SIP to SIP Call Failures on CME to CME - sip-ua conflict/issue?

Hi,
I have two existing CME systems which I wish to allow internal calls between. These calls will go over an IPSec VPN. However the calls are failing.
Phones DN22xx - London CME 2801 - PIX505 --- Internet ---ASA5505 - India CME 2801 - Phones DN400x
I have configured dial peers on both CME's and the IPSec VPN. I can ping between both systems. The VPN allows traffic between the interface IP's of the CME systems only.
London CME (local SCCP phones 22xx):
interface FastEthernet0/0.100
encapsulation dot1Q 100 native
ip address 10.0.10.250 255.255.255.0
voice class codec 101
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 25 voip
description *** SIP Peer to India ***
answer-address 400.
destination-pattern 400.
voice-class codec 101
session protocol sipv2
session target ipv4:192.168.15.10
incoming called-number 400.
no vad
India CME (Local SSCP phones 400x):
interface FastEthernet0/0
ip address 192.168.15.10 255.255.255.0
voice class codec 100
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 10 voip
description *** SIP Peer to London UK ***
answer-address 22..
destination-pattern 22..
voice-class codec 100
session protocol sipv2
session target ipv4:10.0.10.250
incoming called-number 22..
no vad
The CME system at India also has an existing SIP dial peer to a service provider and has sip-ua configured (username, password, realm and registrar).
A call from India (4005) to London (DN2207) fails, the ccsip debug attached. I'm assuming its because the sip-ua configuration is being used for these calls to when I don't want it to be. The from field shows âFrom: <sip:[email protected]â when I need this to be the internal IP 192.168.15.10.
Can anyone offer any assistance with this?
Regards,
Chris

Hi,
thanks for your input however thats not the problem. 201.196.128.56 isn't an address on the router, it only has one IP and its 192.168.15.10.
The 201.196.128.56 address is the NAT'd address on the firewall. So that when a SIP call is made to the internet with sip-ua the from address is the public IP.
Chris

Similar Messages

CME\7960 running SIP firmware - How do i setup incoming calls? - Can anyone help please?

Hi Guys,
I have a SIP trunk setup with a 2811 running CME version 7. I can make outbound calls ok but having issues getting the incoming calls working, i have 1 number on my SIP trunk and that is 01133501788 and i want that to ring my Cisco 7960 which is running SIP firmware not SCCP. I have included by config for anyone who can help me, i just want the incoming call to work.
Many Thanks.
Matthew.
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
logging message-counter syslog
no aaa new-model
clock timezone GMT 0
dot11 syslog
ip source-route
ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 192.168.1.1
ip dhcp excluded-address 10.10.10.1
ip dhcp pool DATA_POOL
   network 10.10.10.0 255.255.255.0
   default-router 10.10.10.1
   dns-server 188.92.232.50 188.92.232.100
ip dhcp pool VOICE_POOL
   network 192.168.1.0 255.255.255.0
   default-router 192.168.1.1
   dns-server 188.92.232.50 188.92.232.100
   option 150 ip 192.168.1.1
ip name-server 188.92.232.50
ip name-server 188.92.232.100
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
bind control source-interface FastEthernet0/1.20
bind media source-interface FastEthernet0/1.20
registrar server
voice class codec 1
codec preference 2 g711ulaw
codec preference 3 g711alaw
voice register global
mode cme
source-address 192.168.1.1 port 5060
max-dn 144
max-pool 42
load 7960-7940 P0S3-8-12-00
authenticate register
tftp-path flash:
create profile sync 0008072514198272
voice register dn 1
number 6999
allow watch
name SIP
label SIP
voice register pool 1
id mac 000F.902B.40E0
type 7960
number 1 dn 1
dtmf-relay sip-notify
username cisco password cisco
codec g711ulaw
voice translation-rule 1
rule 1 /^9$.*$/ /\1/
voice translation-rule 2
rule 1 /^6...$/ /4143*002/
voice translation-profile DiscardDigit9
translate calling 2
translate called 1
voice translation-profile IncomingSIP
translate calling 1133501788
voice-card 0
no dspfarm
username matt privilege 15 secret 5 $1$DCD0$SjWqnKgDSGVzzIKRerXh11
archive
log config
hidekeys
interface FastEthernet0/0
ip address 194.12.0.222 255.255.255.252
ip nat outside
ip virtual-reassembly
duplex auto
speed auto
interface FastEthernet0/1
no ip address
ip nat inside
ip virtual-reassembly
duplex auto
speed auto
interface FastEthernet0/1.10
description DATA
encapsulation dot1Q 10
ip address 10.10.10.1 255.255.255.0
ip nat inside
ip virtual-reassembly
interface FastEthernet0/1.20
description VOICE
encapsulation dot1Q 20
ip address 192.168.1.1 255.255.255.0
ip nat inside
ip virtual-reassembly
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 194.12.0.221
ip http server
ip http authentication local
no ip http secure-server
ip nat inside source list 1 interface FastEthernet0/0 overload
access-list 1 permit 192.168.1.0 0.0.0.255
access-list 1 permit 10.10.10.0 0.0.0.255
tftp-server flash:P003-8-12-00.bin
tftp-server flash:P003-8-12-00.sbn
tftp-server flash:P0S3-8-12-00.loads
tftp-server flash:P0S3-8-12-00.sb2
tftp-server flash:P003-8-12-00
tftp-server flash:P003-8-12-00.loads
tftp-server flash:P003-8-12-00.sb2
tftp-server flash:SIP000F902B40E0.cnf.xml
control-plane
mgcp behavior g729-variants static-pt
dial-peer cor custom
dial-peer voice 2 voip
description Outgoing Geographic
translation-profile outgoing DiscardDigit9
destination-pattern 0[7]........
voice-class codec 1
session protocol sipv2
session target dns:sip.cloudcalling.co.uk
dtmf-relay rtp-nte
no vad
dial-peer voice 1 voip
description IncomingSIP
translation-profile incoming IncomingSIP
voice-class codec 1
session protocol sipv2
session target dns:sip.cloudcalling.co.uk
incoming called-number .T
dtmf-relay sip-notify rtp-nte
no vad
sip-ua
credentials username 4143*002 password 7 password realm sip.cloudcalling.co.uk
authentication username 4143*002 password 7 password
nat symmetric role passive
nat symmetric check-media-src
calling-info sip-to-pstn number set 4143*002
no remote-party-id
retry invite 3
retry register 3
timers connect 100
registrar dns:sip.cloudcalling.co.uk expires 60
sip-server dns:sip.cloudcalling.co.uk
host-registrar
gatekeeper
shutdown
telephony-service
load 7960-7940 P0S3-8-12-00
max-ephones 24
max-dn 30
ip source-address 192.168.1.1 port 2000
max-conferences 8 gain -6
web admin system name Admin secret 5 $1$Fktw$t9GQkdDdHmoYdwptO8.or.
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
line con 0
line aux 0
line vty 0 4
login
scheduler allocate 20000 1000
ntp server 85.119.80.232
end
Router#

