SiP Phone wont dial inbound or outbound
I have 9971 phone and was dialing sip to sip and sip h323 on the network, but now I get Apr 20 14:17:58.911: %VOICE_IEC-3-GW: Application Framework Core: Internal error
(Toll fraud call rejected): IEC=1.1.228.3.31.0 on callID 12 GUID=BCBC7FI
Hi Wharrison,
can you please provide the call flow and where do you see this error.
I am guessing the call is from an IP phone regsitered to CUCM --> SIP truk --> CUBE --> provider.. Is this right?
Please let me know where do you see the error.
Thanks,
Manoj
Similar Messages
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I live in California, daughter is visiting New York and her phone will not dial out. Any suggestions she has tried dialing 1+ area code + number and it still wont work.
guys thanks for those who view my post.
the problem was a bad phone. -
Auto-Dial from SIP phones after Interdigit Timeout
I have a customer that needs to have their phones automatically dial a particular number if no digits are pressed after 10-15 seconds. We are accomplishing this right now with our SCCP phones by having a blank translation pattern that points to an ambiguous hunt pilot.
For example, we create a blank translation pattern that translates to a HP with number 300. We also create a HP with number 3000. The translation pattern is triggered, but because 300 is ambiguous (the call could possibly be destined for 300 or 3000) CUCM waits for the interdigit timeout before calling 300.
This works beautifully for SCCP phones. Unfortunately, it does not work for our SIP phones. Does anyone have any suggestions to make this work? SIP dial rules trigger immediately, so I'm afraid we haven't been able to leverage those.
Thank You!Thanks for your comments, but I'm afraid you guys are missing the issue.
Chris, you are correct that you can get a SIP phone to auto-dial a number via SIP dial rules. The problem is that it triggers immediately. I need the SIP phones to wait.
Here's a scenario - SCCP phones are in use all throughout a nursing home. Residents are told that they can simply knock their phone offhook if they need emergency help. If they don't dial anything within 15 seconds, their phone will automatically call into the local emergency oncall nurse. Sorta an "onstar" emergency help line for an old folks home. We definitely can't have that rule triggering immediately though, as residents would never be able to place a normal call, and there would be no way of distinguishing the emergency calls from the guy who is just trying to call for pizza. :)
This function works perfectly with SCCP phones - while they trigger the blank translation pattern immediately, CUCM doesn't actually begin dialing because there are multiple potential matches for what the translation pattern is dialing. SIP phones do not do this, however. If you have the SIP dial rule applied, it always triggers immediately, and it does not wait to see if callers give any input.
Thanks for any further suggestions you might be able to offer! -
Incoming calls issue in Third Party SIP Phone
Hi,
Yesterday I configured my third party sip phone which is yealink in this case on cucm and successfully registered it with cucm, despite of registration i have some calling issue in this phone. I am able to make outbound calls from this phone to any other phone however issue is related to inbound calls.I tried calling its DN from anywhere but call disconnect after sometime. Also didnt get any proper sip session trace in RTMT. Kindly suggest some step to sortout this issue.
ThanksDear Manish,
Call normally dicsonnected after 30-40 sec with termination code 102 in session trace. PFB SDI trace with 5030 is Thirdparty sip phone and 5033 is c7945. Looking forward for your suggestion.
CallingPartyNumber=5033
|DialingPartition=
|DialingPattern=5030
|FullyQualifiedCalledPartyNumber=5030
|DialingPatternRegularExpression=(5030)
|DialingWhere=
|PatternType=Enterprise
|PotentialMatches=NoPotentialMatchesExist
|DialingSdlProcessId=(0,0,0)
|PretransformDigitString=5030
|PretransformTagsList=SUBSCRIBER
|PretransformPositionalMatchList=5030
|CollectedDigits=5030
|UnconsumedDigits=
|TagsList=SUBSCRIBER
|PositionalMatchList=5030
|VoiceMailbox=
|VoiceMailCallingSearchSpace=PT-LHR-LOCAL:PT-Local:Unityvmpt:PT-F6-Local:PT-ISL-LOCAL:PT-KHI-LOCAL:PT_Operator_LHR:PT_Operator_KHI:PT_Operator_ISL
|VoiceMailPilotNumber=7103
|RouteBlockFlag=RouteThisPattern
|RouteBlockCause=0
|AlertingName=Syed Ahmer
|UnicodeDisplayName=Syed Ahmer
|DisplayNameLocale=1
|OverlapSendingFlagEnabled=0
12:17:38.028 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.16.200.21:[5062]:
[23928282,NET]
INVITE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.11:5060;branch=z9hG4bK1ca0cc6e317649
From: "Syed Ahmer" ;tag=8787406~039e2a80-8561-4586-8954-d01ed2aa12c8-246211918
To:
Date: Thu, 30 Jan 2014 07:17:38 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Send-Info: conference, x-cisco-conference
Alert-Info:
Contact:
Remote-Party-ID: "Syed Ahmer" ;party=calling;screen=yes;privacy=off
Max-Forwards: 70
Content-Length: 0
|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
12:17:38.028 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xaf299320|*^*^*
12:17:38.028 |EnvProcessUdpPort::fireSignal - SEND, destination = 172.16.200.21:5062|*^*^*
12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 850, 172.16.200.21:5062)|*^*^* -
Hi all,
Just looking for some clarification on this one. I'm seeing some conflicting advice about setting the DTMF-Relay on SIP Phones registered to CME with a CUE Module. I've read some documentation indicating that rtp-nte RFC2833 is the only dtmf-relay supported for SIP Phones registered to CME, however I've also read some documents indicating that sip-notify must be configured as the dtmf-relay on SIP phones when they are communicating to a CUE module. I'm assuming I'm going to need to configure an MTP on the CME, but just wondering what the official DTMF config should be under the voice register pool for SIP phones.
Thanks!Hi logan
When doing lab with cme 7.0 and sip phones .sip phones are not recognizing the "sip-notify" dtmf-relay method .It can only recognize "rtp-nte" method and it does not matter weather you are using sip-notify or rtp-nte for a dial-peer pointing to cme .
i configured on cue
ccn subsystem sip
dtmf-relay sip-notify
end
on cme i configured a dial-peer pointing to cue
dial-peer v 3888 voip
destination-pattern 3888
session target ipv4:177.3.11.10
codec g711ulaw
no vad
session protocol sipv2
dtmf-relay sip-notiy
on my sip phones
voice register pool 1
dtmf-relay sip-notify ------> now in this case cue wont recognize dtmf tones
when i change this dtmf-relay method to rtp-nte it recognizes dtmf tones to when recording a message -
Cisco SIP Phone 9971 won't register on CME 8.6
Hello,
I'm facing a very strange problem:
a Cisco SIP Phone 9971 won't register on CME 8.6 running on a 2811
I have read all the related-postings to this and other Forum, but I have not been able to solve it.
