PSTN call transferring between sites
We are planning on having 2 sites
Site A: Main Lync server with mediation/frontend roles. Edge server. SIP trunk
Site B: SBA with PRI
Sites are to be connected with a VPN
We will have users at Site A that will take a call on the PSTN and need to be able to transfer that call to users at Site B. Site B has no receptionist, their receptionist is at Site A.
My initial plan was to have every users line uri set up with the phone number and extension, no DIDs. Site A could be tel:+19997654321;ext=1000 and Site B could be tel:+19991234567;ext=2000.
My question is what is the call path for transferring when a call comes in on Site A PSTN -> Site A User -> Site B User?
Will this call flow through the VPN to Site B from Site A and be using up one of the SIP trunks for the duration?
What happens if I make users at Site B have DIDs instead of extensions for their line uri? Could the call be transferred to the PRI and not have to flow through the VPN?
Hi kuptzlocher,
Have a look at the following picture.
Based on my understanding, if the number was already in a globally unique format (E.164), then Reverse number lookup will query it :
Is this number assigned to anything in Lync? If it is, then Lync will be able to translate the number to a SIP URI and route the call directly to inbound routing based on said URI where the process will then end.(In other words, it
will through the VPN)
Best regards,
Eric
Similar Messages
-
Direct Transfere between 2 Calls each on a different DN
Dears,
I have this client that used to have a PBX and now switched to IP telephony with CUCM 9.x
The operator used to have a feature that allowed him to do a direct transfere between 2 external calls while he himself steps out of the call (No Conferance).
I Need something similar to this other than starting a call with a Meet-Me number and transfering both call to that number. ( The Operators a Dum and found this proceedure very complicated and hard).
I Need a 1 button step solution like a Direct Transfere.
Thanks for your HelpReply From Cisco TAC:
You need to enable this option “Join Across Lines” under phone configuration page >> change its value to ON >> save >> Reset the phone.
Once you enable that feature, and you have 2 active calls on the same phone, press on the JOIN softkey for the first call, then press on the second line, this will make a conference call.
Please make sure this parameter “Drop Ad Hoc Conference” is set to “never”.
Form CUCM webpage >> system >> Service Parameter >> select any server >> select “Call Manager” service >> Change this parameter Drop Ad Hoc Conference to “never” >> save.
When the conference start, the user can leave without any problem. (I have tested that in the lab and it is working fine). -
Branch office PSTN call routing
Customer had HQ and Branch office with 512 MPLS line, in HQ alone they had E1 trunk, in branch office for calling PSTN the call should travel via MPLS to HQ, and the branch PSTN call terminated in HQ only, is it a good design or better we need to add ISDN line in branch level to terminate the PSTN call.
Balamurugan:
1) I think it is possible. My R&S skills are oxidized but you can have MPLS VPN configured and use frame-relay as a serial encapsulation where you will be able to configure RTP header compression.
2) RSVP over MPLS: yes. But as Marwashawi said, I prefer to work with the LLQ QOS for voice traffic. For that you will need to agree that with your service provider.
Rad Baver
BALAMURUGAN SINGARAM escribió:Thanks for the reply, branch to branch they are using 512 KBPS with VPN tunnel, g729 codec is used in between HQ and branch, the issuse is voice quality between the branche and HQ, the ping round trip delay is more than 250 MS [ no packet drops ], QOS is applied in switch and CUCM level, could you please light me do we can fine tune more on WAN interface level.1) do it possible to use header compression on MPLS cloud2) do it possible to enable "RSVP" in MPLS link.Could you please light me on how to enable RSVP & Header compression on MPLS link. -
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
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared" -
CUIC report for the Call transferred.
Hello,
I am looking for CUIC report for the number of calls transfered by an agent. Below is the query that works fine in SQL, but i am unable to generate a CUIC report on this.
select AgentPeripheralNumber,count(*) from t_Termination_Call_Detail where
where DateTime between '02 Aug 2013 00:00' and '18 Sep 2013 23:59'
and
ICRCallKey in (select ICRCallKeyParent from t_Termination_Call_Detail
where DateTime between '02 Aug 2013 00:00' and '18 Sep 2013 23:59'
and CED='2345')
group by AgentPeripheralNumber
Regards,
SandeepTry querying "Termination_Call_Detail" instead of "t_Termination_Call_Detail".
Check out the CallDetailReport.xml report available here:
http://developer.cisco.com/web/ccr/documentation
That should be a good starting point for you.
-Jameson -
Branch phone no audio when making outbound PSTN call - inbound and On-Net calls are good
I am trying a new setup. There are no other phones on branch A configured with CUCM so there is no comparison. I setup another phone in Branch B for comparison.
Scenario:
CUCM (ver9) in Main site
Main Site and Branch A site has VPN for WAN connectivity.
Main Site and Branch B site has VPN for WAN connectivity
Branch A phone is registered to Main site CUCM and configured to use same settings as Branch B phone. Branch B phone has no issues at all
Branch A no audio when making PSTN outbound call
Branch A audio is good when receiving INbound PSTN call and inbound/outbound On-Net call
I am not sure if it is a routing issue or something else. Below is an excerpt from the RTM logs which is attached. I believe issue is on this part after call connects where it shows media error but not sure what to do to correct it. Assuming Firewall is not an issue. If we cant find out what is wrong, i will start looking at the firewall.
called number : 0016326635801
calling number: 4168
Phone IP : 10.100.3.29
Main Site Gateway IP : 10.130.3.9 / H323
CUCM : 10.130.3.115/116
=========================================================================================================