You my friend are a star! worked straight away, many thanks. Just one more thing, when i make an outgoing call, it always appears as "blocked" on my phone, my sip trunk is set to allow CME to alter outgoing CLI's how would i program the outgoing CLI to 01133501788 also?
The new working config is below with your suggestion, which works!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
logging message-counter syslog
no aaa new-model
clock timezone GMT 0
dot11 syslog
ip source-route
ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 192.168.1.1
ip dhcp excluded-address 10.10.10.1
ip dhcp pool DATA_POOL
   network 10.10.10.0 255.255.255.0
   default-router 10.10.10.1
   dns-server 188.92.232.50 188.92.232.100
ip dhcp pool VOICE_POOL
   network 192.168.1.0 255.255.255.0
   default-router 192.168.1.1
   dns-server 188.92.232.50 188.92.232.100
   option 150 ip 192.168.1.1
ip name-server 188.92.232.50
ip name-server 188.92.232.100
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
registrar server
voice class codec 1
codec preference 2 g711ulaw
codec preference 3 g711alaw
voice register global
mode cme
source-address 192.168.1.1 port 5060
max-dn 144
max-pool 42
load 7960-7940 P0S3-8-12-00
authenticate register
tftp-path flash:
create profile sync 0015244443466064
voice register dn 1
number 6999
allow watch
name SIP
label SIP
voice register pool 1
id mac 000F.902B.40E0
type 7960
number 1 dn 1
dtmf-relay sip-notify
username cisco password cisco
codec g711ulaw
voice translation-rule 1
rule 1 /^6...$/ /4143*002/
voice translation-rule 3
rule 1 /^01133501788$/ /6999/
rule 2 /^1133501788$/ /6999/
voice translation-profile IncomingSIP
translate called 3
voice translation-profile Translatetrunk
translate calling 1
voice-card 0
no dspfarm
username matt privilege 15 secret 5 $1$DCD0$SjWqnKgDSGVzzIKRerXh11
archive
log config
hidekeys
interface FastEthernet0/0
ip address 194.12.0.222 255.255.255.252
ip nat outside
ip virtual-reassembly
duplex auto
speed auto
interface FastEthernet0/1
no ip address
ip nat inside
ip virtual-reassembly
duplex auto
speed auto
interface FastEthernet0/1.10
description DATA
encapsulation dot1Q 10
ip address 10.10.10.1 255.255.255.0
ip nat inside
ip virtual-reassembly
interface FastEthernet0/1.20
description VOICE
encapsulation dot1Q 20
ip address 192.168.1.1 255.255.255.0
ip nat inside
ip virtual-reassembly
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 194.12.0.221
ip http server
ip http authentication local
no ip http secure-server
ip nat inside source list 1 interface FastEthernet0/0 overload
access-list 1 permit 192.168.1.0 0.0.0.255
access-list 1 permit 10.10.10.0 0.0.0.255
tftp-server flash:P003-8-12-00.bin
tftp-server flash:P003-8-12-00.sbn
tftp-server flash:P0S3-8-12-00.loads
tftp-server flash:P0S3-8-12-00.sb2
tftp-server flash:P003-8-12-00
tftp-server flash:P003-8-12-00.loads
tftp-server flash:P003-8-12-00.sb2
tftp-server flash:SIP000F902B40E0.cnf.xml
control-plane
mgcp behavior g729-variants static-pt
dial-peer cor custom
dial-peer voice 1 voip
description IncomingSIP
translation-profile incoming IncomingSIP
voice-class codec 1
session protocol sipv2
session target sip-server
incoming called-number .T
dtmf-relay sip-notify rtp-nte
no vad
dial-peer voice 2 voip
description Outgoing Geographic
translation-profile outgoing Translatetrunk
destination-pattern 0[7]........
voice-class codec 1
session protocol sipv2
session target dns:sip.cloudcalling.co.uk
dtmf-relay rtp-nte
no vad
sip-ua
credentials username 4143*002 password 7 password realm sip.cloudcalling.co.uk
authentication username 4143*002 password 7 password
nat symmetric role passive
nat symmetric check-media-src
calling-info sip-to-pstn number set 4143*002
no remote-party-id
retry invite 3
retry register 3
timers connect 100
registrar dns:sip.cloudcalling.co.uk expires 60
sip-server dns:sip.cloudcalling.co.uk
host-registrar
gatekeeper
shutdown
telephony-service
load 7960-7940 P0S3-8-12-00
max-ephones 24
max-dn 30
ip source-address 192.168.1.1 port 2000
max-conferences 8 gain -6
web admin system name Admin secret 5 $1$Fktw$t9GQkdDdHmoYdwptO8.or.
transfer-system full-consult
create cnf-files version-stamp 7960 Dec 17 2013 14:35:13
line con 0
line aux 0
line vty 0 4
login
scheduler allocate 20000 1000
ntp server 85.119.80.232
end
Router#