One of the "potential solutions" was to make sure that the Phone had a Line configured.
But I think that the commands voice register dn and voice register pool are properly configured (see config below)
So frankly, I have no idea what I could be missing.
I'm pasting the Router's config.
I hope somebody is able to point me in the right direction.
Here is the config. Thank you!
C2811#sh run
Building configuration...
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname C2811
no aaa new-model
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 172.25.140.1 172.25.140.10
ip dhcp excluded-address 172.35.140.1 172.35.140.10
ip dhcp pool Data
network 172.25.140.0 255.255.255.0
default-router 172.25.140.1
option 150 ip 172.25.140.1
dns-server 172.25.140.1
ip dhcp pool Voice
network 172.35.140.0 255.255.255.0
default-router 172.35.140.1
option 150 ip 172.35.140.1
dns-server 172.35.140.1
no ip domain lookup
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections sip to sip
sip
registrar server expires max 3600 min 120
voice register global
mode cme
source-address 172.25.140.1 port 5060
max-dn 40
max-pool 42
load 9971 sip9971.9-4-1-9.loads
authenticate register
authenticate realm cisco
tftp-path flash:
create profile sync 0004820400584603
voice register dn 1
number 1010
allow watch
name Phone10
label Phone10
mwi
voice register pool 1
id mac 189C.5DB6.BD09
type 9971
number 1 dn 1
presence call-list
dtmf-relay rtp-nte
username adm password adm
call-forward b2bua busy 68600
codec g711ulaw
no vad
camera
video
voice-card 0
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-1879153754
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-1879153754
revocation-check none
rsakeypair TP-self-signed-1879153754
crypto pki certificate chain TP-self-signed-1879153754
certificate self-signed 01
(details ommited)
license udi pid CISCO2811 sn FTX1146A44H
username admin privilege 15 password 0 admin
redundancy
interface FastEthernet0/0
no ip address
duplex auto
speed auto
interface FastEthernet0/0.25
description Data VLAN
encapsulation dot1Q 25
ip address 172.25.140.1 255.255.255.0
interface FastEthernet0/0.35
description Voice VLAN
encapsulation dot1Q 35
ip address 172.35.140.1 255.255.255.0
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 600 life 86400 requests 10000
tftp-server flash:P00308010200.bin
tftp-server flash:P00308010200.sbn
tftp-server flash:P00308010200.sb2
tftp-server flash:P00308010200.loads
tftp-server flash:SCCP42.9-3-1SR3-1S.loads
tftp-server flash:apps42.9-3-1ES19.sbn
tftp-server flash:cnu42.9-3-1ES19.sbn
tftp-server flash:cvm42sccp.9-3-1ES19.sbn
tftp-server flash:dsp42.9-3-1ES19.sbn
tftp-server flash:jar42sccp.9-3-1ES19.sbn
tftp-server flash:term42.default.loads
tftp-server flash:term62.default.loads
tftp-server flash:SCCP45.9-3-1SR3-1S.loads
tftp-server flash:apps45.9-3-1ES19.sbn
tftp-server flash:cnu45.9-3-1ES19.sbn
tftp-server flash:cvm45sccp.9-3-1ES19.sbn
tftp-server flash:dsp45.9-3-1ES19.sbn
tftp-server flash:jar45sccp.9-3-1ES19.sbn
tftp-server flash:term45.default.loads
tftp-server flash:term65.default.loads
tftp-server flash:/Ringtones/Ringlist.xml alias Ringlist.xml
tftp-server flash:/Ringtones/DistinctiveRingList.xml alias DistinctiveRingList.x
ml
tftp-server flash:sip9971.9-4-1-9.loads
tftp-server flash:kern9971.9-4-1-9.sebn
tftp-server flash:rootfs9971.9-4-1-9.sebn
tftp-server flash:dkern9971.100609R2-9-4-1-9.sebn
tftp-server flash:sboot9971.031610R1-9-4-1-9.sebn
tftp-server flash:skern9971.022809R2-9-4-1-9.sebn
tftp-server flash:/g4-tones.xml alias United_States/g4-tones.xml
tftp-server flash:/gd-sip.jar alias English_United_States/gd-sip.jar
control-plane
mgcp profile default
telephony-service
max-ephones 24
max-dn 48
ip source-address 172.25.140.1 port 2000
cnf-file location flash:
load 7960-7940 P00308010200
load 7942 SCCP42.9-3-1SR3-1S.loads
load 7945 SCCP45.9-3-1SR3-1S.loads
load 7962 SCCP42.9-3-1SR3-1S.loads
load 7965 SCCP45.9-3-1SR3-1S.loads
max-conferences 8 gain -6
dn-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 Feb 11 2014 07:18:32
ephone-dn 1
number 1001
description Phone 1
name Phone 1
hold-alert 30 originator
ephone-dn 2
number 1002
description Phone 2
name Phone 2
hold-alert 30 originator
ephone-dn 3
number 1003
description Phone 3
name Phone 3
hold-alert 30 originator
ephone 1
device-security-mode none
mac-address 001C.58FB.6E0F
button 1:1
ephone 2
device-security-mode none
mac-address 0014.A981.7F8A
button 1:2
ephone 3
device-security-mode none
mac-address 0006.5356.A4B8
button 1:3
alias exec con conf t
alias exec sib show ip int brief
alias exec srb show run | b
alias exec sri show run int
line con 0
exec-timeout 0 0
logging synchronous
line aux 0
line vty 0 4
privilege level 15
login local
transport input telnet ssh
transport output telnet ssh
line vty 5 15
privilege level 15
login local
transport input telnet ssh
transport output telnet ssh
scheduler allocate 20000 1000
ntp master 1
end
C2811#Thank you for your reply.
I did some debugs and the results are very strange!
