
6
Replies

Issue with Shared line Between SIP and SCCP phones.

actualabhishek
Level 1
Level 1

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

6 Replies 6

Rajan
VIP Alumni
VIP Alumni

Hi Abhi,

Have you tried with no huntstop command on the dn configuration ?

HTH

Rajan

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

Can you provide the below debugs for the failed call along with the call details.

debug voip ccapi inout

debug ccsip messages

Thanks 

Rajan

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

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

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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