57315658.000 |22:32:55.911 |SdlSig |CcAlertReq |outgoing_call_proceeding3 |StationCdpc(3,100,59,74928) |StationD(3,100,58,2238) |3,100,13,119955.3^10.130.3.9^* |[R:N-H:0,N:1,L:0,V:0,Z:0,D:0] CI=60724230 CI.branch=0 FDataType=0opId=0ssType=0 SsKey=0invokeId=0resultExp=Fbpda=F pi.piid=30 pi.l=2 pi2.piid=30 pi2.l=0 pi3.piid=30 pi3.l=0IpAddrMode=0 ipAddrType=0 ipv4=0.0.0.0:0 ctiActive=F ctiFarEndDev=2 ctiCMId=3 media=2 lPart=d8997e28-66b6-7783-a5d6-8f46ef5da368 lPatt=4168 lModNum=tn=0npi=0ti=1nd=4168pi=1si1 lName=locale: 1 Name: India Test UnicodeName: India Test pi: 1 cName=locale: 1 Name: UnicodeName: pi: 0 cn:tn=0npi=0ti=1nd=80016326635801pi=0si1 cVMbox= localPatternUsage=2 connectedPatternUsage=5 lCnPart=e769cfa2-b6d1-b061-1b87-a7658c93fe72 lCnPatt=8.001! rn:tn=0npi=0ti=1nd=80016326635801pi=0si1 lLRPart=e769cfa2-b6d1-b061-1b87-a7658c93fe72 lLRPatt=8.001! lOCdpnPart=e769cfa2-b6d1-b061-1b87-a7658c93fe72 lOCdpnPatt=8.001! oCdpn:tn=0npi=0ti=1nd=80016326635801pi=0si1 oRFR =0 lBridgePartID= lCnBridgePartID= DevCEPN=b5ecff4d-f135-76cd-34a3-6e8adb746d1e lineCEPN=ce6c846c-e5d4-214d-8880-68250ba32103 CnDevCEPN=5265beb5-b11c-6218-5580-48cabc44d380 lrnCEPN=183b0e69-399b-9bd6-a49f-48a4ee886155 oCdpnCEPN=183b0e69-399b-9bd6-a49f-48a4ee886155 lHPMemCEPN= cHPMemCEPN=Supp DTMF=1DTMF Cfg=1DTMF Payload=0 isOffNetDev=T protected=1 geolocInfo={geolocPkid=, filterPkid=, geolocVal=, devType=3} locPkid= locName= deductBW=F fateShareId= videoTrafficClass=0TransparentData=null CanSupportSIPTandN=false TransId=0 AllowBitMask=0x0 UserAgentOrServer= OrigDDName=locale: 1 Name: UnicodeName: pi: 0 mCallerId= mCallerName= ignoreEarlyMedia=F
57315659.000 |22:32:55.911 |SdlSig |CcNotifyReq |call_delivered4 |StationCdpc(3,100,59,74928) |StationD(3,100,58,2238) |3,100,13,119955.4^10.130.3.9^* |[R:N-H:0,N:5,L:0,V:0,Z:0,D:0] CI=60724230 CI.branch=0 lPart=d8997e28-66b6-7783-a5d6-8f46ef5da368 lPatt=4168 lModNum=tn=0npi=0ti=1nd=4168pi=1si1 lName=locale: 1 Name: India Test UnicodeName: India Test pi: 1 cName=locale: 1 Name: UnicodeName: pi: 0 cn:tn=0npi=0ti=1nd=0016326635801pi=1si0 cVMbox= localPatternUsage=2 connectedPatternUsage=5 lCnPart=e769cfa2-b6d1-b061-1b87-a7658c93fe72 lCnPatt=8.001! rn:tn=0npi=0ti=1nd=80016326635801pi=0si1 lLRPart=e769cfa2-b6d1-b061-1b87-a7658c93fe72 lLRPatt=8.001! lOCdpnPart=e769cfa2-b6d1-b061-1b87-a7658c93fe72 lOCdpnPatt=8.001! oCdpn:tn=0npi=0ti=1nd=80016326635801pi=0si1 oRFR =0 lBridgePartID= lCnBridgePartID= DevCEPN=b5ecff4d-f135-76cd-34a3-6e8adb746d1e lineCEPN=ce6c846c-e5d4-214d-8880-68250ba32103 CnDevCEPN=5265beb5-b11c-6218-5580-48cabc44d380 lrnCEPN=183b0e69-399b-9bd6-a49f-48a4ee886155 oCdpnCEPN=183b0e69-399b-9bd6-a49f-48a4ee886155 lHPMemCEPN= cHPMemCEPN= onBehalf= whichSide=1 holdFlag=0 notifyMsg=locale: 1 Name: UnicodeName: promptMsg=locale: 1 Name: UnicodeName: apply Instr=0 s.sv=0 promptMsg.userLocale=1 cgDevName=SEP64D989C258FC ctiActive=F ctiFarEndDev=2 ctiCCMId=3 ctiEvt.evtType=0 ctiEvt.transId=0 ctiEvt.ED.succ=F ctiEvt.PD.ParkPart= secureStatus=(F,0) callState=4 media=1 bitMask=80800000 Supp DTMF=1DTMF Cfg=1DTMF Payload=0 notifiedDName= connType=0 connStatus=0newPL=5newPLDmn=0 networkDomain=suppressMOH=F triggerByJoin=F NotifInd= ni.niid=39 ni.l=0 ni.nnd=0deviceCepn= partitionSearchSpace= geolocInfo=null locPkid= locName= deductBW=F fateShareId= videoTrafficClass=0 dtmMcNodeId=0 dtmCurrentCi=0 isOffNetDevice=T ignCntH=F cmDeviceType=7 ssCause=0TransparentData=null CanSupportSIPTandN=false TransId=0 AllowBitMask=0x0 UserAgentOrServer= OrigDDName=locale: 1 Name: UnicodeName: pi: 0 mCallerId= mCallerName= FDataType=0opId=0ssType=0 SsKey=0invokeId=0resultExp=Fbpda=F mobilityEventType=0 CallInstanceNumber=0
57315660.000 |22:32:55.911 |SdlSig |StationOutputCallState |restart0 |StationD(3,100,58,2238) |StationCdpc(3,100,59,74928) |3,100,13,119955.3^10.130.3.9^* |[R:N-H:0,N:5,L:0,V:0,Z:0,D:0] State=3 privacy=0 Line=1 CI=60724230 SCCP P-level=4 P-Domain=0
57315660.001 |22:32:55.911 |AppInfo |StationD: (0002238) CallState callState=3 lineInstance=1 callReference=60724230 privacy=0 sccp_precedenceLv=4 precedenceDm=0
57315661.000 |22:32:55.911 |SdlSig |StationOutputSelectSoftKeys |restart0 |StationD(3,100,58,2238) |StationCdpc(3,100,59,74928) |3,100,13,119955.3^10.130.3.9^* |[R:N-H:0,N:4,L:0,V:0,Z:0,D:0] Line=1 CI=60724230 SKIndex=8 Mask=ffffffff
57315661.001 |22:32:55.911 |AppInfo |StationD: (0002238) SelectSoftKeys instance=1 reference=60724230 softKeySetIndex=8 validKeyMask=ffffffff.
57315662.000 |22:32:55.911 |SdlSig |StationOutputDisplayPromptStatus |restart0 |StationD(3,100,58,2238) |StationCdpc(3,100,59,74928) |3,100,13,119955.3^10.130.3.9^* |[R:N-H:0,N:3,L:0,V:0,Z:0,D:0] TimeOut=0 Status= UnicodeStatus= Line=1 CI=60724230
57315662.001 |22:32:55.911 |AppInfo |StationD: (0002238) DisplayPromptStatus timeOut=0 Status='' content='Ring Out' line=1 CI=60724230 ver=85720016.
57315663.000 |22:32:55.911 |SdlSig |StationOutputCallInfo |restart0 |StationD(3,100,58,2238) |StationCdpc(3,100,59,74928) |3,100,13,119955.3^10.130.3.9^* |[R:N-H:0,N:2,L:0,V:0,Z:0,D:0] cdpn="80016326635801" cdpnVMB="" cdpnParty="locale: 1 Name: UnicodeName: pi: 0" cgpn="4168" cgpnVMB="" cgpnParty="locale: 1 Name: India Test UnicodeName: India Test pi: 1" oCdpn="80016326635801" oCdpnVMb="" oCdpnParty="locale: 1 Name: UnicodeName: pi: 0" OCdpnReason="0" lCdpn="80016326635801" lCdpnVMb="" lCdpnParty="locale: 1 Name: UnicodeName: pi: 0" lCdpnReason="0" line="1" CI="60724230" callInstance="1" callType="2" CallSecurityStatusType="0" restrictionBits="0" huntPilot="" huntPilotParty="locale: 1 Name: UnicodeName: pi: 0"
57315663.001 |22:32:55.911 |AppInfo |StationD: (0002238) (3,100,13,119947) CallInfo callingPartyName='India Test' callingParty=4168 cgpnVoiceMailbox= alternateCallingParty= calledPartyName='' calledParty=80016326635801 cdpnVoiceMailbox= originalCalledPartyName='' originalCalledParty=80016326635801 originalCdpnVoiceMailbox= originalCdpnRedirectReason=0 lastRedirectingPartyName='' lastRedirectingParty=80016326635801 lastRedirectingVoiceMailbox= lastRedirectingReason=0 callType=2(OutBound) lineInstance=1 callReference=60724230. version: 85720016
57315664.000 |22:32:55.911 |SdlSig |DSetCallState |restart0 |StationD(3,100,58,2238) |StationCdpc(3,100,59,74928) |3,100,13,119955.3^10.130.3.9^* |[R:N-H:0,N:1,L:0,V:0,Z:0,D:0] CallState = call_delivered4
57315664.001 |22:32:55.911 |AppInfo |StationD: (0002238) DEBUG- star_DSetCallState(7) State of cdpc(74928) is 6.