This is what I got:
Feb 24 18:01:12.219: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 172.35.140.12:5060;branch=z9hG4bK08011844
From: ;tag=189c5db6bd09000260cf3daf-289a76d1
To: ;tag=52488-160A
Date: Mon, 24 Feb 2014 18:01:12 GMT
Call-ID: [email protected]
CSeq: 1000 REFER
Content-Length: 0
Contact:
Feb 24 18:01:12.291: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
REGISTER sip:172.25.140.1 SIP/2.0
Via: SIP/2.0/UDP 172.35.140.12:5060;branch=z9hG4bK1e9ad079
From: ;tag=189c5db6bd0900032df02e9c-25d79707
To:
Call-ID: [email protected]
Max-Forwards: 70
Date: Fri, 01 Jan 1982 00:02:41 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP9971/9.4.1
Contact: ;+sip.instance="
000000-0000-0000-0000-189c5db6bd09>";+u.sip!devicename.ccm.cisco.com="SEP189C5DB
6BD09";+u.sip!model.ccm.cisco.com="493";video
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-
cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-
cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.2,X-cisco-xsi-
8.0.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:22 Name=SEP189C5DB6BD09 ActiveLoad=sip99
71.9-4-1-9.loads InactiveLoad=sip9971.9-3-2SR1-1.loads Last=reset-reset"
Expires: 3600
Feb 24 18:01:12.395: voice_reg_get_reg_expires_timer: no voice register pool found
Feb 24 18:01:12.395: VOICE_REG_POOL: Register request for (1010) from (172.35.140.12)
Feb 24 18:01:12.395: VOICE_REG_POOL: Contact matches pool 1 number list 1
Feb 24 18:01:12.395: VOICE_REG_POOL: No entry for (172.35.140.12) found in srst contact table
Feb 24 18:01:12.395: VOICE_REG_POOL: key(1010) contact(172.35.140.12:5060) add to contact table
Feb 24 18:01:12.395: VOICE_REG_POOL: No entry for (1010) found in contact table
Feb 24 18:01:12.399: VOICE_REG_POOL: key(1010) contact(172.35.140.12) added to contact table
Feb 24 18:01:12.399: VOICE_REG_POOL: key(172.35.140.12) contact(1010) add to srst contact table
Feb 24 18:01:12.399: VOICE_REG_POOL: No entry for (172.35.140.12) found in srst contact table
Feb 24 18:01:12.399: VOICE_REG_POOL: key(172.35.140.12) contact(1010) added to srst contact table
Feb 24 18:01:12.399: VOICE_REG_POOL pool->tag(1), dn->tag(1), submask(1)
But right after these errors, I get the following:
Feb 24 18:01:12.399: VOICE_REG_POOL: Creating param container for dial-peer 4000
1.VOICE_REG_POOL pool->tag(1), dn->tag(1), submask(1)
VOICE_REG_POOL pool_tag(1), dn_tag(1)
Feb 24 18:01:12.399: VOICE_REG_POOL: Created dial-peer entry of type 0
Feb 24 18:01:12.399: VOICE_REG_POOL: Registration successful for 1010, registration id is 1
Feb 24 18:01:12.411: VOICE_REG_POOL: Contact matches pool 1 number list 1
Feb 24 18:01:12.411: VOICE_REG_POOL: GW SIS: X-cisco-cme-sis-1.0.0
Feb 24 18:01:12.411: VOICE REGISTER POOL-1 has registered.
Name:SEP189C5DB6BD09 IP:172.35.140.12 DeviceType:Phone
Feb 24 18:01:12.411: VOICE_REG_POOL: Pool[1]: service-control (reset type: 2) message sent to sip:[email protected]
Feb 24 18:01:12.411: voice_reg_privacy_update_to_phone: delay sending privacy update during bulk registration
Feb 24 18:01:12.415: //1/7B0070C28003/SIP/Msg/ccsipDisplayMsg:
====================
And when I do a sh voice register pool, I get the following:
C2811#sh voice register pool 1
Pool Tag 1
Config:
Mac address is 189C.5DB6.BD09
Type is 9971
Number list 1 : DN 1
Proxy Ip address is 0.0.0.0
Current Phone load version is Cisco-CP9971/9.4.1
DTMF Relay is enabled, rtp-nte
Call Waiting is enabled
DnD is disabled
Video is enabled
Camera is enabled
Busy trigger per button value is 0
call-forward b2bua busy 68600
keep-conference is enabled
registration expires timer max is 3600 and min is 120
username adm password adm
kpml signal is enabled
Lpcor Type is none
blf call list is enabled
Transport type is udp
service-control mechanism is supported
registration Call ID is [email protected]
Registration method: per line
Privacy feature is not configured.
Privacy button is disabled
active primary line is: 1010
contact IP address: 172.35.140.12 port 5060
Phone SIS Version: 6.0.2
GW SIS Version: 1.0.0
Dialpeers created:
Dial-peers for Pool 1:
dial-peer voice 40001 voip
destination-pattern 1010
session target ipv4:172.35.140.12:5060
session protocol sipv2
dtmf-relay rtp-nte
digit collect kpml
codec g711ulaw bytes 160
no vad
call-fwd-busy 68600
after-hours-exempt FALSE
Statistics:
Active registrations : 4
Total SIP phones registered: 1
Total Registration Statistics
Registration requests : 4
Registration success : 4
Registration failed : 0
unRegister requests : 0
unRegister success : 0
unRegister failed : 0
Attempts to register
after last unregister : 0
Last register request time : 18:11:43.551 UTC Mon Feb 24 2014
Last unregister request time :
Register success time : 18:11:43.551 UTC Mon Feb 24 2014
Unregister success time :
C2811#
So apparently the Phone is actually registered!
However, the Phone screens still shows this message: Phone Not Registered.
So frankly I don't understand what's going on!
I really hope somebody can help. Thanks! -
Cisco SIP Phone 9971 will not register on CME 8.6
Hello,
I'm trying to configure a Cisco SIP Phone 9971,
but it won't register on CME 8.6, which is running on a 2811
The Phone shows this error message: Phone Not Registered.
And when I check the the Status Messages in the Phone, I see the following:
VPN Error: vpn is not configured
Actually, it shows all these 4 messages in a constant Loop:
12:01:59a SEP189C5DB6BD09.cnf.xml (TFTP)
12:01:59a No Trust List instaled
12:01:59a Updating Trust list
12:02:00a VPN Error: VPN is not Configured
It seems that this VPN Error is keeping the Phone from registering.
This is repeated for ever and the Phone never registers; at least that's what it appears.
However, when I do a sh voice register pool, I get the following:
C2811#sh voice register pool 1
Pool Tag 1
Config:
Mac address is 189C.5DB6.BD09
Type is 9971
Number list 1 : DN 1
Proxy Ip address is 0.0.0.0
Current Phone load version is Cisco-CP9971/9.4.1
DTMF Relay is enabled, rtp-nte
Call Waiting is enabled
DnD is disabled
Video is enabled
Camera is enabled
Busy trigger per button value is 0
call-forward b2bua busy 68600
keep-conference is enabled
registration expires timer max is 3600 and min is 120
username adm password adm
kpml signal is enabled
Lpcor Type is none
blf call list is enabled
Transport type is udp
service-control mechanism is supported
registration Call ID is [email protected]
Registration method: per line
Privacy feature is not configured.
