Buy or Renew
Buy or Renew
Cisco + Splunk: It’s a new day for your data. Learn more.
 
cancel
Turn on suggestions
Auto-suggest helps you quickly narrow down your search results by suggesting possible matches as you type.
Showing results for 
Search instead for 
Did you mean: 
cancel
Start a conversation
1432
Views
0
Helpful
10
Replies

call transfers not working after gateway upgrades

bbtMichaelR
Level 1
Level 1

‎10-12-2011 06:35 AM - edited ‎03-17-2019 10:33 PM

we recently upgraded our voice gateways (1861s) from 12.4 to 15.1  --- (C1861-ADVIPSERVICESK9-M), Version 15.1(4)M1

After the upgrade, we cannot transfer calls from phones at the upgraded locations to phones at non-upgraded locations, but visa versa works just fine. All other inbound and outbound calls work just fine.  Also, call forwarding works as well...just not transfers

any ideas?

Labels:
0 Helpful
10 Replies 10

frlindse
Cisco Employee
Cisco Employee

‎10-17-2011 10:32 AM

Hi Michael,

A few questions for clarity:

- What are the phones registered to? (CME,CUCM,etc.)

- What is the connection to the PSTN?

- Do internal transfers (IP phone<-->IP phone<--> IP phone) work?

Thanks,

0 Helpful

bbtMichaelR
Level 1
Level 1
In response to frlindse

‎10-17-2011 10:37 AM

1. CUCM

2. SIP to provider

3. transfers within branch work, transfers to another branch do not, transfers from another branch do.

0 Helpful

frlindse
Cisco Employee
Cisco Employee
In response to bbtMichaelR

‎10-17-2011 12:34 PM

Hi Michael,

could you send a copy of the running config and "debug ccsip messages for an attempted transfer?  Please include the pstn caller, internal called, and transferee numbers.  This will show if it's a CUCM or gateway issue.

Thanks,

0 Helpful

bbtMichaelR
Level 1
Level 1
In response to frlindse

‎10-26-2011 09:53 AM

Here is the config.  running that debug from the gateway gave me no output.

Current configuration : 6677 bytes
!
! No configuration change since last restart
!
version 12.4
no service pad
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime localtime show-timezone
service timestamps log datetime localtime show-timezone
service password-encryption
!
hostname vg1
!
boot-start-marker
boot system flash:c1861-advipservicesk9-mz.151-4-M1.bin
boot-end-marker
!
security passwords min-length 8
logging message-counter syslog
logging buffered 4096
logging console informational
!
aaa new-model
!
!
aaa authentication login default local
aaa authorization exec default local
aaa authorization network default local
!
!
aaa session-id common
mmi polling-interval 60
no mmi auto-configure
no mmi pvc
mmi snmp-timeout 180
crypto pki token default removal timeout 0
!
!
dot11 syslog
no ip source-route
ip cef
!
!
no ip dhcp use vrf connected
ip dhcp excluded-address X.y.z.1 X.y.z.20
ip dhcp excluded-address X.y.z.100

ip dhcp pool voice30
   network X.y.z.0 255.255.255.0
   default-router X.y.z.100
   option 150 ip X.y.m.1
!
!
no ip bootp server
no ip domain lookup
ip domain name localdom.local
no ipv6 cef
!
!
!
!
!
multilink bundle-name authenticated
!
!
!
voice service voip
fax protocol cisco
modem passthrough nse codec g711ulaw
!
!
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
!
!
!
!
voice class h323 1
  h225 timeout tcp establish 3
!
!
!
!
!
!
!
!
!
!
!
voice-card 0

username yyyyyyyyyyyy privilege 15 secret 5 XXXXXXXXXXX
username zzzzzzzzzzzz privilege 15 secret 5 XXXXXXXXXXXXXXXX
!
!
!
!
!
ip tcp synwait-time 10
ip ssh time-out 60
ip ssh version 2
!
!
!
!
interface FastEthernet0/0
ip address X.w.z.90 255.255.255.0
ip verify unicast reverse-path
no ip redirects
no ip unreachables
no ip proxy-arp
duplex auto
speed auto
!
interface FastEthernet0/0.30
encapsulation dot1Q 30
ip address X.y.z.100 255.255.255.0
ip access-group VOIPnetwork out
no ip redirects
no ip unreachables
no ip proxy-arp
h323-gateway voip interface
h323-gateway voip bind srcaddr X.y.z.100
!
interface FastEthernet0/1/0
shutdown
!
interface FastEthernet0/1/1
shutdown
!
interface FastEthernet0/1/2
shutdown
!
interface FastEthernet0/1/3
shutdown
!
interface FastEthernet0/1/4
shutdown
!
interface FastEthernet0/1/5
shutdown
!
interface FastEthernet0/1/6
shutdown
!
interface FastEthernet0/1/7
shutdown
!
interface FastEthernet0/1/8
shutdown
!
interface Vlan1
no ip address
no mop enabled
!
ip default-gateway X.u.z.100
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 X.u.z.100
no ip http server
no ip http secure-server
!
!
!
!
ip access-list extended VOIPnetwork
permit ip X.y.5.0.0 0.0.255.255 X.y.0.0 0.0.255.255
permit ip X.y.0.0 0.0.255.255 X.v.n.0 0.0.0.255
permit ip X.v.n.0 0.0.0.255 X.y.0.0 0.0.255.255
deny   ip any any log

!
!
!

