Hi ,
We have upgarded the the firmware to the 3905.9-2-2ES2 , but show voice register pool phone-load still shows the old firmware, but the phoen itself is showing the new upgraded version on the dsiplay ...any advice is highly appricated,
ADM-CME9#show voice register pool phone-load
Pool Device Name     Current-Version             Previous-Version
==== =============== =========================== ===========================
1    SEP7081053DE72F Cisco/SPA502G-7.4.8a
3    SEP34BDC8C6C412 Cisco-CP3905/9.2.1
4    SEP34BDC8C64561 Cisco-CP3905/9.2.1
5    SEP54781AE1F531 Cisco-CP3905/9.2.1
6    SEP54781AE171D2 Cisco-CP3905/9.2.1
10   SEP54781AE1F544 Cisco-CP3905/9.2.1
15   SEP1CE6C77323CD Cisco-CP3905/9.2.1
16   SEP58971E282A23 Cisco-CP3905/9.2.1
17   SEP58971E2822A8 Cisco-CP3905/9.2.1
19   SEP1CE6C77321F3 Cisco-CP3905/9.2.1
30   SEP54781AE171E2 Cisco-CP3905/9.2.1
31   SEP54781AE16FD4 Cisco-CP3905/9.2.1
32   SEP54781AE16F2F Cisco-CP3905/9.2.1
33   SEP54781A1C77FD Cisco-CP3905/9.2.1
34   SEP54781A1C77DC Cisco-CP3905/9.2.1
35   SEP54781AE17527 Cisco-CP3905/9.2.1
36   SEP54781AE17766 Cisco-CP3905/9.2.1
37   SEP54781AE1731A Cisco-CP3905/9.2.1
38   SEP54781AE08B8D Cisco-CP3905/9.2.1
39   SEP54781AE123B1 Cisco-CP3905/9.2.1

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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.
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dial-peer voice 1003 pots
trunkgroup 1
corlist outgoing CITIES
description CALLING CITIES
destination-pattern 90[1-9]......
forward-digits 8
voice service voip
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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:
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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,

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[edit]
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[/edit]
Attachments:
get-http-service.png ‏22 KB

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FAC LPCOR IS NOT WORKING IN CME SIP PHONES