Privacy button is disabled
active primary line is: 1010
contact IP address: 172.35.140.12 port 5060
Phone SIS Version: 6.0.2
GW SIS Version: 1.0.0
Dialpeers created:
Dial-peers for Pool 1:
dial-peer voice 40001 voip
destination-pattern 1010
session target ipv4:172.35.140.12:5060
session protocol sipv2
dtmf-relay rtp-nte
digit collect kpml
codec g711ulaw bytes 160
no vad
call-fwd-busy 68600
after-hours-exempt FALSE
Statistics:
Active registrations : 4
Total SIP phones registered: 1
Total Registration Statistics
Registration requests : 4
Registration success : 4
Registration failed : 0
unRegister requests : 0
unRegister success : 0
unRegister failed : 0
Attempts to register
after last unregister : 0
Last register request time : 18:11:43.551 UTC Mon Feb 24 2014
Last unregister request time :
Register success time : 18:11:43.551 UTC Mon Feb 24 2014
Unregister success time :
C2811#
This sh voice register pool seems to indicate that the Phone has actually registered.
But I still get the Phone Not Registered message on the screen!
I did some Debugs and they also seem to indicate that the Phone has indeed registered:
Feb 24 18:01:12.399: VOICE_REG_POOL: Creating param container for dial-peer 4000
1.VOICE_REG_POOL pool->tag(1), dn->tag(1), submask(1)
VOICE_REG_POOL pool_tag(1), dn_tag(1)
Feb 24 18:01:12.399: VOICE_REG_POOL: Created dial-peer entry of type 0
Feb 24 18:01:12.399: VOICE_REG_POOL: Registration successful for 1010, registration id is 1
Feb 24 18:01:12.411: VOICE_REG_POOL: Contact matches pool 1 number list 1
Feb 24 18:01:12.411: VOICE_REG_POOL: GW SIS: X-cisco-cme-sis-1.0.0
Feb 24 18:01:12.411: VOICE REGISTER POOL-1 has registered.
Name:SEP189C5DB6BD09 IP:172.35.140.12 DeviceType:Phone
So frankly, I have no idea why the Phone keeps showing the Phone Not Registered message.
I'm pasting the Router's config.
I hope somebody is able to point me in the right direction.
Here is the config. Thank you!
C2811#sh run
Building configuration...
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname C2811
no aaa new-model
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 172.25.140.1 172.25.140.10
ip dhcp excluded-address 172.35.140.1 172.35.140.10
ip dhcp pool Data
network 172.25.140.0 255.255.255.0
default-router 172.25.140.1
option 150 ip 172.25.140.1
dns-server 172.25.140.1
ip dhcp pool Voice
network 172.35.140.0 255.255.255.0
default-router 172.35.140.1
option 150 ip 172.35.140.1
dns-server 172.35.140.1
no ip domain lookup
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections sip to sip
sip
registrar server expires max 3600 min 120
voice register global
mode cme
source-address 172.25.140.1 port 5060
max-dn 40
max-pool 42
load 9971 sip9971.9-4-1-9.loads
authenticate register
authenticate realm cisco
tftp-path flash:
create profile sync 0004820400584603
voice register dn 1
number 1010
allow watch
name Phone10
label Phone10
mwi
voice register pool 1
id mac 189C.5DB6.BD09
type 9971
number 1 dn 1
presence call-list
dtmf-relay rtp-nte
username adm password adm
call-forward b2bua busy 68600
codec g711ulaw
no vad
camera
video
voice-card 0
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-1879153754
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-1879153754
revocation-check none
rsakeypair TP-self-signed-1879153754
crypto pki certificate chain TP-self-signed-1879153754
certificate self-signed 01
(details ommited)
license udi pid CISCO2811 sn FTX1146A44H
username admin privilege 15 password 0 admin
redundancy
interface FastEthernet0/0
no ip address
duplex auto
speed auto
interface FastEthernet0/0.25
description Data VLAN
encapsulation dot1Q 25
ip address 172.25.140.1 255.255.255.0
interface FastEthernet0/0.35
description Voice VLAN
encapsulation dot1Q 35
ip address 172.35.140.1 255.255.255.0
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 600 life 86400 requests 10000
tftp-server flash:P00308010200.bin
tftp-server flash:P00308010200.sbn
tftp-server flash:P00308010200.sb2
tftp-server flash:P00308010200.loads
tftp-server flash:SCCP42.9-3-1SR3-1S.loads
tftp-server flash:apps42.9-3-1ES19.sbn
tftp-server flash:cnu42.9-3-1ES19.sbn
tftp-server flash:cvm42sccp.9-3-1ES19.sbn
tftp-server flash:dsp42.9-3-1ES19.sbn
tftp-server flash:jar42sccp.9-3-1ES19.sbn
tftp-server flash:term42.default.loads
tftp-server flash:term62.default.loads
tftp-server flash:SCCP45.9-3-1SR3-1S.loads
tftp-server flash:apps45.9-3-1ES19.sbn
tftp-server flash:cnu45.9-3-1ES19.sbn
tftp-server flash:cvm45sccp.9-3-1ES19.sbn
tftp-server flash:dsp45.9-3-1ES19.sbn
tftp-server flash:jar45sccp.9-3-1ES19.sbn
tftp-server flash:term45.default.loads
tftp-server flash:term65.default.loads
tftp-server flash:/Ringtones/Ringlist.xml alias Ringlist.xml
tftp-server flash:/Ringtones/DistinctiveRingList.xml alias DistinctiveRingList.x
ml
tftp-server flash:sip9971.9-4-1-9.loads
tftp-server flash:kern9971.9-4-1-9.sebn
tftp-server flash:rootfs9971.9-4-1-9.sebn
tftp-server flash:dkern9971.100609R2-9-4-1-9.sebn
tftp-server flash:sboot9971.031610R1-9-4-1-9.sebn
tftp-server flash:skern9971.022809R2-9-4-1-9.sebn
tftp-server flash:/g4-tones.xml alias United_States/g4-tones.xml
tftp-server flash:/gd-sip.jar alias English_United_States/gd-sip.jar
control-plane
mgcp profile default
telephony-service
max-ephones 24
max-dn 48
ip source-address 172.25.140.1 port 2000
cnf-file location flash:
load 7960-7940 P00308010200
load 7942 SCCP42.9-3-1SR3-1S.loads
load 7945 SCCP45.9-3-1SR3-1S.loads
load 7962 SCCP42.9-3-1SR3-1S.loads
load 7965 SCCP45.9-3-1SR3-1S.loads
max-conferences 8 gain -6
dn-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 Feb 11 2014 07:18:32
ephone-dn 1
number 1001
description Phone 1
name Phone 1
hold-alert 30 originator
ephone-dn 2
number 1002
description Phone 2
name Phone 2
hold-alert 30 originator
ephone-dn 3
number 1003
description Phone 3
name Phone 3
hold-alert 30 originator
ephone 1
device-security-mode none
mac-address 001C.58FB.6E0F
button 1:1
ephone 2
device-security-mode none
mac-address 0014.A981.7F8A
button 1:2
ephone 3
device-security-mode none
mac-address 0006.5356.A4B8
button 1:3
alias exec con conf t
alias exec sib show ip int brief
alias exec srb show run | b
alias exec sri show run int
line con 0
exec-timeout 0 0
logging synchronous
line aux 0
line vty 0 4
privilege level 15
login local
transport input telnet ssh
transport output telnet ssh
line vty 5 15
privilege level 15
login local
transport input telnet ssh
transport output telnet ssh
scheduler allocate 20000 1000
ntp master 1
end
C2811#VPN is not Configured prints on all phones now with the built-in VPN client if VPN isn't configured. That's normal and is just cosmetic. That should not be causing your registration issues.
