11-06-2014 10:22 AM - edited 03-17-2019 12:50 AM
We are testing SIP through one of our offices and are having trouble with the calls going through. When we place the calls they ring like a normal call, but when answered you get a fast busy. We have SIP service provided through Level 3 communications. I spoke with them for debug purposes and they're seeing the call ring acknowledged, an ok when answered and then an immediate bye. I'm not sure if it's a config issue in Call Manager or on the gateway.
here is the level 3 debug, they provided me:
BYE sip:4788450763@207.227.240.79:5060 SIP/2.0
Via: SIP/2.0/UDP 10.101.1.6:5060;branch=z9hG4bK3191331
From: "test" <sip:7246204141@10.101.1.6>;tag=A6877434-491
To: <sip:4788450763@207.227.240.79>;tag=gK00c871fb
Date: Thu, 06 Nov 2014 15:25:09 GMT
Call-ID: EDB70A16-64FF11E4-9589F6A1-7A7E53DF@10.101.1.6
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1415287515
CSeq: 103 BYE
Reason: Q.850;cause=96
P-RTP-Stat: PS=0,OS=0,PR=229,OR=36640,PL=0,JI=0,LA=0,DU=0
Content-Length: 0
Here's a call debug, that I just produced:
SGW_Main#
Nov 6 17:26:52.962: //-1/1780DB800000/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x2B987538
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : xxx6204141
Called Number : 7xxx4542969
Source IP Address (Sig ): x.101.1.6
Destn SIP Req Addr:Port : x.10.1.6:0
Destn SIP Resp Addr:Port : x.10.1.6:5060
Destination Name : x.10.1.6
Nov 6 17:26:52.962: //-1/1780DB800000/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 100
Disconnect Cause (SIP) : 422
SGW_Main#
Nov 6 17:27:02.182: //529700/1780DB800000/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x2B976768
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : xxx6204141
Called Number : xxx4542969
Source IP Address (Sig ): x.101.1.6
Destn SIP Req Addr:Port : x.227.240.79:5060
Destn SIP Resp Addr:Port : x.227.240.79:5060
Destination Name : x.227.240.79
Nov 6 17:27:02.182: //529700/1780DB800000/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711ulaw
Negotiated Codec Bytes : 160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): x.101.1.6
Source IP Port (Media): 30516
Destn IP Address (Media): x.227.240.78
Destn IP Port (Media): 11536
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 6 17:27:02.202: //529699/1780DB800000/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x2B987538
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : xxx6204141
Called Number : 7xxx4542969
Source IP Address (Sig ): x.101.1.6
Destn SIP Req Addr:Port : x.10.1.6:5060
Destn SIP Resp Addr:Port : x.10.1.6:5060
Destination Name : x.10.1.6
Nov 6 17:27:02.202: //529699/1780DB800000/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): x.101.1.6
Source IP Port (Media): 21416
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 6 17:27:02.202: //529699/1780DB800000/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 86
Disconnect Cause (SIP) : 500
Nov 6 17:27:02.238: //529700/1780DB800000/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x2B976768
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : xxx6204141
Called Number : xxx4542969
Source IP Address (Sig ): x.101.1.6
Destn SIP Req Addr:Port : x.227.240.79:5060
Destn SIP Resp Addr:Port : x.227.240.79:5060
Destination Name : x.227.240.79
SGW_Main#
Nov 6 17:27:02.238: //529700/1780DB800000/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711ulaw
Negotiated Codec Bytes : 160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): x.101.1.6
Source IP Port (Media): 30516
Destn IP Address (Media): x.227.240.78
Destn IP Port (Media): 11536
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 6 17:27:02.238: //529700/1780DB800000/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 96
Disconnect Cause (SIP) : 200
error messages:
11-06-2014 11:17 AM
Hi somersettrust,
Can you paste your CUBE configuration here?
11-06-2014 11:32 AM
11-06-2014 11:56 AM
Yes, running-config from your VGW would be helpful.
11-06-2014 01:32 PM
here's the vg config: We have approximately 25 offices running through 2 PRI's. we just implemented SIP for a new office.
SGW_Main#sh run
Building configuration...
Current configuration : 9078 bytes
!
! Last configuration change at 16:29:33 EST Wed Nov 5 2014 by mostoller
! NVRAM config last updated at 17:10:37 EST Wed Nov 5 2014 by mostoller
! NVRAM config last updated at 17:10:37 EST Wed Nov 5 2014 by mostoller
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
hostname SGW_Main
!
boot-start-marker
boot-end-marker
!