!
control-plane
!
!
!
voice-port 0/0/0
!
voice-port 0/0/1
!
voice-port 0/0/2
!
voice-port 0/0/3
!
voice-port 0/1/0
connection plar 2259
!
voice-port 0/1/1
!
voice-port 0/1/2
!
voice-port 0/1/3
!
voice-port 0/4/0
auto-cut-through
signal immediate
input gain auto-control
description Music On Hold Port
!
!
dial-peer voice 1 pots
destination-pattern 91[2-9]..[2-9]......
port 0/1/0
forward-digits 11
!
dial-peer voice 2 pots
destination-pattern 9913[2-9]......
port 0/1/0
forward-digits 10
!
dial-peer voice 3 pots
destination-pattern 9816[2-9]......
port 0/1/0
forward-digits 10
!
dial-peer voice 4 pots
destination-pattern 911
port 0/1/0
forward-digits 3
!
dial-peer voice 5 pots
destination-pattern 9911
port 0/1/0
forward-digits 3
!
dial-peer voice 200 pots
incoming called-number .
port 0/1/0
!
dial-peer voice 201 voip
preference 1
destination-pattern 2259
session target ipv4:X.v.n.1
dtmf-relay h245-alphanumeric
!
dial-peer voice 202 voip
preference 2
destination-pattern 2259
session target ipv4:X.y.m.1
dtmf-relay h245-alphanumeric
!
!
!
dtmf-relay h245-alphanumeric
!
!
!
!
call-manager-fallback
max-conferences 2 gain -6
transfer-system full-consult
ip source-address X.y.z.100 port 2000
max-ephones 12
max-dn 24 dual-line
!

0 Helpful

fb_webuser
Level 6
Level 6

‎10-26-2011 10:40 AM

Depending on a bunch of factors/setup it is possible that the router is thinking the incoming call setup is toll fraud and blocking it. Make sure the IP ranges of your other sites are in the list. Just a thought.

---

Posted by WebUser Scott Haligowski

0 Helpful

bbtMichaelR
Level 1
Level 1
In response to fb_webuser

‎10-26-2011 02:13 PM

It was my understanding that the "debug voip ccapi inout" command would have shown this to me, but I don't see anything when I run that.

Also, when the employee transfers the call, the destination phone rings (there is no ring / hold music for the outside caller), but then the call is dropped as soon as the destination phone is picked up.

0 Helpful

bbtMichaelR
Level 1
Level 1
In response to bbtMichaelR

‎10-27-2011 08:18 AM

Here is the output of "debug CCSIP calls".  I think the disconnect cause (47) here is unusual, but I don't know where to go from there.

031737: Oct 27 14:33:06.028: //113608/655FFDD2B447/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x49E7BC78
State of The Call        : STATE_ACTIVE
TCP Sockets Used         : NO
Calling Number           : XXXYYY1236
Called Number            : VVVWWW3160
Source IP Address (Sig  ): 90.90.234.242
Destn SIP Req Addr:Port  : 90.91.147.141:5060
Destn SIP Resp Addr:Port : 90.91.147.141:5060
Destination Name         : 90.91.147.141

031738: Oct 27 14:33:06.028: //113608/655FFDD2B447/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g729r8
Negotiated Codec Bytes   : 20
Nego. Codec payload      : 18 (tx), 18 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 90.90.234.242
Source IP Port    (Media): 17740
Destn  IP Address (Media): 90.91.147.140
Destn  IP Port    (Media): 14534
Orig Destn IP Address:Port (Media): [ - ]:0

031747: Oct 27 14:33:22.189: //113608/655FFDD2B447/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x49E7BC78
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : XXXYYY1236
Called Number            : VVVWWW3160
Source IP Address (Sig  ): 90.90.234.242
Destn SIP Req Addr:Port  : 90.91.147.141:5060
Destn SIP Resp Addr:Port : 90.91.147.141:5060
Destination Name         : 90.91.147.141

031748: Oct 27 14:33:22.189: //113608/655FFDD2B447/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g729r8
Negotiated Codec Bytes   : 20
Nego. Codec payload      : 18 (tx), 18 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 90.90.234.242
Source IP Port    (Media): 17740
Destn  IP Address (Media): 90.91.147.140
Destn  IP Port    (Media): 14534
Orig Destn IP Address:Port (Media): [ - ]:0

031749: Oct 27 14:33:22.189: //113608/655FFDD2B447/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 47

Disconnect Cause (SIP) : 200

Attached is a very basic diagram of our setup.  I forgot to put the call manager in here, and realize that there may be something going on there too (lots of pieces to troubleshoot).   The above debug came from the "Main GW". Call test was a cell phone calling a direct extension at the Branch.  The employee I called from the cell transfered the call to an extension back at our main location.  The extension rings, but when I pick it up, the voip phone has a fast busy, the cell phone hears nothing, and then the cell phone detects the droped call.

0 Helpful

frlindse
Cisco Employee
Cisco Employee
In response to bbtMichaelR

‎10-27-2011 10:14 AM

Could be the toll fraud mentioned earlier, cherck out

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080b3e123.shtml#t4

0 Helpful

bbtMichaelR
Level 1
Level 1
In response to frlindse

‎10-28-2011 08:32 AM

I tried adding to our ip address trusted list, and the call still drops.

0 Helpful

Thomas Dillingham
Level 1
Level 1
In response to bbtMichaelR

‎11-02-2011 10:33 AM

voice service voip

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

Try this.

TwD

0 Helpful
Getting Started

Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community:

Customers Also Viewed These Support Documents