-
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
thanksHi 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, -
Bandwidth required during registration of ip sip phone
what is the bandwidth requirement of cisco 9971 model phone during registration?
Thanks a lot Vivek... I went through it...but I think it talks about the bandwidth provisioning for voice traffic..for signalling and once the call is established....I neeed to know the bandwidth required just for registration ...i.e. in the following steps ( only registration without placing any call)
1. The phone contacts the TFTP server and requests the Certificate Trust List file .
2. The phone contacts the TFTP server and requests its SEP<mac-address>.cnf.xml configuration file.
3. The Phone downloads the default configuration XMLDefault.cnf.xml file from the TFTP server.
4. The SIP phone requests a firmware upgrade (Load ID file) and upgrades the firmware image automatically when required for a new version of CUCM.
5. The phone downloads the SIP dial rules configured for that phone.
6. The phone Establish connection with the primary CUCM and the TFTP server end to end.
7. The phone Registers with the primary CUCM server listed in its configuration file.
8. The phone downloads the appropriate localization files from TFTP.
9. The phone downloads the softkey configurations from TFTP.
10. The phone downloads custom ringtones (if any) from TFTP.
Also, I need t o know if the bandwidth required for this process is same for all phone models or different? Specifically, I need this data for Cisco 9971 model.Please help...Thanks.. -
SIP- h323 in a AS5850 - Not able to send h323 calls coming from a SIP Phone
Dear All!
I have an AS5850 configured as a SIP Gateway and as a H323 Gateway. I'm planning to use this equipment as an interconnection point between PSTN,SIP and H323.
I already have a functional H323 Network with ISDN trunks to the pstn and it is working fine. I added SIP configuration to the AS5850 in order to be able to route calls out to the PSTN or H323 remote ends coming from a SIP Phone registered with a third-party SIP Proxy.
When the calls coming from the SIP Phone goes to a PSTN destination the calls completes properly, but i am having problems trying to send calls coming from the SIP phone to a remote h323 gateway(also cisco)
Attached is my configuration and the error i'm getting in my cdr. It seems that the "ext" number of the phone is being used as destination string in the last call leg, but i'm not sure.
Please Help!
dial-peer voice 100 pots
application session
destination-pattern 5T
port 2/6:D
forward-digits all
dial-peer voice 102 pots
application session
destination-pattern 044T
port 2/6:D
forward-digits all
dial-peer voice 103 voip
application session
incoming called-number 001T
destination-pattern 001T
session protocol sipv2
session target ipv4:20X.21X.17X.1X
tech-prefix 10511
sip-ua
sip-server ipv4:20X.6X.14X.18X
CDR ERROR:
.Mar 24 2004 18:31:42.620 GMT: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 2, ConnectionId 9F74CE17 7D2A11D8 82A09B41 D2C3D418, SetupTime .18:31:42.470 GMT Wed Mar 24 2004, ***PeerAddress 2006***, PeerSubAddress , DisconnectCause 3 , DisconnectText no route to destination (3), ConnectTime .18:31:42.620 GMT Wed Mar 24 2004, DisconnectTime .18:31:42.620 GMT Wed Mar 24 2004, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0, ReceivePackets 0, ReceiveBytes 0
Thanks.
Attached you can find the debug ccsip messages output.There are 2 solutions here.
1. Use of SIP/H.323 Signalling Gateway as the protocol convertor. Search google will yield heaps of hits on this subject. Product available both commercial and open source, trial, etc. Using this method means that the SIP End Point will communicate with H.323 End Point without going out the PSTN. I believe this is what you want to achieve in the long term. You are trying the AS5xxx as the protocol convertor for you, which it will not work. A call flow will be something like SIP IP Phone->SIP Server->SIP-to-H.323 Gateway->H.323 Gatekeeper->H.323 End Point. Of couse there is a SIP server that do the protocol convertor in the same box but the functionality is the still the same. Performance and concurrent call setup differ from products to products. Going for this solution would require you to find such products and test it on the your network.
2. If you do not wish to try on Soluton 1, this solution is a workaround way by not getting device but using the existing equipment that you have right now. Onto whether this good long term solution for depends on what you want to achieve both in term of commercially and technically. A call flow will be SIP End Point->SIP Server->Voice Gateway (AS5xxx)->PSTN Switch(ISDN/PRI)->Voice Gateway->H.323 Gatekeeper>H.323 End Point. The key is the Voice session must traverse the ISDN link. In other words your dial pattern must be setup is such as way that will go out thru the dial peer pots to pstn switch then come back to another dial-peer pots. I am not saying this is the most efficient way of doing it, I merely suggesting a workable way to achieve your desired goal without soluton 1.
Hopes you get better understanding now.
Thanks
SSng -
Changed number online now phone wont work what do i do?
my husband went on line and changed to a local number - now the phone wont work... i dont know whats wrong but hes stuck in flordia working third shift and im in sc and the account holder. everytim i call it it goes straight to voicemail and he has messaged me via the computer that it wont allow him to dial out.HELP PLEASE
Did you try shutting the phone off and then re-powering it with any results.. ? Give that a try see what it does it may help push the Number through or Try this Dial * 228 send then 1 to reactivate it try that see if that works.. Let me know if it did or not I'm on the forum for a while this morning.. b33
-
Phones wont signin in after CA upgrade
Hi All,
I currently have an issue when Lync Phone Edition and Polycom VVX model phones wont sign in to our Lync 2010 infrastructure.