57315665.000 |22:32:55.911 |SdlSig |StationOutputCallInfo |restart0 |StationD(3,100,58,2238) |StationCdpc(3,100,59,74928) |3,100,13,119955.4^10.130.3.9^* |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0] cdpn="0016326635801" cdpnVMB="" cdpnParty="locale: 1 Name: UnicodeName: pi: 0" cgpn="4168" cgpnVMB="" cgpnParty="locale: 1 Name: India Test UnicodeName: India Test pi: 1" oCdpn="80016326635801" oCdpnVMb="" oCdpnParty="locale: 1 Name: UnicodeName: pi: 0" OCdpnReason="0" lCdpn="80016326635801" lCdpnVMb="" lCdpnParty="locale: 1 Name: UnicodeName: pi: 0" lCdpnReason="0" line="1" CI="60724230" callInstance="1" callType="2" CallSecurityStatusType="0" restrictionBits="0" huntPilot="" huntPilotParty="locale: 1 Name: UnicodeName: pi: 0"
57315665.001 |22:32:55.911 |AppInfo |StationD: (0002238) (3,100,13,119947) CallInfo callingPartyName='India Test' callingParty=4168 cgpnVoiceMailbox= alternateCallingParty= calledPartyName='' calledParty=0016326635801 cdpnVoiceMailbox= originalCalledPartyName='' originalCalledParty=80016326635801 originalCdpnVoiceMailbox= originalCdpnRedirectReason=0 lastRedirectingPartyName='' lastRedirectingParty=80016326635801 lastRedirectingVoiceMailbox= lastRedirectingReason=0 callType=2(OutBound) lineInstance=1 callReference=60724230. version: 85720016
57315666.000 |22:32:55.911 |Created | | |SdlTCPConnection(3,100,13,119956) |SdlTCPConnector(3,100,12,106644) | |NumOfCurrentInstances: 101
57315667.000 |22:32:55.911 |Stopping | | |SdlTCPConnector(3,100,12,106644) |SdlTCPConnector(3,100,12,106644) | |NumOfCurrentInstances: 1
57315668.000 |22:32:55.912 |SdlSig |H245TcpConnectionInfo |waitForSdlRsp |TranslateAndTransport(3,100,21,53370) |H245Handler(3,100,29,1) |3,100,12,106644.1^*^* |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0]
57315668.001 |22:32:55.912 |AppInfo |TranslateAndTransport(53370)::waitForSdlRsp_H245TcpConnectionInfo - received H245TcpConnectionInfo from H245Handler
57315669.000 |22:32:55.912 |SdlSig |TtControlChannelEstablished |waitForTransportEstablishment |H245SessionManager(3,100,28,53370) |TranslateAndTransport(3,100,21,53370) |3,100,12,106644.1^*^* |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0]
57315670.000 |22:32:55.915 |SdlSig |SdlDataInd |wait |H245Handler(3,100,29,1) |SdlTCPConnection(3,100,13,119956) |3,100,13,119956.2^*^* |*TraceFlagOverrode
57315670.001 |22:32:55.915 |AppInfo |H245ASN - TtPid=(53370) [0xb42276e0 1964 bytes] -Incoming #455653 -value MultimediaSystemControlMessage ::= request : terminalCapabilitySet :
sequenceNumber 1,
protocolIdentifier { 0 0 8 245 0 7 },
multiplexCapability h2250Capability :
maximumAudioDelayJitter 20,
receiveMultipointCapability
multicastCapability FALSE,
multiUniCastConference FALSE,
mediaDistributionCapability
centralizedControl FALSE,
distributedControl FALSE,
centralizedAudio FALSE,
distributedAudio FALSE,
centralizedVideo FALSE,
distributedVideo FALSE
transmitMultipointCapability
multicastCapability FALSE,
multiUniCastConference FALSE,
mediaDistributionCapability
centralizedControl FALSE,
distributedControl FALSE,
centralizedAudio FALSE,
distributedAudio FALSE,
centralizedVideo FALSE,
distributedVideo FALSE
receiveAndTransmitMultipointCapability
multicastCapability FALSE,
multiUniCastConference FALSE,
mediaDistributionCapability
centralizedControl FALSE,
distributedControl FALSE,
centralizedAudio FALSE,
distributedAudio FALSE,
centralizedVideo FALSE,
distributedVideo FALSE
mcCapability
centralizedConferenceMC FALSE,
decentralizedConferenceMC FALSE
rtcpVideoControlCapability FALSE,
mediaPacketizationCapability
h261aVideoPacketization FALSE
logicalChannelSwitchingCapability FALSE,
t120DynamicPortCapability FALSE
capabilityTable
capabilityTableEntryNumber 27,
capability receiveUserInputCapability : basicString : NULL
capabilityTableEntryNumber 3,
capability receiveAudioCapability : g711Ulaw64k : 20
capabilityDescriptors
capabilityDescriptorNumber 1,
simultaneousCapabilities
3
27
57315670.002 |22:32:55.915 |AppInfo |DET-H245Log-- : H323-2833. H245CapabilityDefinition lookupOutBandSignalCapEntry: entryNumber=27, receiveInputCap=2
57315670.003 |22:32:55.915 |AppInfo |DET-H245Log-- : H323-2833. H245CapabilityDefinition lookupOutBandSignalCapEntry: No SignalType UserInputCapability, put Alphanumeric type back, entryNumber=27, UserCap=2,
57315671.000 |22:32:55.915 |SdlSig |CeseTerminalCapabilitySet |wait |TranslateAndTransport(3,100,21,53370) |H245Handler(3,100,29,1) |3,100,13,119956.2^*^* |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0] SeqNo=1 len=1964 TCS heap-> 0xb42276e0 capCount=2
57315672.000 |22:32:55.915 |SdlSig |SdlDataInd |wait |H245Handler(3,100,29,1) |SdlTCPConnection(3,100,13,119956) |3,100,13,119956.3^*^* |*TraceFlagOverrode
57315672.001 |22:32:55.915 |AppInfo |H245ASN - TtPid=(53370) [0xb27ceab8 1444 bytes] -Incoming #455654 -value MultimediaSystemControlMessage ::= request : masterSlaveDetermination :
terminalType 60,
statusDeterminationNumber 8220
57315673.000 |22:32:55.915 |SdlSig |MsdseMasterSlaveDetermination |wait |TranslateAndTransport(3,100,21,53370) |H245Handler(3,100,29,1) |3,100,13,119956.3^*^* |[R:N-H:0,N:1,L:0,V:0,Z:0,D:0]
57315674.000 |22:32:55.915 |SdlSig |CeseTerminalCapabilitySet |paused |CeseIncoming(3,100,20,53370) |TranslateAndTransport(3,100,21,53370) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:1,L:0,V:0,Z:0,D:0] SeqNo=1 len=1964 TCS heap-> 0xb42276e0 capCount=2
57315675.000 |22:32:55.915 |SdlSig |MsdseMasterSlaveDetermination |paused |Msdse(3,100,23,53370) |TranslateAndTransport(3,100,21,53370) |3,100,13,119956.3^10.130.3.9^Port 49839 |[R:N-H:0,N:1,L:0,V:0,Z:0,D:0]