Dear Team,
We are planning to put FAC on SIP (7821) phones for calling long distance.
CME version is 10.0
Please find the config
gw-accounting aaa
aaa new-model
aaa authentication login h323 local
aaa authorization exec h323 local
aaa authorization network h323 local
aaa session-id common
voice register global
mode cme
source-address 10.X.X.X port 5060
timeouts interdigit 5
max-dn 500
max-pool 475
load 7821 sip78xx.10-2-1-12
authenticate register
authenticate realm router.local
timezone 35
tftp-path flash:
create profile sync 0003478159444525
voice lpcor enable
voice lpcor custom
group 1 national-FAC
voice lpcor policy national-FAC
service fac
accept national-FAC fac
application
package auth
param max-retries 0
param passwd-prompt flash0:/enter_pin.au
param abort-digit *
param user-prompt flash0:/enter_account.au
param passwd 12345
param term-digit #
param max-digits 32
username 1111 password 7 1446435A5D
voice register pool 2
lpcor type local
lpcor incoming national-FAC
lpcor outgoing national-FAC
busy-trigger-per-button 1
id mac F09E.636F.0F4B
type 7821
number 1 dn 2
template 1
cor incoming NATIONALACCESS 1 3002
dtmf-relay rtp-nte sip-notify
username user2 password cisco
codec g711ulaw
voice register dn 2
number 3002
call-forward b2bua noan 5000 timeout 20
allow watch
name 3002
label 3002
dial-peer voice 117 voip
destination-pattern 905[0256].......$
lpcor outgoing national-FAC
session protocol sipv2
session target ipv4:10.X.X.X
incoming called-number .
codec g711ulaw
no vad
label 3004
CME#dir flash:
Directory of flash0:/
1 -rw- 96910452 Oct 21 2014 03:35:18 +04:00 c3900e-universalk9-mz.SPA.153-3.M3.bin
2 -rw- 3064 Oct 21 2014 03:49:06 +04:00 cpconfig-39xx.cfg
3 drw- 0 Nov 12 2014 14:09:54 +04:00 its
14 drw- 0 Nov 12 2014 15:16:46 +04:00 GUI
32 drw- 0 Oct 21 2014 03:49:32 +04:00 ccpexp
273 -rw- 2464 Oct 21 2014 03:51:18 +04:00 home.shtml
274 -rw- 116608 Nov 12 2014 15:10:32 +04:00 CME-GUI.rar
275 -rw- 3090800 Nov 12 2014 16:35:04 +04:00 kern78xx.10-2-1-12.sbn
276 -rw- 36307280 Nov 12 2014 16:36:00 +04:00 rootfs78xx.10-2-1-12.sbn
277 -rw- 364072 Nov 12 2014 16:36:10 +04:00 sboot78xx.10-2-1-12.sbn
278 -rw- 1228 Nov 12 2014 16:36:32 +04:00 sip78xx.10-2-1-12.loads
279 -rw- 496521 Nov 12 2014 17:01:40 +04:00 music-on-hold.au
280 -rw- 23660 Nov 19 2014 14:48:40 +04:00 enter_account.au
281 -rw- 24164 Nov 19 2014 14:48:48 +04:00 enter_pin.au
261324800 bytes total (120594432 bytes free)
But when ever iam trying mobile calls call is disconnecting. please find the debugs
Nov 19 20:04:34.661: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_peer:
peer tag 40001, direction 0
Nov 19 20:04:34.661: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
lpcor International-FAC
Nov 19 20:04:34.661: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
lpcor International-FAC index 1
Nov 19 20:04:34.661: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_peer:
Return Lpcor Index 1 for Peer Tag 40001
DAMAC_AKOYA-CME#
Nov 19 20:04:39.225: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
lpcor International-FAC
Nov 19 20:04:39.225: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
lpcor International-FAC index 1
Nov 19 20:04:39.225: //-1/xxxxxxxxxxxx/LPCOR/lpcor_index_is_valid:
lpcor index 1 is valid
Nov 19 20:04:39.225: //-1/xxxxxxxxxxxx/LPCOR/lpcor_policy_validate_internal:
Source LPCOR Index=1, Target LPCOR Policy=International-FAC
Nov 19 20:04:39.225: -Traceback= 19E34D6z 581A4DCz 581AC7Fz 55B3588z 55B3063z 55AEA34z 55B6A53z 55B7255z 55B649Bz 55F5AF5z 5583EEFz 55894B6z 5589734z 55F5AF5z 558A44Az 55CF04Ez
DAMAC_AKOYA-CME#
Nov 19 20:04:39.225: //-1/xxxxxxxxxxxx/LPCOR/lpcor_policy_validate_internal:
Validate Pass; lpcor (source[1] target[1])
Nov 19 20:04:39.225: //-1/xxxxxxxxxxxx/LPCOR/lpcor_policy_validate_get_service:
FAC is enabled; lpcor (source[1] target[1])
Please advise. If iam missing any steps in this.
Thanks,
LAJAN JALEEL