I see the Phone attempt registration, and just get an unauthorised message and stops trying, only a single time not like the Lync client get unauthorised the 3 times.
I have tried both via PIN Auth, and via USB connectivity same result either way.
I am looking at certificate as the cause, as we re-issued certificates to all Lync servers right before the issue occurred (new CA in the environment).
A "Test-CsPhoneBootstrap" gives a success result, but when attempted from the phones nothing.
I have been checking the logs on the phones and can see some errors but not sure where to look next.
An Extract of the logs from the LPE device:
+++++++++++++++++++++++++++++++++++++
ERROR :: WebServices::CManagedWebRequest::ExecuteInternal: Webrequest failed with non-continuable error, hr=0x80004005
WARN :: NModel::CCertificateProvider::GetDiagCode: Neither Ms-diagnostic or soap fault is found.
WARN :: NModel::CUserAccountProfiles::OnEvent: Bootstrapper failed: status=3, hr=0x80004005, component will be stopped
ERROR :: NModel::CBootstrapTask::OnExecution: Error: hr = 80004005 - Failed to resolve user sip uri: phone=186!
+++++++++++++++++++++++++++++++++++++
And when tracing the UserPinService on the Lync FE
+++++++++++++++++++++++++++++++++++++
Component: UserPinService
Level: TL_VERBOSE
Flag: TF_COMPONENT
Function: TimedHash<T_V>.TimerCallback
Source: timedhash.cs(269)
Local Time: 12/10/2013-19:42:01.780
Sequence# : 00001B95
CorrelationId :
ThreadId : 16D4
ProcessId : 1068
CpuId : 2
Original Log Entry :
TL_VERBOSE(TF_COMPONENT) [2]1068.16D4::12/10/2013-08:42:01.780.00001b95 (UserPinService,TimedHash<T_V>.TimerCallback:timedhash.cs(269))(00000000005A137D)key :
[email protected]
+++++++++++++++++++++++++++++++++++++
I have seem some posts about factory resetting the phones for when the root CA has changed, we have done that and still nothing.
I have also seen the posts about updating the rootca store via the New-CsWebTrustedCACertificate, but from my reading of the set-cswebserviceconfiguration that InferCertChainFromSS is set to true (our server is) the
TrustedCACerts is ignored, so I don't think this would do anything as we should be downloading the certificate chain anyway..
Any ideas would be appreciated.
Cheers
Jason
My UC ThoughtsHi,
Is there any other model of Lync phone in your environment? Does the issue happen for all Lync phone devices?
Please check your automatic configuration record as Lync Phone Edition cannot utilize Manual Configuration.
Here is a great blog troubleshooting Lync Phone Edition:
http://blog.schertz.name/2012/03/troubleshooting-lync-phone-edition-issues/
Note: Microsoft is providing this information as a convenience to you. The sites are not controlled by Microsoft. Microsoft cannot make any representations regarding the quality, safety, or suitability of any software or information found there. Please
make sure that you completely understand the risk before retrieving any suggestions from the above link.
Kent Huang
TechNet Community Support -
Configure SPA2102 as SIP Phone to SIP Proxy Server
I have a SIP2102 which needs to be configured as SIP phone to a SIP Proxy Server. All calls will stay within the local network. I need to point SPA2102 to my SIP Proxy Server and assign an extension such that it is recognized by the SIP Proxy Server as part of its pool of valid extensions. The documentation is not clear when just trying to set these parameters.
if what you are trying to accomplish is simply register the SPA2102 with your Proxy server then the only thing you need to do is configure a USER ID (this is the extension number you want the SPA to have) and a Proxy (IP address of your SIP server) -- outbound proxy is only needed if your server requires the device to have this..
Both of these parameters can be found under Line 1 tab and these settings should come from your SIP server-- Internet port of the device must be connected to your network
SIP Port is another to consider in case your SIP server is using another port other than 6060..
| isolate! isolate! isolate! | -
Unitye Express doesn't not recognize SIP phone 3911
Hello community,
I have added sip phone 3911 successfully to UC520, however the unityexpress doesn't know these phones ( they do not appear under "Configure" ->"Phones" ). So the call from pstn get to AA and then dial sip exttion, it says that "invalid phone number". Would you please advise how to fix this issue ?
Thanks,
FDCHi Marcos,
We followed your instruction and the SIP phones 3911 have registered with CME. Calls between sip phones are ok. Just when I dial the AA number, then dial the extension of any sip phone, AA said that it is invalid number. If I dial the extension of analog phones ( we have four analog phones ), it works.
I logged into admin page of CUE, and go to "Configure" -> "Phones", I don't see any sip phone, so I guess this is the problem. Do you have that issue ?
When I show ephone regitstered, I also don't see sip phones, although they have already registered and call eaach other.
Best regards,
An -
A question about call manager traces for Sip phones.
So today I create a sip based ip communicator and pressed the new call button and heard a dial tone. I started typing my telephone number. Half way through, I heard another secondary dial tone (which indicates mis-configured route pattern somewhere) .
However, When I look at the call manager logs, I do not actually see the digits that I was typing. With SCCP, I can see the keypad button press messages in the traces, but here, I cannot see the pressed buttons in my CUCM traces. Can anyone help with telling me how I can see button presses going to call manager . All I can see are the logs below which came up as soon as I got the dial tone and the final sip invite messages. I see nothing in-between.
|SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.xx.4.xx on port 56714 index 31809 with 973 bytes:
[6387070,NET]
NOTIFY sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP 10.x.x.66:56714;branch=z9hG4bK00005b1e
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=00ffb00bc50a00340000499f-00006ab4
Call-ID: [email protected]
Date: Sat, 14 Feb 2015 14:17:40 GMT
CSeq: 19 NOTIFY
Event: dialog
Subscription-State: active
Max-Forwards: 70
Contact: <sip:[email protected]:56714;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 350
Content-Type: application/dialog-info+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8" ?>
<dialog-info xmlns:call="urn:x-cisco:parmams:xml:ns:dialog-info:dialog:callinfo-dialog" version="18" state="partial" entity="sip:[email protected]">
<dialog id="12" call-id="[email protected]" local-tag="00ffb00bc50a003300006390-00002d4f"><state>trying</state></dialog>
</dialog-info>
SIPStationD(12991) - processCommonDialogNotifyInd: Did 12 Sending Notified SIPOffHook to new CdfcHere is a more detailed explanation of how SIP calls notify cucm when they go off hook to make a call. The digit dialled here is 4080
+++++ Analysis of SIP Phone making a call +++++++++
The user picks up the phone and the IP Phone sends a NOTIFY to CUCM to indicate the start of a new dialog. This dialog begings by an offhook event
00869539.002 |14:58:13.837 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 976 bytes:
[46240,NET]
NOTIFY sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00002531
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=544e42f26d0b001e000056e7-0000311c
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:13 GMT
CSeq: 11 NOTIFY
Event: dialog
Subscription-State: active
Max-Forwards: 70
Contact: <sip:[email protected]:52910;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 350
Content-Type: application/dialog-info+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8" ?>
<dialog-info xmlns:call="urn:x-cisco:parmams:xml:ns:dialog-info:dialog:callinfo-dialog" version="10" state="partial" entity="sip:[email protected]">
<dialog id="6" call-id="[email protected]" local-tag="544e42f26d0b001d00007cc9-000044a3"><state>trying</state></dialog>
</dialog-info>
++++ CUCM SIP stack processes the new connection for the phone+++++++
00869540.001 |14:58:13.837 |AppInfo |//SIP/Stack/Info/0x0/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 (SIP_NETWORK_MSG), for event 1 (SIPSPI_EV_NEW_MESSAGE)
00869540.002 |14:58:13.837 |AppInfo |//SIP/Stack/Transport/0x0/sipTransportProcessNWNewConnMsg: context=(nil)
00869540.003 |14:58:13.837 |AppInfo |//SIP/Stack/Transport/0x0/sipConnectionManagerProcessNewConnMsg: gConnTab=0xe81c0d70, addr=10.50.16.1, port=52910, connid=2748, transport=TCP
++++ Next CUCM allocates a call id for this call +++++
00869546.002 |14:58:13.838 |AppInfo |LineControl(66) - Get call instance=1 for CI=24419584
+++Next CUCM sends a 200 OK to the NOTIFY request for the new dialog ++++
00869555.007 |14:58:13.839 |AppInfo |//SIP/Stack/Transport/0x0xe7df4d48/sipTransportPostSendMessage: Posting send for msg=0xefbe9910, addr=10.50.16.1, port=52910, connId=2748 for
00869555.008 |14:58:13.839 |AppInfo |//SIP/Stack/Info/0x0/act_dialog_pending_resp_event: Changing from State: SUBSCRIBE_STATE_DIALOG_PENDING to state SUBSCRIBE_STATE_ACTIVE
00869556.000 |14:58:13.839 |SdlSig |SIPSPISignal |wait |SIPTcp(1,100,71,1) |SIPHandler(1,100,79,1) |1,100,14,31314.75^10.50.16.1^SEP00909E9D106C |*TraceFlagOverrode
00869556.001 |14:58:13.839 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46241,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00002531
From: <sip:[email protected]>;tag=544e42f26d0b001e000056e7-0000311c
To: <sip:[email protected]>;tag=1822746380
Date: Mon, 16 Feb 2015 12:58:13 GMT
Call-ID: [email protected]
CSeq: 11 NOTIFY
Server: Cisco-CUCM10.5
Content-Length: 0
++++ The IP Phone sends its connection ID to CUCM, its ip address and its port number+++++++++
00869541.001 |14:58:13.838 |AppInfo |SIPStationInit: connID=2748, SEP00909E9D106C, 10.50.16.1:52910, Routed signal by connection index to (1,100,73,66)
++++ Next CUCM informs us that the NOTIFY message is for an offhook event ++++++
00869542.003 |14:58:13.838 |AppInfo |SIPStationD(66) - processCommonDialogNotifyInd: Notified Dialogs - Did 6 State trying
00869542.004 |14:58:13.838 |AppInfo |SIPStationD(66) - processCommonDialogNotifyInd: Did 6 Sending Notified SIPOffHook to new Cdfc
00869542.010 |14:58:13.838 |AppInfo |SIPStationD(66) - processSIPOffHook Primary Call Not-Found
00869543.000 |14:58:13.838 |SdlSig |SIPOffHookInd
+++ The next thing is the USER dials a digit on the phone ++++++
This is where it gets a little complicated. So lets examine this. The first digit that is dialled generates an INVITE to CUCM like this:
In this example the user dialled "4" first so we see an "INVITE sip:4@host-IP"
00869559.002 |14:58:14.064 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 1445 bytes:
[46242,NET]
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000015ec
From: "Emre ESEN" <sip:[email protected]>;tag=544e42f26d0b001d00007cc9-000044a3
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
Max-Forwards: 70
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 101 INVITE
User-Agent: Cisco-SIPIPCommunicator/9.1.1
Contact: <sip:[email protected]:52910;transport=tcp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Emre ESEN" <sip:[email protected]>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.1.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 373
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 21020 0 IN IP4 10.50.16.1
s=SIP Call
t=0 0
m=audio 20250 RTP/AVP 0 8 18 9 116 124 101
c=IN IP4 10.50.16.1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:124 ISAC/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
+++++ NEXT CUCM sends a trying for the INVITE it received +++++++++++
00869562.001 |14:58:14.065 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46243,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000015ec
From: "Emre ESEN" <sip:[email protected]>;tag=544e42f26d0b001d00007cc9-000044a3
To: <sip:[email protected];user=phone>
Date: Mon, 16 Feb 2015 12:58:14 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: presence
Content-Length: 0
++++NOW CUCM evaluates the DTMF supported by the phone to determine how to inform the phones to send the remaining dtmf digits++++
From the INVITE cucm concludes that KPML and rtp-nte is supported
00869566.009 |14:58:14.066 |AppInfo |setEndpointsDtmfCaps: KPML Supported.