Outbound Call Failure - SIP Trunk

All phones are unable to dial a single target number on the PSTN. The symptom is that it rings once and goes fast busy.
The call flow is:
Phone >>> CUCM >>> CUBE >>> Verizon SIP Trunk >>> PSTN >>> Target Number
As seen in the CUBE debug ccsip messages, the CUBE receives a "SIP/2.0 480 Temporarily unavailable" message. debug ccsip messages, dial-peer and voice class information follows:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:[email protected]>
Date: Wed, 18 Dec 2013 21:48:27 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:192.168.106.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 0520523008-0000065536-0000067523-0191539392
Session-Expires: 1800
P-Asserted-Identity: "" <sip:[email protected]>
Remote-Party-ID: "" <sip:[email protected]>;party=calling;screen=yes;privacy=off
Contact: <sip:[email protected]:5060;transport=tcp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 390
v=0
o=CiscoSystemsCCM-SIP 4037968 1 IN IP4 192.168.106.11
s=SIP Call
c=IN IP4 10.139.64.171
b=TIAS:64000
b=AS:64
t=0 0
m=audio 30688 RTP/AVP 0 8 116 18 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:116 iLBC/8000
a=ptime:20
a=maxptime:60
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Sent:
INVITE sip:[email protected]:5073 SIP/2.0
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
Remote-Party-ID: "" <sip:[email protected]>;party=calling;screen=yes;privacy=off
From: "" <sip:[email protected]>;tag=78FC5414-198D
To: <sip:[email protected]>
Date: Wed, 18 Dec 2013 21:40:10 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0520523008-0000065536-0000067523-0191539392
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1387402810
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 348
v=0
o=CiscoSystemsSIP-GW-UserAgent 4778 3356 IN IP4 10.139.64.52
s=SIP Call
c=IN IP4 10.139.64.52
t=0 0
m=audio 23372 RTP/AVP 0 8 116 18 101
c=IN IP4 10.139.64.52
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079120: Dec 18 2013 16:40:10.008: //314738/1F068D000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:[email protected]>
Date: Wed, 18 Dec 2013 21:40:09 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079121: Dec 18 2013 16:40:10.080: //314739/1F068D000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
From: "" <sip:[email protected]>;tag=78FC5414-198D
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1387402810
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079122: Dec 18 2013 16:40:11.176: //314739/1F068D000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
From: "" <sip:[email protected]>;tag=78FC5414-198D
To: <sip:[email protected]>;tag=182903799-1387403308449
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1387402810
Supported:
Contact: <sip:[email protected]:5073;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:[email protected]>;tag=78FC58A8-1B6B
Date: Wed, 18 Dec 2013 21:40:09 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
Contact: <sip:[email protected]:5060;transport=tcp>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079128: Dec 18 2013 16:40:12.384: //314739/1F068D000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 480 Temporarily unavailable
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
From: "" <sip:[email protected]>;tag=78FC5414-198D
To: <sip:[email protected]>;tag=182903799-1387403308449
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1387402810
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Sent:
SIP/2.0 480 Temporarily Not Available
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:[email protected]>;tag=78FC58A8-1B6B
Date: Wed, 18 Dec 2013 21:40:09 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=18
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079146: Dec 18 2013 16:40:12.388: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5073 SIP/2.0
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
From: "" <sip:[email protected]>;tag=78FC5414-198D
To: <sip:[email protected]>;tag=182903799-1387403308449
Date: Wed, 18 Dec 2013 21:40:10 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079147: Dec 18 2013 16:40:12.404: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:[email protected]>;tag=78FC58A8-1B6B
Date: Wed, 18 Dec 2013 21:48:27 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
dial-peer voice 9100 voip
description inboubd dial-peer for outgoing calls from CUCM (11D)
preference 1
session protocol sipv2
incoming called-number ^1..........$
voice-class codec 10
dtmf-relay rtp-nte digit-drop
ip qos dscp cs5 media
ip qos dscp cs3 signaling
no vad
outbound DP
dial-peer voice 8100 voip
description outbound dial-peer for outgoing calls to Verizon (11D)
destination-pattern ^1..........$
session protocol sipv2
session target sip-server
voice-class codec 10
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
dtmf-relay rtp-nte digit-drop
ip qos dscp cs5 media
ip qos dscp cs3 signaling
no vad
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
voice class codec 10
codec preference 1 transparent
voice class codec 2
codec preference 1 g711ulaw
codec preference 2 g722-64