Dear Amit,
The configuration is working fine. We find out what is the exact issue.
*****Issue******
Debug voice application auth was giving an out put that
user prompt url is not found , Even though the exact directory was there under flash0:
CME#dir flash:
Directory of flash0:/
1 -rw- 96910452 Oct 21 2014 03:35:18 +04:00 c3900e-universalk9-mz.SPA.153-3.M3.bin
2 -rw- 3064 Oct 21 2014 03:49:06 +04:00 cpconfig-39xx.cfg
3 drw- 0 Nov 12 2014 14:09:54 +04:00 its
14 drw- 0 Nov 12 2014 15:16:46 +04:00 GUI
32 drw- 0 Oct 21 2014 03:49:32 +04:00 ccpexp
273 -rw- 2464 Oct 21 2014 03:51:18 +04:00 home.shtml
274 -rw- 116608 Nov 12 2014 15:10:32 +04:00 CME-GUI.rar
275 -rw- 3090800 Nov 12 2014 16:35:04 +04:00 kern78xx.10-2-1-12.sbn
276 -rw- 36307280 Nov 12 2014 16:36:00 +04:00 rootfs78xx.10-2-1-12.sbn
277 -rw- 364072 Nov 12 2014 16:36:10 +04:00 sboot78xx.10-2-1-12.sbn
278 -rw- 1228 Nov 12 2014 16:36:32 +04:00 sip78xx.10-2-1-12.loads
279 -rw- 496521 Nov 12 2014 17:01:40 +04:00 music-on-hold.au
280 -rw- 23660 Nov 19 2014 14:48:40 +04:00 enter_account.au
281 -rw- 24164 Nov 19 2014 14:48:48 +04:00 enter_pin.au
261324800 bytes total (120594432 bytes free)
my application configuration was like this:
application
package auth
param passwd-prompt flash0:/enter_pin.au
param max-retries 0
param term-digit #
param user-prompt flash0:/enter_account.au
param abort-digit *
param passwd 12345
param max-digits 32
*****Resolution*****
changed user prompt and password prompt from flash0:/ to flash:/
application
package auth
param passwd-prompt flash:/enter_pin.au
param max-retries 0
param term-digit #
param user-prompt flash:/enter_account.au
param abort-digit *
param passwd 12345
param max-digits 32
Right now lpcor is working fine for Long Distance calls with the same config.
Thanks All

IP phones only working one way

Sorry, I'm a complete VoIP novice...
We have a bit of a weird problem in that when you dial an extension which is our other office (connected by 2 x 2mpbs MPLSs) the caller can hear the recieving person's voice but the the recieving can't hear the caller. This happens both ways, i.e. when we call them and when they call us. We do have an access-list in place and I have opened up the ports that were logged as being denied when the calls were placed (namely 51000 range) and now nothing is being logged as denied.
Apparently this has worked when initially tested, so I'm not sure what could have changed.
Any ideas? Has anyone come across this before? We are using Avaya phones.
Many thanks,
J2026

Thanks for your response Martin, much appreciated.
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2) Definitely connectivity is OK, OSPF routes in place, lots of traffic passes between the 2 regularly. Verifyied by ping as suggested.
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Output queue: 0/1000/64/1067272 (size/max total/threshold/drops)
However, I have cleared the counters and tried again, with same results, yet no more packets had been dropped.
4) We currently only have 4 IP phones (1 at this site, 3 at other) as we are still setting it up. The problem is with all 4 phones, when going between offices. They work fine within the same office (to IP and standard analogue phones). Also, I can use the IP phone here to dial out to my mobile with no problems (this does not traverse the 2mpbs MPLS).
We will be implementing some QoS queueing tools but I was under the impression that we should be able to get at least some, albeit choppy, audio working. Do you think it's likely/reasonable that this problem won't go away until proper queuing is implemented?
Many thanks for your help.
Regards, J

My phone only syncs one way

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Message was edited by: Wolf68k

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Hello,
I am wondering if anyone has seen this before. I have the latest version of Anyconnect for the IPhone and Jabber app, running on the newest version of CUCM 8.6 and Jabber 8.6.
The Jabber phone registers fine and I can make internal calls without an issue. I also can make external calls to H323 and HGCP PRI gateways without an issue. However, when I make a call from the Jabber client that goes out a SIP gateway and SIP trunk I get one way audio. I do not see any packets even trying to leave inside interface of the ASA headed toward the SIP trunk IP.
I've also verified that useing the same VPN profile I can make 2 way audio calls out the SIP trunk from a softphone running on windows.
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Thanks.