00869566.010 |14:58:14.066 |AppInfo |setEndpointsDtmfCaps: Detected inband DTMF support
Next CUCM generates kpml event pkg which is going to be used to receive the remaining digits from the phone
00869590.001 |14:58:14.067 |AppInfo |SIPEventPkg::SIPEventPkg 0xe4a1d1e0 scbId[16725], event name[kpml; [email protected]; from-tag=544e42f26d0b001d00007cc9-000044a3], id[]
+++ Next CUCM sends a SUBSCRIBE to the IP phone for kpml event +++++
00869594.001 |14:58:14.068 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46244,NET]
SUBSCRIBE sip:[email protected]:52910 SIP/2.0
Via: SIP/2.0/TCP 10.28.132.111:5060;branch=z9hG4bKce719b37856
From: <sip:[email protected]>;tag=480227084
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 SUBSCRIBE
Date: Mon, 16 Feb 2015 12:58:14 GMT
User-Agent: Cisco-CUCM10.5
Event: kpml; [email protected]; from-tag=544e42f26d0b001d00007cc9-000044a3
Expires: 7200
Contact: <sip:[email protected]:5060;transport=tcp>
Accept: application/kpml-response+xml
Max-Forwards: 70
Content-Type: application/kpml-request+xml
Content-Length: 424
<?xml version="1.0" encoding="UTF-8" ?>
<kpml-request xmlns="urn:ietf:params:xml:ns:kpml-request" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="urn:ietf:params:xml:ns:kpml-request kpml-request.xsd" version="1.0">
<pattern criticaldigittimer="1000" extradigittimer="500" interdigittimer="15000" persist="persist">
<regex tag="Backspace OK">[x#*+]|bs</regex>
</pattern>
</kpml-request>
+++ Next we get a 200 OK to the SUBSCRIBE from the ip phone ++++
00869595.002 |14:58:14.118 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 459 bytes:
[46245,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.28.132.111:5060;branch=z9hG4bKce719b37856
From: <sip:[email protected]>;tag=480227084
To: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 101 SUBSCRIBE
Server: Cisco-SIPIPCommunicator/9.1.1
Contact: <sip:[email protected]:52910;transport=TCP>
Expires: 7200
Content-Length: 0
+++ NEXT the IP phones sends the remaining digit dialled on the phone to CUCM +++
00869603.002 |14:58:14.183 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 573 bytes:
[46247,NET]
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000045c8
To: <sip:[email protected]>;tag=480227084
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 1000 NOTIFY
Event: kpml
Subscription-State: active; expires=7200
Max-Forwards: 70
Contact: <sip:[email protected]:52910;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 0
00869608.001 |14:58:14.183 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46248,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000045c8
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
To: <sip:[email protected]>;tag=480227084
Date: Mon, 16 Feb 2015 12:58:14 GMT
Call-ID: [email protected]
CSeq: 1000 NOTIFY
Server: Cisco-CUCM10.5
Content-Length: 0
+++Next the IP phone sends the next digit. Here its important to note that the NOTIFY doesnt contain the next digit,
the NOTIFY is still the same as the first digit but the next digit is carried in the xml document attached to the NOTIFY.
At this point I will insert a paragraph from the RFC 4730 for SIP KPML
+++++++++++++
The event package uses SUBSCRIBE
messages and allows for XML documents that define and describe filter
specifications for capturing key presses (DTMF Tones) entered at a
presentation-free User Interface SIP User Agent (UA). The event
package uses NOTIFY messages and allows for XML documents to report
the captured key presses (DTMF tones), consistent with the filter
specifications, to an Application Server +++++++++++++++++++++++++++
00869609.002 |14:58:14.209 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 877 bytes:
[46249,NET]
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00003c9d
To: <sip:[email protected]>;tag=480227084
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 1001 NOTIFY
Event: kpml
Subscription-State: active; expires=7200
Max-Forwards: 70
Contact: <sip:[email protected]:52910;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 209
Content-Type: application/kpml-response+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8"?>
<kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false" forced_flush="false" digits="0" tag="Backspace OK"/>
00869622.001 |14:58:14.210 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46250,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00003c9d
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
To: <sip:[email protected]>;tag=480227084
Date: Mon, 16 Feb 2015 12:58:14 GMT
Call-ID: [email protected]
CSeq: 1001 NOTIFY
Server: Cisco-CUCM10.5
Content-Length: 0
+++ Again we get the next digit ++++
00869624.002 |14:58:14.262 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 877 bytes:
[46251,NET]
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK0000310f
To: <sip:[email protected]>;tag=480227084
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 1002 NOTIFY
Event: kpml
Subscription-State: active; expires=7200
Max-Forwards: 70
Contact: <sip:[email protected]:52910;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 209
Content-Type: application/kpml-response+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8"?>
<kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false" forced_flush="false" digits="8" tag="Backspace OK"/>
00869637.001 |14:58:14.263 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46252,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK0000310f
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
To: <sip:[email protected]>;tag=480227084
Date: Mon, 16 Feb 2015 12:58:14 GMT
Call-ID: [email protected]
CSeq: 1002 NOTIFY
Server: Cisco-CUCM10.5
Content-Length: 0
+++ Finally we get the last digit ++++
00869638.002 |14:58:14.390 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 877 bytes:
[46253,NET]
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00006c1c
To: <sip:[email protected]>;tag=480227084
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 1003 NOTIFY
Event: kpml
Subscription-State: active; expires=7200
Max-Forwards: 70
Contact: <sip:[email protected]:52910;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 209
Content-Type: application/kpml-response+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8"?>
<kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false" forced_flush="false" digits="0" tag="Backspace OK"/>
Once digit collection is completed CUCM proceeds to finalise its digit analysis process.
Note that digit analysis is carried out for each digit that is recieved. I have only included the final DA here
00869648.003 |14:58:14.391 |AppInfo |Digit Analysis: star_DaReq: Matching SIP URL, Numeric User, user=4080
00869648.004 |14:58:14.391 |AppInfo |Digit Analysis: getDaRes data: daRes.ssType=[0] Intercept DAMR.sstype=[0], TPcount=[0], DAMR.NotifyCount=[0], DaRes.NotifyCount=[0]
00869648.005 |14:58:14.391 |AppInfo |Digit Analysis: getDaRes - Remote Destination [4080] isURI[0]
00869648.012 |14:58:14.391 |AppInfo |Digit analysis: match(pi="2", fqcn="9106", cn="9106",plv="5", pss="", TodFilteredPss="", dd="4080",dac="0")
00869648.013 |14:58:14.391 |AppInfo |Digit analysis: analysis results
00869648.014 |14:58:14.391 |AppInfo ||PretransformCallingPartyNumber=9106
|CallingPartyNumber=9106
|DialingPartition=
|DialingPattern=4XXX
|FullyQualifiedCalledPartyNumber=4080
|DialingPatternRegularExpression=(4[0-9][0-9][0-9])
|DialingWhere=
+++++Once this is done CUCM then proceeds to send the call out to to the intended destination as configured in the RL ++++
00869701.001 |14:58:14.435 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.250.0.13 on port 5060 index 2754
[46256,NET]
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.28.132.111:5060;branch=z9hG4bKce931ee3d74
From: "Emre ESEN" <sip:[email protected]>;tag=16726~813ee89e-33db-4d58-9f6a-61542cc840ee-24419585
To: <sip:[email protected]>
Date: Mon, 16 Feb 2015 12:58:14 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
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