I created the new voice class and mapped it to the outgoing dial-peer 8100. The call was then successful.
See new voice class:
#sh run | be voice class codec 11
voice class codec 11
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
See revised dial-peer 8100:
dial-peer voice 8100 voip
description outbound dial-peer for outgoing calls to Verizon (11D)
destination-pattern ^1..........$
session protocol sipv2
session target sip-server
voice-class codec 11
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
dtmf-relay rtp-nte digit-drop
ip qos dscp cs5 media
ip qos dscp cs3 signaling
no vad
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
My only remaining question is why did the CUBE invite NOT include the m line for g729r8?
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
See the ccapi inout snippet showing the hit with dial-peer 8100:
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
080927: Dec 19 2013 15:27:57.810: //316459/32C4F8800001/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=TRUE, Mode=0,
Outgoing Dial-peer=8100, Params=0x2B912E08, Progress Indication=NULL(0)
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
See the debug ccsip messages output showing original CUCM invite received by CUBE with 5 a line references:
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
080907: Dec 19 2013 15:27:57.806: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6d715c9c6ad1
From: "XXXXXXXXXX" ;tag=4077346~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65761788
To:
Date: Thu, 19 Dec 2013 20:36:14 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 0851769472-0000065536-0000068412-0191539392
Session-Expires: 1800
P-Asserted-Identity: "XXXXXXXXXX"
Remote-Party-ID: "XXXXXXX" ;party=calling;screen=yes;privacy=off
Contact:
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 464
v=0
o=CiscoSystemsCCM-SIP 4077346 1 IN IP4 192.168.106.11
s=SIP Call
c=IN IP4 10.139.64.52
b=TIAS:64000
b=AS:64
t=0 0
m=audio 26738 RTP/AVP 0 8 116 116 18 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:116 iLBC/8000
a=ptime:20
a=maxptime:60
a=fmtp:116 mode=20
a=rtpmap:116 iLBC/8000
a=ptime:30
a=maxptime:60
a=fmtp:116 mode=30
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
See ccsip messages output showing CUBE sending invite to Verizon:
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Sent:
INVITE sip:[email protected]:5073 SIP/2.0
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK63F9C611
Remote-Party-ID: "David Callahan" ;party=calling;screen=yes;privacy=off
From: "David Callahan" ;tag=7DE0957C-1CAB
To:
Date: Thu, 19 Dec 2013 20:27:57 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0851769472-0000065536-0000068412-0191539392
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1387484877
Contact:
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 6966 4178 IN IP4 10.139.64.52
s=SIP Call
c=IN IP4 10.139.64.52
t=0 0
m=audio 32502 RTP/AVP 0 8 101
c=IN IP4 10.139.64.52
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Call forward to PSTN on cme