Hi,
try to disable early media on 180 under sip profile settings for jabber client.
HTH
Anas
please rate if itnis helpful
Sent from Cisco Technical Support Android App

CUCM 8.6 Dropped call transfers involving SIP phones

Hi All,
I am a developer who has been tasked with figuring out why call transfers are being dropped by Cisco CUCM when the original call comes from a SIP phone. This scenario works:
Cisco phone calls another Cisco phone, which transfers the original call to a SIP phone
These scenarios do not work:
SIP phone calls Cisco phone, which transfers the original call to another Cisco phone
SIP phone calls Cisco phone, which transfers the original call to another SIP phone
I have researched the Call Manager traces to the best of my ability, and I see some info in there that could potentially point to the source of the problem. I am just unable to understand what the trace means:
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/active_CcDisconnReq: ccDisconnReq.onBehalfOf=Media : ccDisconnReq.s.sv=2 : ccDisconnReq.c.cv=47 |1,100,63,1.93259^10.10.10.85^*
10:23:08.672 |//SIP/Stack/Info/0x0/sipConstructContainerContext #### Created container=0xb0b42f58|1,100,71,1.1^*^*
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendReasonHdr: appendReasonHdr - Invalid Disconnect Cause(cause=47), No Reason Header Appended|1,100,63,1.93259^10.10.10.85^*
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendRPIDHdrForOriginalCalledParty: SIP device does not Support Orig Dialled Phone nego: 0|1,100,63,1.93259^10.10.10.85^*
I have been wondering whether this could be a codec issue, however the SIP phones we are using are configured with the following codecs:
G711U
G711A
G722
ILBC
GSM
and our SIP software is also set to accept the first codec offered by the remote side. It seems from the SIP client logs that G722 is being used as the codec to communicate with the Cisco phones, but perhaps I'm misinterpreting.
I have attached a CUCM trace of a call from a SIP phone (ext. 491) to a Cisco handset (ext. 170) where the Cisco handset attempts to transfer the call to another SIP phone (ext. 492). The trace snippet shown above is from this log.
I would really appreciate it if someone more experienced with VoIP/SIP/CUCM could take a look and offer any ideas on what the issue might be, and also how we might be able to address it. I can try to provide more info about our CUCM configuration if needed.
Thanks in advance!