57315676.000 |22:32:55.915 |SdlSig-S |MsdseMasterSlaveDetermination |paused |Msdse(3,100,23,53370) |TranslateAndTransport(3,100,21,53370) |3,100,13,119956.3^10.130.3.9^Port 49839 |
57315677.000 |22:32:55.915 |SdlSig |CeseTransferIndication |capabilityExchange |H245SessionManager(3,100,28,53370) |CeseIncoming(3,100,20,53370) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0] len=1964 TCS heap-> 0xb42276e0 capCount=2
57315678.000 |22:32:55.915 |SdlSig |H245CapabilitiesIncomingIndication |waitForCapabilitiesExchange |H245Interface(3,100,185,53370) |H245SessionManager(3,100,28,53370) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:1,L:0,V:0,Z:0,D:0] len=1964 TCS heap-> 0xb42276e0 capCount=2
57315678.001 |22:32:55.915 |AppInfo |DET-H245Interface-(53370), star_H245CapabilitiesIncoming , received sdpMode = 0, videoCapable=0, dataCapable=0, videoSetupAfterAudio= 0,mMXOfferNeeded= 0
57315678.002 |22:32:55.915 |AppInfo |DET-H245Interface-(53370)::convertH245CapabilitiesToCapabilities, H323-2833. Incoming OOB user Input Cap choice =2, oobUserInputCap=1
57315678.003 |22:32:55.915 |AppInfo |DET-H245Interface-(53370)::convertH245CapabilitiesToCapabilities, cmCloudH245ICTVersion(0), H323-2833. After incoming TCS DTMFProfile updated, DTMF Method=1, 2833 payloadNum=0, OOB cap=1
57315678.004 |22:32:55.915 |AppInfo |DET-MediaUtility-::getCodecPrefOption, xferModeA=7 xferModeB=4 honorOfferCodecPrefA=0 honorOfferCodecPrefB=0 PREF_LIST
57315678.005 |22:32:55.915 |AppInfo |DET-MediaUtility-::setCodecPrefOptionAndRegionB, audioPassThru=0 myRegion=SIN-REG peerRegion=SIN-REG farEndRegion= regionB=SIN-REG PREF_LIST
57315678.006 |22:32:55.915 |AppInfo |DET-H245Interface-(53370)::setCodecPrefOptionAndOtherSideRegion, otherSideRegion=SIN-REG, PREF_LIST
57315678.007 |22:32:55.915 |AppInfo |DET-RegionsServer::sortMediaPayload-capCount=1, regionA=SIN-REG, regionB=SIN-REG, fkCodecList=911b707a-7d0e-c4cb-cc2a-89b4178491da
57315678.008 |22:32:55.915 |AppInfo |DET-MediaUtility-::getCodecPrefOption, xferModeA=7 xferModeB=4 honorOfferCodecPrefA=0 honorOfferCodecPrefB=0 PREF_LIST
57315678.009 |22:32:55.915 |AppInfo |DET-MediaUtility-::setCodecPrefOptionAndRegionB, audioPassThru=0 myRegion=SIN-REG peerRegion=SIN-REG farEndRegion= regionB=SIN-REG PREF_LIST
57315678.010 |22:32:55.915 |AppInfo |DET-H245Interface-(53370)::setCodecPrefOptionAndOtherSideRegion, otherSideRegion=SIN-REG, PREF_LIST
57315678.011 |22:32:55.915 |AppInfo |DET-RegionsServer::matchCapabilities-- savedOption=1, PREF_LIST, regionA=SIN-REG regionB=SIN-REG latentCaps(A=0, B=0) kbps=64, capACount=1, capBCount=0
57315678.012 |22:32:55.915 |AppInfo |RegionsServer: applyCodecFilterIfNeeded - no codecs remained after filtering so restored original 0 caps
57315679.000 |22:32:55.915 |SdlSig |CeseTransferResponse |paused |CeseIncoming(3,100,20,53370) |H245SessionManager(3,100,28,53370) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:1,L:0,V:0,Z:0,D:0]
57315680.000 |22:32:55.915 |SdlSig-S |CeseTransferResponse |paused |CeseIncoming(3,100,20,53370) |H245SessionManager(3,100,28,53370) |3,100,13,119956.2^10.130.3.9^Port 49839 |
57315681.000 |22:32:55.915 |SdlSig |MXCapabilitiesIncoming |waitInterfacesCapabilities |MediaExchange(3,100,138,127079) |H245Interface(3,100,185,53370) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0] audioCapCount=1 Caps[4(20)] videoCapCount=0, [] extendedVidCount=0, [] Supp.Payload RFC[0 0 0 0 0 ] h245ICTVersion0 useOldGWBytesForGSMConversion=F cryptoCapCount=0 cryptoVidDataCapCount=0 DTMF Profile(1,1,0,1,F)LatentCaps=null
57315681.001 |22:32:55.915 |AppInfo |DET-MediaExchange-(127079)::canForwardCapsToOtherEnd, activeCapEnabled(0, 0), canForwardCapsToOtherEnd=0
57315681.002 |22:32:55.915 |AppInfo |DET-MediaExchange-(127079)::finishCapExchange, capFromTwoIFs=1,capFromFarEnd=0,aPT=0,vPT=2,capE2E=0,capDone=1
57315681.003 |22:32:55.915 |AppInfo |DET-MediaExchange-(127079)::handleInterfaceVisited, returned finishCapExchange
57315681.004 |22:32:55.915 |AppInfo |DET-MediaExchange-(127079)::handleInterfaceVisited, allowReConnect(1) partyAHasCapsorACE(1)partybHasCapsorACE (0)
57315682.000 |22:32:55.915 |SdlSig |AuReConnectRequest |waitDisconnect |MediaManager(3,100,133,117676) |MediaExchange(3,100,138,127079) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0] Party1: MR=0 CI=60724230 audioCapCount=9 region=SIN-REG xferMode=4 mrid=0 audioId=0 MMCap=0x1 activeCap=0 cryptoCapCount=0 flushIns=0 dtmCall=0 dtmPrimaryCI=0 IFPid=(3,100,234,63534) dtMedia=F honorCodec=F EOType=0 MohType=0DTMF Caps(1,1,0,1,F) confID=0 connType=3 connStatus=0 mtpPre=F teleEve=0 IFCreated=T IFHandling=0 FS=0 mcNodeId=0LatentCaps=null Party2: MR=0 CI=60724231 audioCapCount=1 region=SIN-REG xferMode=7 mrid=0 audioId=0 MMCap=0x9 activeCap=0 cryptoCapCount=0 flushIns=0 dtmCall=0 dtmPrimaryCI=0 IFPid=(3,100,185,53370) dtMedia=F honorCodec=F EOType=0 MohType=0DTMF Caps(1,1,0,1,F) confID=0 connType=3 connStatus=0 mtpPre=F teleEve=0 IFCreated=T IFHandling=0 FS=0 mcNodeId=0LatentCaps=null reConnType=0 videoCall=F AllowedCallType=0x0 mtpChanged=F precLvl=5 resCap=0 party1.mMediaCoordinatorNodeId=0 party2.mMediaCoordinatorNodeId=0
57315682.001 |22:32:55.915 |AppInfo |SIG-MediaManager-(117676)::waitDisconnect_AuReConnectRequest, reConnectType(0)
57315682.002 |22:32:55.915 |AppInfo |DET-MediaManager-(117676)::waitDisconnect_AuReConnectRequest, Update AuConnectRequestMsg party capability. isDeviceVideoCapable (party1=0, party2=0) AllowedCallType=0x00000000
57315682.003 |22:32:55.915 |AppInfo |DET-MediaManager-(117676)::waitDisconnect_AuReConnectRequest, ReConnect--sending disconnect, Party1DTMFmethod(1) Party2DTMFMethod(1)