Hi,
unable to set up call forward to PSTN.
/* Style Definitions */
table.MsoNormalTable
{mso-style-name:"Table Normal";
mso-tstyle-rowband-size:0;
mso-tstyle-colband-size:0;
mso-style-noshow:yes;
mso-style-priority:99;
mso-style-qformat:yes;
mso-style-parent:"";
mso-padding-alt:0cm 5.4pt 0cm 5.4pt;
mso-para-margin:0cm;
mso-para-margin-bottom:.0001pt;
mso-pagination:widow-orphan;
font-size:11.0pt;
font-family:"Calibri","sans-serif";
mso-ascii-font-family:Calibri;
mso-ascii-theme-font:minor-latin;
mso-fareast-font-family:"Times New Roman";
mso-fareast-theme-font:minor-fareast;
mso-hansi-font-family:Calibri;
mso-hansi-theme-font:minor-latin;
mso-bidi-font-family:"Times New Roman";
mso-bidi-theme-font:minor-bidi;}
I have tried activating the Call forward via the phone or manually via the config, but when I attempt a call to IP Communicator from PSTN or via extn I am not seeing re-INVITE which should be generated for the forwarded call. Am i missing something?
PSTN / IP phone ------> Calling extn on CME (which is call forwarded to another PSTN number)
config below:
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server expires max 250 min 200
asserted-id pai
localhost dns:XXXXX
outbound-proxy dns:XXXXX
dial-peer voice 100 voip
description ** Incoming call from SIP trunk **
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
dial-peer voice 101 voip
description ** Outgoinging call to SIP trunk **
translation-profile outgoing SIPOUT
destination-pattern 1T
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip profiles 101
dtmf-relay rtp-nte
no vad
dial-peer voice 102 voip
description ** Outgoinging call to SIP trunk **
destination-pattern 0[2-9].T
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
telephony-service
max-ephones 4
max-dn 12
ip source-address 192.168.100.2 port 2000
calling-number initiator
timeouts interdigit 5
load 7960-7940 P00308010200
date-format dd-mm-yy
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
transfer-system full-consult dss
transfer-pattern .T
transfer-pattern 0.T
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1 dual-line
number 4961 secondary 99474961 no-reg both
label 4961
name 4961
call-forward all 021605547

/* Style Definitions */
table.MsoNormalTable
{mso-style-name:"Table Normal";
mso-tstyle-rowband-size:0;
mso-tstyle-colband-size:0;
mso-style-noshow:yes;
mso-style-priority:99;
mso-style-qformat:yes;
mso-style-parent:"";
mso-padding-alt:0cm 5.4pt 0cm 5.4pt;
mso-para-margin:0cm;
mso-para-margin-bottom:.0001pt;
mso-pagination:widow-orphan;
font-size:11.0pt;
font-family:"Calibri","sans-serif";
mso-ascii-font-family:Calibri;
mso-ascii-theme-font:minor-latin;
mso-fareast-font-family:"Times New Roman";
mso-fareast-theme-font:minor-fareast;
mso-hansi-font-family:Calibri;
mso-hansi-theme-font:minor-latin;}
Does a direct call (without forwarding) work through this dial-peer? YES
The session target of dial-peer 101 is the "sip-server". In wich way is configured? Is it an IP address or a name? FQDN
Can you ping it from the CME? YES
The CME can resolve the name via DNS? Resolved on the CME Can you post the sip-ua config?
sip-ua
credentials number 99474960 username 99474960 password 7 XXXXXXXXX realm as-test.xys.net
authentication username 99474960 password 7 XXXXXXX
calling-info pstn-to-sip asserted-id number set 99474960
no remote-party-id
disable-early-media 180
retry invite 2
retry register 3
timers connect 100
registrar dns:as-test.xys.net expires 60 sip-server dns:as-test.xys.net
host-registrar

HELP!!! - EDI Outbound HTTP call failure

Our EDI outbound (HTTPS-OXTA) is failing since Monday in production. We narrowed down the area that might be an issue. This is what we see in the Apache log,
[Tue May 22 01:31:56 2012] [debug] opm_ew.c(469): OPM: EW: Enters opm_ew_broadcast()
[Tue May 22 01:31:56 2012] [debug] opm_ew.c(517): OPM: EW: Broadcasts msg: cmd=Broadcast&<serverName>&8001&1337662614&JServ&DiscoGroup&<server_url>&1&1&0&31490&17001;FormsGroup&<server_url>&1&1&0&31491&18001;OACoreGroup&<server_url>&1&1&0&31489&16001;XmlSvcsGrp&<server_url>&1&1&0&31492&19001
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(291): OPM:hc: Connecting to url: <server_url>:8101/oprocmgr-service
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(314): OPM:hc: Connection to host: <server_url>, port: 8101
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(438): OPM:hc: HTTP Request sent to server: POST /oprocmgr-service<server_url> HTTP/1.1^M
Host: <server_url>^M
Content-Type: application/x-www-form-urlencoded^M
Content-Length: 269^M
cmd=Broadcast&<serverName>&8001&1337662614&JServ&DiscoGroup&<server_url>&1&1&0&31490&17001;FormsGroup&<server_url>&1&1&0&31491&18001;OACoreGroup&<server_url>&1&1&0&31489&16001;XmlSvcsGrp&<server_url>&1&1&0&31492&19001
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(808): OPM:hc: headers[0] is HTTP/1.1 404 Not Found
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(808): OPM:hc: headers[1] is Date: Tue, 22 May 2012 05:31:56 GMT
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(808): OPM:hc: headers[2] is Transfer-Encoding: chunked
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(808): OPM:hc: headers[3] is Content-Type: text/html; charset=iso-8859-1
[Tue May 22 01:31:56 2012] [debug] opm_ew.c(525): OPM: EW: Broadcasts to <server_url> and send result=404
I'm trying to understand the steps of the process. Does "HTTP/1.1 404 Not Found" response to the opm_hc.c(438) call? When I type "<server_url>:8101" in the browser, I get "The webpage cannot be displayed" error. Does this should work?
EDI outbound is routed to proxy and confirmed that call from OTA was never made to proxy. Switched protocol to SMTP and it worked. There is no issue other than HTTP initial call failure. Any help you can give me I'd appreciated.