Leslie, so here is what I found from the traces....
To understand the difference we need to understand how cucm performs call transfers from a sccp signalling point and a sip signalling point
SCCP
When the transfer key is pressed
1. CUCM sends a CloseReceiveChannel and StopMediaTransmission to the IP phone involved in active media (referenced by the callids)
NB, here CUCM updates the call state on the phone to a call state of 8 which is "Hold"
2.CUCM tells the held party to listen MOH from MOH server
3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
5..CUCM sends a CloseReceiveChannel between the held phone and MOH server (to tear down the media)
6. Next CUCM sends a CloseReceiveChannel and StopMediaTransmission to the transfering party & transfered party to remove Xferring party from call
7. finally CUCM sends OpenReceiveChannel between the original called party and the transfered party..and call is done
For SIP signalling. when the first transfer key is pressed
1. CUCM sends invite (re-invite) with an inactive SDP (a=inactive) to indicate a break in media path
2. CUCM sends a Delayed offer to insert MOH or to resume Media stream
NB: CUCM expects a sendrecv offer with SDP to the DO. (NB:if cucm gets an inactive offer SDP in the 200 OK instead of providing a send-recv offer SDP, the media path remains in an inactive state and causes calls to dropcall will drop),CUCM sends an ACK with sendonly to the 200 OK
3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
5. Next CUCM sends a re-invite with an inactive SDP to indicate a break in media path to MOH (in attempt to complete transfer)
6.Next CUCM sends an inactive SDP to indicate a break in media path between transfering party & transfered party to remove Xferring party from call
7. Next CUCM sends a DO re-invite to connect the transfered party. The far end then sends 200 OK with the required SDP to connect the call
Now having explained all of these, we need to look at where the call is failing for SIP-----SCCP----SIP calls without MTP
lets look at succesful SCCP-----SCCP-----SIP without MTP
Point 4 above
++++++++Extension 170 presses the transfer button to connect the two calls (Callid=24378483)+++++++++++++
(0003395) SoftKeyEvent softKeyEvent=4(Trnsfer) lineInstance=1 callReference=24378483
Point 5 above
++++Next CUCM closed the media between extension 160 and MOH server callid=24378480(this is the only active call on this callid)+++
(0003396) CloseReceiveChannel conferenceID=24378480 passThruPartyID=16777845. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
Point 6 Above
+++++Next cucm closes the call between extension 170 and 490 callid=(24378483)++++++++
(0003395) CloseReceiveChannel conferenceID=24378483 passThruPartyID=16777847. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
(0003395) StopMediaTransmission conferenceID=24378483 passThruPartyID=16777847. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
Point 6 above for the sip side (since the destination is SIP, to tear down media to SCCP phone, so as to connect the caller to the xfered party)
+++++++Next CUCM sends a re-invite with a=inactive SDP to the sip phone ++++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885626,NET]
INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23332dbee978
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
o=CiscoSystemsCCM-SIP 192115 2 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 0.0.0.0
m=audio 24560 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=ptime:20
a=inactive-----------------------------------------------------Inactive
Still part of Point 6 for SIP signalling
++++++++++++Next sip phone responds with a 200 OK recevonly SDP +++++++++++++++++++
//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
[885628,NET]
SIP/2.0 200 OK
v=0
o=- 18077 11099 IN IP4 10.10.10.104
s=yasdjip
c=IN IP4 10.10.10.104
t=0 0
a=ptime:20
a=recvonly-------------------------------------a=recvonly
Finally Point 7 above..
+++++++++++++=Next cucm sends a DO re-invite to extension 492-sip phone++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885630,NET]
INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
+++++++Next we get a 200 OK from sip phone with sdp=sendrecv+++++++++=
/SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
[885634,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
Contact:
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
Call-ID: [email protected]
v=0
o=- 18077 11099 IN IP4 10.10.10.104
s=yasdjip
c=IN IP4 10.10.10.104
t=0 0
m=audio 16574 RTP/AVP 9 101
a=rtpmap:101 TELEPHONE-EVENT/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
+Now CUCM sends an OpenReceiveChannel and start media xmission to sccp phone (callid=24378480) with media parameters of sip phone++++++
(0003396) OpenReceiveChannel conferenceID=24378480 passThruPartyID=16777848 millisecondPacketSize=20 compressionType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierIn=? sourceIpAddr=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104). myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
(0003396) startMediaTransmission conferenceID=24378480 passThruPartyID=16777848 remoteIpAddress=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104)
remotePortNumber=16574 milliSecondPacketSize=20 compressType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierOut=?. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
+++++++++++=Next Phone sends its ACK+++++++++++++++
(0003396) OpenReceiveChannelAck Status=0, IpAddr=IpAddr.type:0 ipAddr:0x0a0a0a89000000000000000000000000(10.10.10.137), Port=20352, PartyID=16777848
+++++++++++=Next CUCM sends ACK to 200 OK from SIP Phone+++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885635,NET]