57315683.000 |22:32:55.915 |SdlSig |AuDisconnectRequest |waitCleanup |MediaManager(3,100,133,117676) |MediaManager(3,100,133,117676) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0] CI1=60724230 CI2=60724231 sc=0 disconnType=0 ssReason=1 clearType=0 IF1Created=T IF2Created=T party1.mMediaCoordinatorNodeId=0 party2.mMediaCoordinatorNodeId=0 party1.dtmCall=0 party2.dtmCall=0 reconnectPending=F forceStop=F
57315683.001 |22:32:55.915 |AppInfo |!!ERROR!! -MediaManager-(117676)::handle_AuDisconnectRequest, mCleanupPreallocatedMTP=0
57315683.002 |22:32:55.915 |AppInfo |DET-MediaManager-(117676)::handle_AuDisconnectRequest, mrid(0,0) ci(6072423060724231) size(1), dt(0)
57315683.003 |22:32:55.915 |AppInfo |DET-MediaManager-(117676)::keepMTPConnection, sr(1), resrcAllocateSide(0), party1CI(60724230), bRet(0)
57315683.004 |22:32:55.915 |AppInfo |DET-MediaManager-(117676) - sendDisconnectReqToMX - disconnType=0, Party1DTMFmethod(1) Party2DTMFMethod(1) party1capCount(9) party2capCount(0), MC(0,0), deviceVideoCap(0, 0)
57315684.000 |22:32:55.915 |SdlSig |AuDisconnectRequest |waitInterfacesCapabilities |MediaExchange(3,100,138,127079) |MediaManager(3,100,133,117676) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0] CI1=60724230 CI2=60724231 sc=0 disconnType=0 ssReason=1 clearType=0 IF1Created=T IF2Created=T party1.mMediaCoordinatorNodeId=0 party2.mMediaCoordinatorNodeId=0 party1.dtmCall=0 party2.dtmCall=0 reconnectPending=F forceStop=F
57315684.001 |22:32:55.915 |AppInfo |DET-MediaExchange-(127079)::wait_Disconnect, dt=0,stReason=1,IFHandling(0,0)
57315685.000 |22:32:55.915 |SdlSig-Q |MXInterfaceEstablished |waitStopped |MediaExchange(3,100,138,127079) |AgenaInterface(3,100,234,63534) |3,100,13,119955.3^10.130.3.9^* |
57315686.000 |22:32:55.915 |SdlSig-D |MXInterfaceEstablished |waitStopped |MediaExchange(3,100,138,127079) |AgenaInterface(3,100,234,63534) |3,100,13,119955.3^10.130.3.9^* |
57315687.000 |22:32:55.915 |SdlSig |MXInterfaceStopStreaming |waitForMXCapabilitiesorOfferorAnswer |AgenaInterface(3,100,234,63534) |MediaExchange(3,100,138,127079) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:1,L:0,V:0,Z:0,D:0] ClearType= 0 StoppedBy= 0 DisconnecType= 0 StopStreamingReason=1 reconPending= FmHoldingPartyCI= 0mForceStop= F
57315687.001 |22:32:55.915 |AppInfo |DET-AgenaInterfaceBase-(63534)::closeRecvForAllAudioChannels, mAudioIncomingLC2AGIDMap size = 0
57315688.000 |22:32:55.915 |SdlSig |MXInterfaceStopStreaming |waitForCapabilitiesExchange |H245Interface(3,100,185,53370) |MediaExchange(3,100,138,127079) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:1,L:0,V:0,Z:0,D:0] ClearType= 0 StoppedBy= 0 DisconnecType= 0 StopStreamingReason=1 reconPending= FmHoldingPartyCI= 0mForceStop= F
57315688.001 |22:32:55.915 |AppInfo |DET-H245Interface-(53370)::handleStopStreaming, stopStreamingRecdInWaitForCapExchgState=0
57315688.002 |22:32:55.915 |AppInfo |DET-H245Interface-(53370)::handleStopStreaming, mInitialCallSendDumyCapsIfNeeded=1
57315689.000 |22:32:55.915 |SdlSig |MXInterfaceStoppedStreaming |waitStopped |MediaExchange(3,100,138,127079) |AgenaInterface(3,100,234,63534) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:1,L:0,V:0,Z:0,D:0]
57315690.000 |22:32:55.915 |SdlSig |MXInterfaceStoppedStreaming |waitStopped |MediaExchange(3,100,138,127079) |H245Interface(3,100,185,53370) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0]
57315690.001 |22:32:55.915 |Stopping | | |MediaExchange(3,100,138,127079) |MediaExchange(3,100,138,127079) | |NumOfCurrentInstances: 1
57315691.000 |22:32:55.915 |SdlSig |MXNewParentPid |restart |AgenaInterface(3,100,234,63534) |MediaExchange(3,100,138,127079) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:2,L:0,V:0,Z:0,D:0] parent pid:nodeId=3.PN=134.PI=1.vPT=2 allow2833=F injectDigitstoMTP=F subscribetoMTP=F passthru2833=F
57315692.000 |22:32:55.915 |SdlSig |MXNewParentPid |waitReconnect |H245Interface(3,100,185,53370) |MediaExchange(3,100,138,127079) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:1,L:0,V:0,Z:0,D:0] parent pid:nodeId=3.PN=134.PI=1.vPT=2 allow2833=F injectDigitstoMTP=F subscribetoMTP=F passthru2833=F
57315693.000 |22:32:55.915 |SdlSig |AuDisconnectReply |waitCleanup |MediaManager(3,100,133,117676) |MediaExchange(3,100,138,127079) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0] CI1=60724230 CI2=60724231 sc=0 disconnType=0 ssReason=1 clearType=0 IF1Created=T IF2Created=T party1.mMediaCoordinatorNodeId=0 party2.mMediaCoordinatorNodeId=0 party1.dtmCall=0 party2.dtmCall=0 reconnectPending=F forceStop=F
57315693.001 |22:32:55.915 |AppInfo |DET-MediaManager-(117676)::waitCleanup_AuDisconnectReply, CI(60724230,60724231), disconnType(0), stopStreamingReason(1) DTMFMethod(1 1),MC(0,0),rf(1), nD(1,1)
57315693.002 |22:32:55.915 |AppInfo |DET-MediaManager-(117676)::waitCleanup_AuDisconnectReply, videoCap (0, 0), AllowedCallType=0
57315693.003 |22:32:55.915 |AppInfo |SIG-MediaManager-(117676)::waitCleanup_AuDisconnectReply - recv all disconn replies, send ReConnReq to MC, reConnectType(0), party(60724230,60724231) mrid(0 0) party1DTMF(1 1 0) part2DTMF(1 1 0), MC(0,0), deviceVideo (0, 0), AllowedCallType=0x00000000
57315693.004 |22:32:55.915 |Stopping | | |MediaManager(3,100,133,117676) |MediaManager(3,100,133,117676) | |NumOfCurrentInstances: 1