George great support so far (+5)
Hi Robert
debug ccsip all is very intensive so you should do the following before enabling the debug
service sequence-numbers
service timestamps debug datetime localtime msec
logging buffered 10000000 debug
no logging console
no logging monitor
default logging rate-limit
default logging queue-limit
Then..
<Enable debugs, then test again.>
debug ccsip all
<Enable session capture to txt file in terminal program.> (such as Putty)
then do the ff:
terminal length 0
show logging
++++
What is even more strange is that the call appears to be disconnected from the far end. From the logs below the outbound call leg (45) is where the disconnect is coming from and the "cc_api_call_disconnected" shows this call leg talking to CCAPI..
001858: *Jan 20 13:18:19.102: //45/8B56ECEE8011/CCAPI/cc_api_call_disconnected:
Cause Value=16, Interface=0x3CE6D670, Call Id=45
001859: *Jan 20 13:18:19.102: //45/8B56ECEE8011/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=TRUE, Cause Value=16, Retry Count=0)
Can you also send a debug voip ccapi onout from the CUBE. we need to check if the call arrives there, though we don't see any INVITE request sent out.

Error messages in 2651XM GW, cause outbound call failure, reboot fix it

Cisco 2651XM as Gateway, it keep posting these error message and after a period of time, it cause outbound call failure.
Reboot fix it but there're still error messages...
How to fix it? It's IOS bug or hardware issue? How to identify?
Cisco IOS Software, C2600 Software (C2600-IPVOICE-M), Version 12.3(8)T10, RELEASE SOFTWARE (fc2)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2005 by Cisco Systems, Inc.
Compiled Wed 03-Aug-05 20:45 by hqluong
ROM: System Bootstrap, Version 12.2(7r) [cmong 7r], RELEASE SOFTWARE (fc1)
cpchn1-g1 uptime is 6 hours, 56 minutes
System returned to ROM by reload at 03:52:44 NZST Tue Apr 17 2007
System restarted at 03:56:27 NZST Tue Apr 17 2007
System image file is "flash:c2600-ipvoice-mz.123-8.T10.bin"
Cisco 2651XM (MPC860P) processor (revision 0x100) with 118784K/12288K bytes of memory.
Processor board ID JAE072000AJ (1555074759)
M860 processor: part number 5, mask 2
2 FastEthernet interfaces
62 Serial interfaces
2 Channelized E1/PRI ports
32K bytes of NVRAM.
32768K bytes of processor board System flash (Read/Write)
See attach detail error messages

Cisco 2651XM as Gateway, it keep posting these error message and after a period of time, it cause outbound call failure.
Reboot fix it but there're still error messages...
How to fix it? It's IOS bug or hardware issue? How to identify?
Cisco IOS Software, C2600 Software (C2600-IPVOICE-M), Version 12.3(8)T10, RELEASE SOFTWARE (fc2)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2005 by Cisco Systems, Inc.
Compiled Wed 03-Aug-05 20:45 by hqluong
ROM: System Bootstrap, Version 12.2(7r) [cmong 7r], RELEASE SOFTWARE (fc1)
cpchn1-g1 uptime is 6 hours, 56 minutes
System returned to ROM by reload at 03:52:44 NZST Tue Apr 17 2007
System restarted at 03:56:27 NZST Tue Apr 17 2007
System image file is "flash:c2600-ipvoice-mz.123-8.T10.bin"
Cisco 2651XM (MPC860P) processor (revision 0x100) with 118784K/12288K bytes of memory.
Processor board ID JAE072000AJ (1555074759)
M860 processor: part number 5, mask 2
2 FastEthernet interfaces
62 Serial interfaces
2 Channelized E1/PRI ports
32K bytes of NVRAM.
32768K bytes of processor board System flash (Read/Write)
See attach detail error messages

Call failures when lockfs -f is ran of /opt/SMAW

During our update scenario, we initially disable local mirrors for all the local file systems like, /,/opt/SMAW,/var,/export/home.
During this disabling was call failures observed when "lockfs" was executed on the /opt/SMAW file system.
There was no call failure observed during disabling of other local file systems.
As a result of this some of the processes are blocked.
A Perf tool was run on the node to collect information like CPU.Load etc. At the time lockfs -f /opt/SMAW was executed , they was temporarily no response.
One point important here is that the disabling of all the local mirrors finished successfully.
Next a fresh test with disabling the lockfs option resulted in no call failures. So the question here is why lockfs behave like this only on /opt/SMAW .
My understanding of lockfs is -- . Lockfs with "-w" option, is used to put a "write lock" { all lock operations implicitly perform a flush } on the mirror peice on which fallback will happen if need be.So it is necessary that the data in the unmirrored part is intact.
The steps we follow while disabling are as follows:---
1> We first check if all the mirrors for a volume is in state OKAY.
2> Then it is checked in vfstab and "mount" output if the file system is mounted.
3> The lockfs will be done for the file system to ensure file systems are synced.
4> Next the first mirror is detached. fsck is ran on the mirror piece to ensure data is in good state.
Edited by: rbruby on Mar 25, 2009 11:19 PM