ACK sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23366067b8c0
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
Date: Tue, 19 Feb 2013 21:44:45 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 237
v=0
o=CiscoSystemsCCM-SIP 192115 3 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.137
b=TIAS:64000
b=AS:64
t=0 0
m=audio 20352 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Now at this point all is well...and the call is connected....
Now here is where the call is failing on the SIP-SCCP-SIP call without MTP
From Point 2 above, CUCM sends a DO to insert MOH, and then gets response, then sends an ACK to 200 Ok back to SIP Phone..
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
[881160,NET]
ACK sip:[email protected]:53361;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22035ecc1fcb
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Date: Tue, 19 Feb 2013 17:38:50 GMT
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
Max-Forwards: 70
CSeq: 102 ACK
o=CiscoSystemsCCM-SIP 190666 3 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195---------------------------------------IP address of MOH server
t=0 0
m=audio 4000 RTP/AVP 0--------------------------------MOH port 4000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendonly---------------------------------------------------------sendonly
+++++NOW Point 6 above (SIP) CUCM sends a=inactive to break media path to MOH server to connect caller and xfered party++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
[881161,NET]
INVITE sip:[email protected]:53361;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Date: Tue, 19 Feb 2013 17:39:04 GMT
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
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: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 164
v=0
o=CiscoSystemsCCM-SIP 190666 4 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 0.0.0.0--------------------------------------------------------------------Media IP is 0.0.0.0
t=0 0
m=audio 4000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=inactive---------------------------------------------------------------------media inactive
At this point, we should get a response back from the sip phone...
and here is what we got..
++Trying which is expected++++
//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 331 bytes:
[881162,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
CSeq: 103 INVITE
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Content-Length: 0
++++++++Then we get a BYE+++++++++++++++
/SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 576 bytes:
[881163,NET]
BYE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.104:53361;branch=z9hG4bKa2vdQvR7J9OiMyjU;rport
Contact:
Max-Forwards: 70
From: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
User-Agent: Acrobits Softphone Business/2.4.8
To: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
CSeq: 3 BYE
Content-Length: 0
So this is the root cause of the problem. Your SIP phone does not know how to respond to multiple media break between it and the MOH server.
The difference between this and the succesful SCCP-SIP--SCCP, is that the held party was a sccp phone, hence the sip phone only has to process one a=inactive SDP message, where as in the SIP-SCCP-SIP, the help party was sip, so the sip phone has to process two a=inactive SDP messages
Now what is the difference when MTP is involved! A Big difference. MTP stays in the media path. So there is never a break in media and no inactive SDP attribute is sent. The flow looks like below:
for the initial call...The SIP phone sends its media to MTP and likewise the SCCP phone
SIP------Media------MTP------------Media-------SCCP Phone
When the new destination is dialled and transfer is commited,
SIP-------------media----MTP--------media---------MTP
The final invoite sent to connect 492 and 491 has MTP as the IP address to connect Media to.
++++++++Ivite to 492 ++++++++++++++
INVITE sip:[email protected]:61303;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231a3b24b862
From: ;tag=192048~d8e94532-127d-4dca-bba0-64b1675da032-24378472
To: ;tag=78FF5BF6C019A55EA020B69BB6A767E2
Date: Tue, 19 Feb 2013 21:24:59 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: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 214
v=0
o=CiscoSystemsCCM-SIP 192048 1 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195---------------------------------------------------------------the MTP ip address
t=0 0
m=audio 25038 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++Invite to 491 +++++++++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.94 on port 50376 index 1887
[885429,NET]
INVITE sip:[email protected]:50376;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231b78d1b56
From: ;tag=192046~d8e94532-127d-4dca-bba0-64b1675da032-24378467
To: "491" ;tag=F13CE94DE942C47680356A647DC7F916
Date: Tue, 19 Feb 2013 21:24:59 GMT
Call-ID: AE7045FFB2D8D9C28D54651473A14A5D41B5B93C
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
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: "Leslie2" ;party=calling;screen=no;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 237
v=0
o=CiscoSystemsCCM-SIP 192046 1 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195----------------------------------------MTP
b=TIAS:64000
b=AS:64
t=0 0
m=audio 25030 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Wao! That was a long one isnt it...It was fun too.
So now you can look at your sip phones and see if they can accept two inactive sdp messages within the same call. That way you can remove MTP. otherwise you will have MTP involved in every single call involving a sip phone, even if they do not involve transfers
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