57315694.000 |22:32:55.915 |SdlSig |AuReConnectRequest |wait |MediaCoordinator(3,100,134,1) |MediaManager(3,100,133,117676) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0] Party1: MR=0 CI=60724230 audioCapCount=9 region=SIN-REG xferMode=4 mrid=0 audioId=0 MMCap=0x1 activeCap=0 cryptoCapCount=0 flushIns=0 dtmCall=0 dtmPrimaryCI=0 IFPid=(3,100,234,63534) dtMedia=F honorCodec=F EOType=0 MohType=0DTMF Caps(1,1,0,1,F) confID=0 connType=3 connStatus=0 mtpPre=F teleEve=0 IFCreated=T IFHandling=0 FS=0 mcNodeId=0LatentCaps=null Party2: MR=0 CI=60724231 audioCapCount=1 region=SIN-REG xferMode=7 mrid=0 audioId=0 MMCap=0x9 activeCap=0 cryptoCapCount=0 flushIns=0 dtmCall=0 dtmPrimaryCI=0 IFPid=(3,100,185,53370) dtMedia=F honorCodec=F EOType=0 MohType=0DTMF Caps(1,1,0,1,F) confID=0 connType=3 connStatus=0 mtpPre=F teleEve=0 IFCreated=T IFHandling=0 FS=0 mcNodeId=0LatentCaps=null reConnType=0 videoCall=F AllowedCallType=0x0 mtpChanged=F precLvl=5 resCap=0 party1.mMediaCoordinatorNodeId=0 party2.mMediaCoordinatorNodeId=0
57315694.001 |22:32:55.915 |AppInfo |SIG-MediaCoordinator-wait_AuReConnectRequest, reConnectType(0)
57315694.002 |22:32:55.915 |AppInfo |SIG-MediaCoordinator-wait_AuReConnectRequest - removing MediaManager(117676) from connection list
57315695.000 |22:32:55.915 |SdlSig |AuConnectRequest |wait |MediaCoordinator(3,100,134,1) |MediaCoordinator(3,100,134,1) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0] Party1: MR=0 CI=60724230 audioCapCount=9 region=SIN-REG xferMode=4 mrid=0 audioId=0 MMCap=0x1 activeCap=0 cryptoCapCount=0 flushIns=0 dtmCall=0 dtmPrimaryCI=0 IFPid=(3,100,234,63534) dtMedia=F honorCodec=F EOType=0 MohType=0DTMF Caps(1,1,0,1,F) confID=0 connType=3 connStatus=0 mtpPre=F teleEve=0 IFCreated=T IFHandling=0 FS=0 mcNodeId=0LatentCaps=null Party2: MR=0 CI=60724231 audioCapCount=1 region=SIN-REG xferMode=7 mrid=0 audioId=0 MMCap=0x9 activeCap=0 cryptoCapCount=0 flushIns=0 dtmCall=0 dtmPrimaryCI=0 IFPid=(3,100,185,53370) dtMedia=F honorCodec=F EOType=0 MohType=0DTMF Caps(1,1,0,1,F) confID=0 connType=3 connStatus=0 mtpPre=F teleEve=0 IFCreated=T IFHandling=0 FS=0 mcNodeId=0LatentCaps=null reConnType=0 videoCall=F AllowedCallType=0x0 mtpChanged=F precLvl=5 resCap=0 party1.mMediaCoordinatorNodeId=0 party2.mMediaCoordinatorNodeId=0
57315695.001 |22:32:55.915 |Created | | |MediaManager(3,100,133,117677) |MediaCoordinator(3,100,134,1) | |NumOfCurrentInstances: 1
57315695.002 |22:32:55.915 |AppInfo |SIG-MediaCoordinator-wait_AuConnectRequest - new MediaManager(133,117677) started
57315696.000 |22:32:55.915 |SdlSig |AuConnectRequest |waitConnectRequest |MediaManager(3,100,133,117677) |MediaCoordinator(3,100,134,1) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0] Party1: MR=0 CI=60724230 audioCapCount=9 region=SIN-REG xferMode=4 mrid=0 audioId=0 MMCap=0x1 activeCap=0 cryptoCapCount=0 flushIns=0 dtmCall=0 dtmPrimaryCI=0 IFPid=(3,100,234,63534) dtMedia=F honorCodec=F EOType=0 MohType=0DTMF Caps(1,1,0,1,F) confID=0 connType=3 connStatus=0 mtpPre=F teleEve=0 IFCreated=T IFHandling=0 FS=0 mcNodeId=0LatentCaps=null Party2: MR=0 CI=60724231 audioCapCount=1 region=SIN-REG xferMode=7 mrid=0 audioId=0 MMCap=0x9 activeCap=0 cryptoCapCount=0 flushIns=0 dtmCall=0 dtmPrimaryCI=0 IFPid=(3,100,185,53370) dtMedia=F honorCodec=F EOType=0 MohType=0DTMF Caps(1,1,0,1,F) confID=0 connType=3 connStatus=0 mtpPre=F teleEve=0 IFCreated=T IFHandling=0 FS=0 mcNodeId=0LatentCaps=null reConnType=0 videoCall=F AllowedCallType=0x0 mtpChanged=F precLvl=5 resCap=0 party1.mMediaCoordinatorNodeId=0 party2.mMediaCoordinatorNodeId=0
57315696.001 |22:32:55.915 |AppInfo |SIG-MediaManager-(117677)::wait_AuConnectRequest, CI(60724230,60724231), capCount(9,1), mcNodeId(0,0), xferMode(4,7), reConnectType(0), mrid (0, 0) IFCreated(1 1) proIns(63534 53370), AC(0,0), party1DTMF(1 1 0 1 0) party2DTMF(1 1 0 1 0),reConnFlag=1, connType(3,3), IFHand(0,0),MTP(0,0),MRGL(5212b81b-1ba4-b897-0a93-0125344b429e,5212b81b-1ba4-b897-0a93-0125344b429e) videoCap(0 0), mmCallType(0),FS(0,0), IpAddrMode(0 0) aPid(3, 58, 2238), bPid(3, 189, 55784) EOType(0 0) MOHAnnConnType(0 0) honorCodec(0 0)
57315697.000 |22:32:55.915 |SdlSig |CACInfoReq |wait |ReservationMgr(3,100,103,1) |MediaManager(3,100,133,117677) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0] CI= 0 aCI=60724230 bCI=60724231
57315698.000 |22:32:55.915 |SdlSig |CACInfoReq |active |LBMInterface(3,100,169,1) |ReservationMgr(3,100,103,1) |3,100,13,119956.2^10.130.3.9^Port 49839 |[T:N-H:0,N:0,L:0,V:0,Z:0,D:0] CI= 0 aCI=60724230 bCI=60724231
57315698.001 |22:32:55.915 |AppInfo |LBMIF: CI: 60724230 INFOREQ 3,100,58,2238
57315698.002 |22:32:55.915 |AppInfo |LBMIF: CI: 60724231 INFOREQ' 3,100,189,55784
57315699.000 |22:32:55.915 |SdlSig |CACInfoRes |wait |ReservationMgr(3,100,103,1) |LBMInterface(3,100,169,1) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0] CI= 0 aCI=60724230 bCI=60724231 pol=0 rsvpStatus=1 sessJoined=F staIdx_no_agent=0 AudioBWReserved eoSent=F aAgent: confID =0 ci =0 capCt =0 reg= mtpType =2 agentCt =0 mmCapSet=0 agentAllo =0 RemoAgent=F DevCap=0 ipAddrMode=0 bAgent: confID =0 ci =0 capCt =0 reg= mtpType =2 agentCt =0 mmCapSet=0 agentAllo =0 RemoAgent=F DevCap=0 ipAddrMode=0 aPort: NumPort =0 bPort: NumPort =0 otherAgentPort: NumPort =0
57315700.000 |22:32:55.915 |SdlSig |CACInfoReq |wait |RSVPSession(3,100,100,79309) |ReservationMgr(3,100,103,1) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0] CI= 0 aCI=60724230 bCI=60724231
57315701.000 |22:32:55.916 |SdlSig |CACInfoRes |wait |ReservationMgr(3,100,103,1) |RSVPSession(3,100,100,79309) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0] CI= 60724230 aCI=60724230 bCI=60724231 pol=0 rsvpStatus=1 sessJoined=F staIdx_no_agent=0 NoBWReserved eoSent=F aAgent: confID =0 ci =0 capCt =0 reg= mtpType =2 agentCt =0 mmCapSet=0 agentAllo =0 RemoAgent=F DevCap=0 ipAddrMode=0 bAgent: confID =0 ci =0 capCt =0 reg= mtpType =2 agentCt =0 mmCapSet=0 agentAllo =0 RemoAgent=F DevCap=0 ipAddrMode=0 aPort: NumPort =0 bPort: NumPort =0 otherAgentPort: NumPort =0