!
card type t1 0 0
logging buffered 100000000
!
no aaa new-model
clock timezone EST -5 0
clock summer-time EDT recurring
network-clock-participate wic 0
network-clock-select 1 T1 0/0/0
!
no ipv6 cef
ip source-route
ip cef
!
!
!
!
!
multilink bundle-name authenticated
!
!
!
!
isdn switch-type primary-4ess
!
crypto pki token default removal timeout 0
!
crypto pki trustpoint TP-self-signed-4018802375
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-4018802375
revocation-check none
rsakeypair TP-self-signed-4018802375
!
!
crypto pki certificate chain TP-self-signed-4018802375
certificate self-signed 01
quit
voice-card 0
dsp services dspfarm
!
!
!
voice service voip
ip address trusted list
ipv4 X.227.240.0 255.255.255.0
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
min-se 14400 session-expires 14400
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
!
!
!
!
voice translation-rule 1
rule 1 /^\+1......\(....\)$/ /\1/
!
voice translation-rule 7
rule 1 /^7/ //
!
!
voice translation-profile INCOMING-L3
translate called 1
!
voice translation-profile OUTGOING-L3
translate called 7
!
!
!
application
service CMM http://10.10.1.6:8080/ccmivr/pages/IVRMainpage.vxml
!
global
service alternate Default
!
!
license udi pid CISCO2911/K9 sn FTX1701A104
hw-module pvdm 0/0
!
hw-module pvdm 0/1
!
!
!
archive
log config
logging enable
logging size 1000
notify syslog contenttype plaintext
hidekeys
!
redundancy
!
!
controller T1 0/0/0
cablelength long 0db
pri-group timeslots 1-24 service mgcp
!
ip ssh version 2
!
!
!
!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0
ip address x.101.1.6 255.255.252.0
duplex auto
speed auto
!
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
interface GigabitEthernet0/2
no ip address
shutdown
duplex auto
speed auto
!
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
!
ip forward-protocol nd
!
no ip http server
ip http access-class 10
ip http authentication local
no ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
!
ip route 0.0.0.0 0.0.0.0 x.101.1.1
!
access-list 10 permit x.x.1.29
access-list 10 permit x.x.1.55
access-list 10 permit x.x.1.32
access-list 10 permit x.x.1.70
access-list 10 permit x.x.1.99
access-list 10 permit x.x.1.101
access-list 10 permit x.x.1.164
!
!
snmp-server community hansolo RO 10
snmp-server community jarjarbinks RW 10
snmp-server ifindex persist
!
control-plane
!
!
voice-port 0/0/0:23
!
ccm-manager fallback-mgcp
ccm-manager redundant-host x.10.2.7
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server x.10.1.6
ccm-manager config
!
mgcp
mgcp call-agent x.10.1.6 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp quarantine persistent-event disable
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp rtp payload-type g726r16 static
!
mgcp profile default
!
sccp local GigabitEthernet0/0
sccp ccm x.10.2.7 identifier 2 version 7.0
sccp ccm x.10.1.6 identifier 1 version 7.0
sccp ip precedence 3
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 1 register MTP-SIP-Som
!
sccp ccm group 999
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 999 register GW1-XCODE
!
sccp ccm group 992
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 992 register GW1-CB
!
dspfarm profile 999 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 3
associate application SCCP
!
dspfarm profile 992 conference
codec g729br8
codec g729r8
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
maximum sessions 3
associate application SCCP
!
dspfarm profile 1 mtp
codec pass-through
codec g711ulaw
maximum sessions software 10000
associate application SCCP
!
dial-peer voice 1 pots
destination-pattern 9911
forward-digits 3
!
dial-peer voice 2 pots
description dial-peer for local 7 digit dialing
destination-pattern 9[2-9]......
port 0/0/0:23
forward-digits 7
!
dial-peer voice 3 pots
description dial-peer for LD dialing
destination-pattern 91..........
port 0/0/0:23
forward-digits 11
!
dial-peer voice 100 pots
description dial-peer for inbound DID
incoming called-number .
direct-inward-dial
port 0/0/0:23
!
dial-peer voice 8100 voip
destination-pattern 81..
session target ipv4:x.10.1.6
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
dial-peer voice 88100 voip
destination-pattern 88...
session target ipv4:x.10.1.6
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
dial-peer voice 207 voip
translation-profile incoming INCOMING-L3
translation-profile outgoing OUTGOING-L3
destination-pattern 7.T
session protocol sipv2
session target ipv4:x.227.240.79
session transport udp
incoming called-number .
voice-class codec 1
voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad
!
dial-peer voice 208 voip
destination-pattern [0-6,8-9]...
session protocol sipv2
session target ipv4:x.10.1.6
session transport udp
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
!