Thanks Lasse,
When running that tool, all the links are showing zero utilization - not sure if the tool is able to show any more detailed information?
I should state that this Lync implementation is not live, there no Lync clients out on the corporate network, and I'm not expecting any traffic on the WAN site links at this point.
It just seems to be an issue when accessing as a remote user. At this point I think it must be because I have misconfigured CAC. And because it is happening for a conference and not peer to peer for the remote user, then it is probably the relationship
between the front end server pool and Edge server / remote user.
I have a local / central site defined in CAC with no bandwidth policy applied. Initially the subnets assigned to this central site were just the LAN subnets for the central site that I knew would have Lync clients running on them. Then as part
of troubleshooting decided to add the Edge Internal IP, and then Edge external IPs as subnets associated with this central site. Still getting the same problem though.
Its around about times like now where I wonder if it is something simple like a service needing restarting so I'll double check things like this. If you have any other ideas it would be greatly appreciated - a bit stuck.
Regards,
James

Won't end call, Call Failure, Muting...

My iPhone works pretty well except for the following:
1. When I'm done with a call and go to hang up, the screen it is still blank and won't let me touch anything.
2. When someone calls and doesn't leave a voicemail it says call failure, call failed and when I slide the lever, it automatically calls them back.
3. When I'm in the middle of a call, the mute button always gets pressed.
Any advice for any of these?

I'm having the same issue with not being able to hang up after a call. I'm not using a case, and after a call, my screen wont come back to life after i take it away from my ear. I understand that there is a sensor that turns the touchscreen off when held to your face to avoid unintentional inputs, and i wonder if mine is not working, and if there is a way to test this.

Cme 2 cme voip calls work every other time

have 2 cme's with checkpoint in between.
cme A & cme B
ip phone on cme a call cue on cme b works every time
ip phone on cme a calls ip phone on cme b it works every other time consistantly. One time ringback next time busy.
phones in both locations are on different subnet from cue.
i've tried to disable ip cef to no avail and still getting problem.
debugs give disconnect cause code as network not available.
any ideas as to why it would work one time and not the next?
some sort of timer / timeout?
thanks,
paul.

You've probably mapped that key to a key command or controller assignment.
If you know what you are doing, then check the key commands and controller assignments window for unwanted MIDI mappings.
If you are not sure, then back up and move your Logic preference files and restart Logic to see if it goes away.
If it still happens, option click on File -> New to create a new song and try that. It it works OK in that, you may have something in your environment that's interpreting specific note values and doing something with them (or filtering them).

Blackberry Z10 "Call Failure"

Hi Can anyone help? My Blackberry Z10 has been working fine for a few months. This week I can not make calls nor can I take calls. When I dial out the phone displays "Call Failure" Contact your service provider. I have been in contact with Virgin and they say nothing is wrong with the account or sim. I have the sim in another blackberry phone and everything works just fine. I have done the security wipe, still the same fault. Does anyone have any ideas what I can do. I purchased the phone online "New" from a private seller.

I would get the sim replaced. I have seen similar issues with sim cards. I had a problem with my data plan on my sim. Wouldn't work on my Z10 but worked in a 9810. One think it was a problem with the device. Turned out that a replacement sim solved the problem.
Get a new sim.
1. Please thank those who help you by clicking the "Like" button at the bottom of the post that helped you.
2. If your issue has been solved, please resolve it by marking the post "Solution?" which solved it for you!

Call failures with SQN-100-5 (T-Mobile) on AT&T

I bought a factory-sealed, T-Mobile-branded Q10 on eBay a couple of months ago. At the time I was with T-Mobile, no problems with making or receiving calls. I wanted to switch to AT&T for the better coverage, so I got the unlock code from T-Mobile. Now when I try to place calls, half the time I will get a call failure message. Sometimes it takes 3 or 4 attempts to make a successful call. And a lot of calls I receive will go directly to voicemail. I spoke with someone in AT&T support and she said that, even though the phone was unlocked, because it was a -5 model instead of a -1, that I would continue to have this problem and, sorry, there was nothing they could do about it.
I'm no expert concerning cell radios & bands, but I see on the Q10 Wikipedia page that the only difference between the two models is with 3G. AT&T is tri-band, while T-Mobile is quad-band. T-Mobile also uses 1700 which AT&T doesn't.
Is this the reason for the call failures? If not, is it worth pursusing this with maybe Level 2 or Blackberry Support at AT&T to see if anything can be done on their end?