57315702.000 |22:32:55.916 |SdlSig |CACInfoRes |waitCACInfoRes |MediaManager(3,100,133,117677) |ReservationMgr(3,100,103,1) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0] CI= 0 aCI=60724230 bCI=60724231 pol=0 rsvpStatus=1 sessJoined=F staIdx_no_agent=0 AudioBWReserved eoSent=F aAgent: confID =0 ci =0 capCt =0 reg= mtpType =2 agentCt =0 mmCapSet=0 agentAllo =0 RemoAgent=F DevCap=0 ipAddrMode=0 bAgent: confID =0 ci =0 capCt =0 reg= mtpType =2 agentCt =0 mmCapSet=0 agentAllo =0 RemoAgent=F DevCap=0 ipAddrMode=0 aPort: NumPort =0 bPort: NumPort =0 otherAgentPort: NumPort =0
57315702.001 |22:32:55.916 |AppInfo |DET-MediaManager-(117677) - waitCACInfoRes_CACInfoRes- qosType=0 videoEsc=0 mNoVideoResvAttempted=1 VideoCall=0
57315702.002 |22:32:55.916 |AppInfo |DET-MediaManager-(117677)::waitCACInfoRes_CACInfoRes, rsvp(0,0), aE2ERegion(64) deviceAcaps(0) deviceBCaps(0),noVideoResv(1), mmAllowedCallType(0x00000000)
57315702.003 |22:32:55.916 |AppInfo |DET-MediaManager-(117677)::bothPartiesVideoCapable=0 MainVideoCap=0 SecondVideoCap=0
57315702.004 |22:32:55.916 |AppInfo |DET-MediaManager-(117677)::mapCapabilitiesToMMCallType, policy=0, hasRSVP=0, mainVideoCap=0,dataCap=0, allowedCallType=0x00000001, V region(e2e=384, 1)
57315702.005 |22:32:55.916 |AppInfo |DET-MediaManager-(117677)::buildMtpXcoderAllocList, savedConnectionCount=0, QosType=0
57315702.006 |22:32:55.916 |AppInfo |DET-RegionsServer::matchCapabilities-- savedOption=3, PREF_NONE, regionA=(null) regionB=(null) latentCaps(A=0, B=0) kbps=64, capACount=9, capBCount=1
57315702.007 |22:32:55.916 |AppInfo |DET-RegionsServer::handleMatchCapabilities()-- BEFORE MATCHING LOGIC applied(after filtering), sideARefCaps=1 refCapsSaveOpt=1 otherCapsSaveOpt=1 capsA[7]::capCount=7 (Cap,ptime)= (4,40) (2,40) (86,60) (15,60) (16,60) (11,60) (12,60) capsB[1]::capCount=1 (Cap,ptime)= (4,20)
57315702.008 |22:32:55.916 |AppInfo |DET-RegionsServer::handleMatchCapabilities()-- AFTER MATCHING LOGIC applied, capsA[1]::capCount=1 (Cap,ptime)= (4,40) capsB[1]::capCount=1 (Cap,ptime)= (4,20) numMatchedCaps=1
57315702.009 |22:32:55.916 |AppInfo |DET-MediaManager-(117677)::checkAudioPassThru, param(0,1), capCount(9,1), mtpPT=1, aPT=1
57315702.010 |22:32:55.916 |AppInfo |DET-MediaManager-(117677)::preCheckCapabilities, region1=SIN-REG, region2=SIN-REG, Pty1 capCount=9 (Cap,ptime)= (25,40) (4,40) (2,40) (86,60) (15,60) (16,60) (11,60) (12,60) (257,1), Pty2 capCount=1 (Cap,ptime)= (4,20)
57315702.011 |22:32:55.916 |AppInfo |DET-RegionsServer::matchCapabilities-- savedOption=0, PREF_NONE, regionA=(null) regionB=(null) latentCaps(A=0, B=0) kbps=64, capACount=9, capBCount=1
57315702.012 |22:32:55.916 |AppInfo |DET-RegionsServer::handleMatchCapabilities()-- BEFORE MATCHING LOGIC applied(after filtering), sideARefCaps=1 refCapsSaveOpt=0 otherCapsSaveOpt=0 capsA[7]::capCount=7 (Cap,ptime)= (4,40) (2,40) (86,60) (15,60) (16,60) (11,60) (12,60) capsB[1]::capCount=1 (Cap,ptime)= (4,20)
57315702.013 |22:32:55.916 |AppInfo |DET-RegionsServer::handleMatchCapabilities()-- AFTER MATCHING LOGIC applied, capsA[7]::capCount=7 (Cap,ptime)= (4,40) (2,40) (86,60) (15,60) (16,60) (11,60) (12,60) capsB[1]::capCount=1 (Cap,ptime)= (4,20) numMatchedCaps=1
57315702.014 |22:32:55.916 |AppInfo |DET-MediaManager-(117677)::prepareInitialConnectionList, Party1CapCount=9 Party2CapCount=1 XcoderRequired=0 xcodingSide=0 allowMTP=0
57315702.015 |22:32:55.916 |AppInfo |DET-MediaManager-(117677)::AllocateXcoderandMTPWithZeroSavedConnection
57315702.016 |22:32:55.916 |AppInfo |DET-MediaManager-(117677)::isMTPNeededForDTMF, both parties are non-SIP and we'll do OOB for DTMF
57315702.017 |22:32:55.916 |AppInfo |DET-MediaManager-(117677) - isIpv6CapableMTPNeeded(0) ipAddrMode(0 0) mtpside(0)
57315702.018 |22:32:55.916 |AppInfo |SIG-MediaManager-(117677)::handleAuConnectRequest, connCount=0
57315702.019 |22:32:55.916 |Created | | |MediaExchange(3,100,138,127080) |MediaManager(3,100,133,117677) | |NumOfCurrentInstances: 1
57315702.020 |22:32:55.916 |AppInfo |DET-MediaManager-(117677)::getAllow2833FortheCall, 2833 is not allowed for the call
57315703.000 |22:32:55.916 |SdlSig |AuConnectRequest |waitAuConnectRequest |MediaExchange(3,100,138,127080) |MediaManager(3,100,133,117677) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0] Party1: MR=0 CI=60724230 audioCapCount=9 region=SIN-REG xferMode=4 mrid=0 audioId=0 MMCap=0x1 activeCap=0 cryptoCapCount=0 flushIns=0 dtmCall=0 dtmPrimaryCI=0 IFPid=(3,100,234,63534) dtMedia=F honorCodec=F EOType=0 MohType=0DTMF Caps(1,1,0,1,F) confID=0 connType=3 connStatus=0 mtpPre=F teleEve=0 IFCreated=T IFHandling=0 FS=0 mcNodeId=0LatentCaps=null Party2: MR=0 CI=60724231 audioCapCount=1 region=SIN-REG xferMode=7 mrid=0 audioId=0 MMCap=0x9 activeCap=0 cryptoCapCount=0 flushIns=0 dtmCall=0 dtmPrimaryCI=0 IFPid=(3,100,185,53370) dtMedia=F honorCodec=F EOType=0 MohType=0DTMF Caps(1,1,0,1,F) confID=0 connType=3 connStatus=0 mtpPre=F teleEve=0 IFCreated=T IFHandling=0 FS=0 mcNodeId=0LatentCaps=null reConnType=0 videoCall=F AllowedCallType=0x1 mtpChanged=F precLvl=5 resCap=0 party1.mMediaCoordinatorNodeId=0 party2.mMediaCoordinatorNodeId=0
57315703.001 |22:32:55.916 |AppInfo |DET-MediaExchange-(127080)::wait_AuConnectRequest, party1(CI=60724230, MXIFHandling=0, MediaReq=0,xferMode=4),party2(CI=60724231, MXIFHandling=0,xferMode=7), farEnd(XferMode=0,AC=0,T38=0,FS=0, mediaReq=0), PT(2,2),connType(3,3), farendIpAddrMode=0
57315703.002 |22:32:55.916 |AppInfo |DET-MediaExchange-(127080)::addPartyAandPartyB, party1:xferMode=4, party2:xferMode=7