!
!
gatekeeper
shutdown
!
!
call-manager-fallback
max-conferences 8 gain -6
transfer-system full-consult
ip source-address x.101.1.6 port 2000
max-ephones 58
max-dn 300 dual-line
system message primary call manager down
system message secondary call manager down
keepalive 10
!
!
banner motd ^C
******************************************************
****** Property ******
***** Access Restricted *****
**** Do not attempt to logon unless authorized ****
******************************************************
^C
!
line con 0
logging synchronous
login local
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
session-timeout 60
access-class 10 in
privilege level 15
logging synchronous
login local
transport input ssh
transport output ssh
line vty 5 15
session-timeout 60
access-class 10 in
privilege level 15
logging synchronous
login local
transport input ssh
transport output ssh
!
scheduler allocate 20000 1000
ntp update-calendar
ntp server x.x.254.1
end
SGW_Main#exit
11-06-2014 03:23 PM
You should have a region for your Branch office, and a region for your SIP trunk between CUCM and your GW. <- Very important.
Your current call flow:
IP phone --- sccp to ---> CUCM ---SIP trunk to -->CUBE gateway ---SIP Trunk to ITSP--> .
To test make sure you have your Iphone in X_REGION and your CUCM sip trunk in Y_Region. Then make sure your region relationship between X and Y = 64 kbps. Then test.
Also....
The Cube config has a voice class with g711ulaw and g729r8 allowed to the ITSP. But your transcoder profile does not have g729r8 in it. I suspect if your region is negotiating g.729 between iphone and CUCM-CUBE sip trunk, it won't be able to transcode to g711ullaw to leave to the ITSP.
Let me know.
11-06-2014 04:35 PM
I'm making the changes now, but won't be able to test it until tomorrow morning. As soon as I test it, I will let you know.
Thanks for the advice.
11-07-2014 06:46 AM
Glad it worked! The CUBE has to have a 911 voip dial peer with the voice class specified.
On another note, your calls were not going through because transcoding is not setup properly on your CUBE. What was happening was your IPphone was negotiating G.729 all the way to the CUBE, but the the CUBE negotiated G.711 to the ITSP. So when the call connected, your CUCM noticed the mismatch and disconnected by sending the "bye" sip message. This can be fixed if your CUBE is properly setup for transcoding, because using G.711 all the time will consume more bandwidth, and you always want to have both codecs working anyways.
Please make sure your SIP Trunk's MGRL in call manager is in the same region as your SIP trunk, and then fix the config on the CUBE. You need to include G.729r8 under the profile for transcoding.
Good luck and rate useful!
11-07-2014 08:30 AM
I have 911 working. When I would call it, it would say "call can not be completed as dialed". I created a new dial-peer specifically for 911 and that fixed it.
dial-peer voice 209 voip
translation-profile incoming INCOMING-L3
translation-profile outgoing OUTGOING-L3
destination-pattern 911
session protocol sipv2
session target ipv4:x.227.240.79
session transport udp
incoming called-number .
voice-class codec 1
voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad
I will verify the SIP Trunk's MGRL relation to the Region. I will also check the config for the transcoding.
Thanks again
11-07-2014 06:24 AM
I added the Region and tweaked a few other settings and the calls went through. Thanks!
I'm still having an issue calling 911, but I haven't looked at the dial peers in the VG. I suspect that may be the problem.
11-06-2014 12:29 PM
I had this issue once, it turned out to be a codec mismatch. One side supported G.711 only while the other side was negotiating G.729 and the transcoders were not setup properly. I initially created a separate voice class in the CUBE to force G.711 and that's how I figured out what was wrong.
Make sure your call manager regions' relationships are setup properly.
Thanks,
11-06-2014 01:38 PM
I tried modifying the region, of the office that we are testing SIP in and that didn't work. Do I need to modify the codec for the new office in relation to all the other offices (we have 20 plus offices) or just the main office (containing the VG and CUCM)?
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