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SIP Outbound calls

G3261
Level 4
Level 4

I am  having problems with SIP trunks and outbound call from one specific shared line. when in SRST, if I dial any number from this shared line (0100) I get fast busy!!! Below is fallback mode configuration. Do you see anything wrong with this? Having same number 0100 as call-forward busy can create any issues?

call-manager-fallback
 secondary-dialtone 9
 max-conferences 8 gain -6
 transfer-system full-consult
 limit-dn 79xx 2
 limit-dn 79xx 2
 limit-dn 79xx 2
 limit-dn 79xx 2
 ip source-address 192.168.XXX.XXX port 2000
 max-ephones 75
 max-dn 75
 system message primary Fallback Mode
 no huntstop
 pickup 0100
 call-forward busy 0100
 time-zone 10
 time-format 24
 date-format dd-mm-yy
!

7 Replies 7

Aseem Anand
Cisco Employee
Cisco Employee

Hi,

Can you collect the below debugs for a test call:

Debug ccsip messages

debug voice ccapi inout

Aseem

ccsip messages----

000367: Oct 26 2016 12:24:19.222 xxxx: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:5@192.168.XXX.XXX;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.XXX.XXX:5060;branch=z9hG4bK0c79e3d8
From: "Site Name" <sip:XXXX@192.168.XXX.XXX>;tag=3820560c506c8de64b1e0bc5-2bd7a0e9
To: <sip:5@192.168.XXX.XXX>
Call-ID: 3820560c-506c0fa2-68233502-15aadced@192.168.XXX.XXX
Max-Forwards: 70
Session-ID: 4e08a85d00105000a0003820560c506c;remote=00000000000000000000000000000000
Date: Wed, 26 Oct 2016 18:24:19 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7821/11.0.1
Contact: <sip:52998a58-8326-80ed-8260-016bf8aae6cd@192.168.XXX.XXX:5060;transport=udp>;+u.sip!devicename.ccm.cisco.com="SEP3820560C506C"
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Site Name" <sip:XXXX@192.168.XXX.XXX>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,X-cisco-srtp-fallback,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 353
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 22137 0 IN IP4 192.168.XXX.XXX
s=SIP Call
b=AS:4064
t=0 0
m=audio 19716 RTP/AVP 0 8 116 18 101
c=IN IP4 192.168.XXX.XXX
b=TIAS:64000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

000368: Oct 26 2016 12:24:19.234 xxxx: //1159/3E7023B28846/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.XXX.XXX:5060;branch=z9hG4bK0c79e3d8
From: "Site Name" <sip:XXXX@192.168.XXX.XXX>;tag=3820560c506c8de64b1e0bc5-2bd7a0e9
To: <sip:5@192.168.XXX.XXX>
Date: Wed, 26 Oct 2016 18:24:19 GMT
Call-ID: 3820560c-506c0fa2-68233502-15aadced@192.168.XXX.XXX
CSeq: 101 INVITE
Allow-Events: kpml, telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.M4
Content-Length: 0


000369: Oct 26 2016 12:24:19.234 xxxx: //1159/3E7023B28846/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.XXX.XXX:5060;branch=z9hG4bK0c79e3d8
From: "Site Name" <sip:XXXX@192.168.XXX.XXX>;tag=3820560c506c8de64b1e0bc5-2bd7a0e9
To: <sip:5@192.168.XXX.XXX>;tag=293F5C08-4B6
Date: Wed, 26 Oct 2016 18:24:19 GMT
Call-ID: 3820560c-506c0fa2-68233502-15aadced@192.168.XXX.XXX
CSeq: 101 INVITE
Allow-Events: kpml, telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.M4
Reason: Q.850;cause=21
Content-Length: 0


000370: Oct 26 2016 12:24:19.270 xxxx: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:5@192.168.XXX.XXX;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.XXX.XXX:5060;branch=z9hG4bK0c79e3d8
From: "Site Name" <sip:XXXX@192.168.XXX.XXX>;tag=3820560c506c8de64b1e0bc5-2bd7a0e9
To: <sip:5@192.168.XXX.XXX>;tag=293F5C08-4B6
Call-ID: 3820560c-506c0fa2-68233502-15aadced@192.168.XXX.XXX
Session-ID: 4e08a85d00105000a0003820560c506c;remote=00000000000000000000000000000000
Max-Forwards: 70
Date: Wed, 26 Oct 2016 18:24:19 GMT
CSeq: 101 ACK
Content-Length: 0

Hi,

I am assuming this call was made from a phone registered in SRST with CME.

Can you please collect the output of:

sh voice register pool all brief

show ip address trusted list (Do you see the phone's IP address there?)

Show license feature

Show license detail cme-srst

Aseem

for security reasons, I am not allow to post certain things!!! Is there anything specific you are looking for in these debugs?

I don't have CME but has cube router.

run the command sh voice register pool all brief and note the IP address of the phone you are having issues with and then run the command show ip address trusted list and check if the IP address is there in the list?

Aseem

ok it will take sometime for me to take them down for SRST to kick in....however there are 3 phones with 0100 shared number. If I use their 2nd line 0200 and dial a number in SRST, all works well. But if I dial same number using 0100 (shared line) it doesn't!!! So having the right IP in trusted matters????

see ccsip in srst