Hello,
There are multiple sub-models of each BB10 device, each supporting specific radio frequencies, bands, and technologies. The challenge in the USA is that we allowed our cellular industry to grow relatively unregulated all by itself, which has resulted in great variability between the carriers in this regard. And, to make a device actually able to handle ALL of those would mean that the device is so physically large (multiple internal antennas) as to be unacceptable to the end user. Hence, the multiple different sub-models of devices.
What you were told is indeed the case...your specific sub-model of Q10 is only partially compatible with AT&T's network, and you will indeed continue to have challenges. Unfortunately, there is nothing anyone can do to alter the (in)compatibility between your -5 device and AT&T's network.
If you truly want the best compatibility, you need to obtain a device that is fully compatible with AT&T's network. And, you of course can then sell your -5 to someone on TMO who desires it.
Good luck!
Occam's Razor nearly always applies when troubleshooting technology issues!
If anyone has been helpful to you, please show your appreciation by clicking the button inside of their post. Please click here and read, along with the threads to which it links, for helpful information to guide you as you proceed. I always recommend that you treat your BlackBerry like any other computing device, including using a regular backup schedule...click here for an article with instructions.
Join our BBM Channels
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PIN: C0001B7B4 Display/Scan Bar Code
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SOAP Call Failure

Hi All,
We are facing the below error in receiver SOAP channel.. We are on PI7.1 ..Are there any installation checks that needs to be looked at as we are facing lot of inconsistencies with SOAP...
What may be the cause of the below error...Also please let me know what all checks are needed as there are inconsistencies with SOAP adapter...
SOAP: call failed: java.net.SocketException: Connection reset
Delivering the message to the application using connection SOAP_http://sap.com/xi/XI/System failed, due to: com.sap.engine.interfaces.messaging.api.exception.MessagingException: java.net.SocketException: Connection reset.

Hi Amit,
Having the same issue :
SOAP: call failed: java.net.SocketException: Connection reset
has your issue been solved and how ?
Regards,

[svn:bz-trunk] 21661: Avoid calling throwNotSubscribedException() from inside synchronized blocks to prevent potential issues acquiring the lock .

Revision: 21661
Revision: 21661
Author:   [email protected]
Date:     2011-07-21 06:21:07 -0700 (Thu, 21 Jul 2011)
Log Message:
Avoid calling throwNotSubscribedException() from inside synchronized blocks to prevent potential issues acquiring the lock.
Checkin-Tests: Pass
QA: Yes
Doc: No
Modified Paths:
    blazeds/trunk/modules/core/src/flex/messaging/client/FlexClient.java

CME SIP phone outside call issue

Dear all,
i have cme version 9.1 on router 2921 with 7962 sccp phones and 3905 sip phone.
when i place outside call ( to pstn) using the below dial peer, call is processed.
when the call is answered by the autoattendent of the called company ( assume i called x company) , i cant press any other numbers using the sip phones.
i mean if i want to press zero for help or internal extension of the x company, these pressed numbered are not recognized by the analog panasonic PBX of the x company.
Sccp phones works well.
Any help please and below is the dial-peer.
dial-peer voice 1003 pots
trunkgroup 1
corlist outgoing CITIES
description CALLING CITIES
destination-pattern 90[1-9]......
forward-digits 8
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip handle-replaces
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/2.10
bind media source-interface GigabitEthernet0/2.10
registrar server expires max 36000 min 600
voice class codec 5
codec preference 1 g729r8
codec preference 2 g711ulaw
voice register global
mode cme
source-address 10.100.4.20 port 5060
max-dn 200
max-pool 100
load 3905 CP3905.9-2-1-0.loads
authenticate register
timezone 31
date-format D/M/Y
voicemail 177
tftp-path flash:
create profile sync 000473524028932A
conference hardware
voice register dn 1
number 109
allow watch
pickup-call any-group
pickup-group 170
shared-line max-calls 3
voice register pool 1
id mac 6C99.8984.9678
type 3905
number 1 dn 1
template 1
dtmf-relay sip-notify
voice-class codec 5
username SFD1 password SFD1
thanks

Hi Yahsiel,
firstly thanks for help, secondly if you don't mind i want to ask you the below if possible:
1- in my cme, is there a way when i call an internal extension (e.g 110) from an internal phone it rings normally but when i call from outside-->autoattendent answers-->when i press 110 it get transferred to another phone (e.g 111)....????
2- when i call from outside(pstn) to the cme -->when the plar command is directly to the internal extension the caller id appears but when the autoattendent answers and then transfer to the operator (by pressing zero) the caller id appears as unknown number ??????
3- is the 3905 sip phone support 1Gbps when connected to the PC, as after connecting the phones to the PCs the speed decreased up to 100Mbps?? or it is another matter?
(poe switches is cisco SG200)
regards,

About cme 4.x, SIP softphone and g729

Hi folks,
it's normal that a x-lite softphone (mac os x version) registered on CME 4.x system:
- calls correctly all internal numbers (ip phones Cisco)
- the client supports only g711a/u, GSM, iLBC and SPX ... then it uses g711a/u ... but all dial-peer voip use g729, then when I try to call a voip number, cme try to connect that with dial-peer pots, and it doesn't work ... no way to do that? I could only:
1- change dial-peer voip codec to g711
2- change softphone with g729 support
any advice will be appreciated
Regards
Andrea

It is possible to register a sip softphone (like x-lite, sjphone, etc) and make a call via voip trunk. This URL should help you:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00802df071.html

SIP to SIP Call Failures on CME to CME - sip-ua conflict/issue?

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