57315703.003 |22:32:55.916 |AppInfo |DET-MediaExchange-(127080)::setEOOpaqueMediaInfoForInterface, peerEarlyOfferType(0), farEndEarlyOfferType(0), h323EarlyOfferInfo(0)
57315703.004 |22:32:55.916 |AppInfo |DET-MediaExchange-(127080)::setEOOpaqueMediaInfoForInterface, peerEarlyOfferType(0), farEndEarlyOfferType(0), h323EarlyOfferInfo(0)
57315703.005 |22:32:55.916 |AppInfo |DET-MediaExchange-(127080)::setRegionAndHonorIncomingCodecPrefForInterface, Party1: myRegion(SIN-REG), peerRegion(SIN-REG), farEndRegion(), myHonorIncomingCodecPref(0), peerHonorIncomingCodecPref(0), farEndHonorIncomingCodecPref(0) Party2: myRegion(SIN-REG), peerRegion(SIN-REG), farEndRegion(), myHonorIncomingCodecPref(0), peerHonorIncomingCodecPref(0), farEndHonorIncomingCodecPref(0)
57315703.006 |22:32:55.916 |AppInfo |DET-MediaExchange-(127080)::addPartyAandPartyB, videoCall(0), normalStart(0), audioRegion(64), videoRegion(384), party1VidCapable(0), party2VidCapable(0)
57315704.000 |22:32:55.916 |SdlSig |MXNewParentPid |restart |AgenaInterface(3,100,234,63534) |MediaExchange(3,100,138,127080) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:3,L:0,V:0,Z:0,D:0] parent pid:nodeId=3.PN=138.PI=127080.vPT=2 allow2833=F injectDigitstoMTP=F subscribetoMTP=F passthru2833=F
57315705.000 |22:32:55.916 |SdlSig |MXInterfaceRestart |restart |AgenaInterface(3,100,234,63534) |MediaExchange(3,100,138,127080) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:2,L:0,V:0,Z:0,D:0] pid=0.0.0 partySide=A mAudioRegionKbps=64 mVideoRegionKbps=384 mMulticastFlag=F silence=F network=0 sessionId=0 ipAddrType=0 ipv4=10.130.3.9 port=49839 My DTMF (config=1 support=1 2833recvPT=0) Peer DTMF (config=1 support=1 2833recvPT=0) Media DTMF (subscribe=F passthru2833=F injectDigitstoMTP=F allow2833=F) cryptocapcount=0 RSVP nPortsOpened=0 AllowedCallType=0x1 aPT=2 vPT=2 e2eAR0 e2eVR0 capE2E=F dscpRSVPFail=F mtpChanged=F allowOneWay=F my connType=3 peer connType=3 mouth mode=0 mouth party=0 ear mode=0 ear party=0 mIsSRTP=T farEndXferM=0 farEndMediaReq=0 joined=F NormalReconn=T callIpAddrMode(my=0 peer=0 far=0) moh=F mediaReq=0 mPrecLvl=5 allowVideo=F SIPDevMediaInfo: [ vidPref=T isPartyA=F stackIdx=1 SDPInBeginMedia=F isJoined=F videoP=T EO=0 renegMedi=F
57315705.001 |22:32:55.916 |AppInfo |DET-kirra-ipv6 AgenaInterfaceBase-(63534)::initializeMemberData mNegotiatedIPAddrType = 0
57315706.000 |22:32:55.916 |SdlSig |MXNewParentPid |waitReconnect |H245Interface(3,100,185,53370) |MediaExchange(3,100,138,127080) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:3,L:0,V:0,Z:0,D:0] parent pid:nodeId=3.PN=138.PI=127080.vPT=2 allow2833=F injectDigitstoMTP=F subscribetoMTP=F passthru2833=F
57315707.000 |22:32:55.916 |SdlSig |MXInterfaceRestart |waitReconnect |H245Interface(3,100,185,53370) |MediaExchange(3,100,138,127080) |3,100,13,119956.2^10.130.3.9^Port 49839 |[R:N-H:0,N:2,L:0,V:0,Z:0,D:0] pid=0.0.0 partySide=B mAudioRegionKbps=64 mVideoRegionKbps=384 mMulticastFlag=F silence=F network=7 sessionId=0 ipAddrType=0 ipv4=10.130.3.9 port=49839 My DTMF (config=1 support=1 2833recvPT=0) Peer DTMF (config=1 support=1 2833recvPT=0) Media DTMF (subscribe=F passthru2833=F injectDigitstoMTP=F allow2833=F) cryptocapcount=0 RSVP nPortsOpened=0 AllowedCallType=0x1 aPT=2 vPT=2 e2eAR0 e2eVR0 capE2E=F dscpRSVPFail=F mtpChanged=F allowOneWay=F my connType=3 peer connType=3 mouth mode=0 mouth party=0 ear mode=0 ear party=0 mIsSRTP=T farEndXferM=0 farEndMediaReq=0 joined=F NormalReconn=T callIpAddrMode(my=0 peer=0 far=0) moh=F mediaReq=0 mPrecLvl=5 allowVideo=F SIPDevMediaInfo: [ vidPref=T isPartyA=F stackIdx=1 SDPInBeginMedia=F isJoined=F videoP=T EO=0 renegMedi=F
57315707.001 |22:32:55.916 |AppInfo |DET-H245Interface-(53370)::mapRestartSignal, mDTMFProfile oobUserInputCapability=1Could be a codec issue. Verify what codec you are trying to use when the call fails.
Is this H323 or MGCP, what is the PSTN connectivity, SIP or PRI or FX0?
Might be helpful to post the router config. -
Transferring a site from GoDaddy
I have a client who's transferring her site from GoDaddy to another hosting provider. Her GD acct. doesn't expire for a month or so. If she can change her DNS to the new hosting service can she put a redirect on her GD page to the new site? Or does GD immediately deactivate her acct? Or is there a better way to make this transition without the site being down for a couple days? Is there an issue with two DNS's for the same site? I don't know how it works.
Well I certainly agree with you about the other two companies but Godaddy is a different situation than either of the other two companines.
The only beef I have with Adobe is their support. We bought new DW & PS cs4's and they wouldn't load on our XP machines. I called Adobe support and they admitted there was a problem with the two programs loading on XP machines. They then told me nothing we can do about it so check our the forums and then hung the phone up on me.
Microsoft has always given us awesome support. We bought copies of Expression Web 4, didn't do the up grades bought new disc.It loaded fine on all our machines including the XP machines but one and it wouldn't activate. They worked with us on the phone for several days everyday until they had the problem resolved and then asked us to help them test a beta version of some software.
Godaddy is a registrar and a hosting company not a software company like Adobe and Microsoft. I remember when Godaddy was a decent company but undoubtly they've gotten greedy with some arrogance thrown in also. In the eyes of many 1000's of web designers and developers they are the scum of the earth as a registrar and a hosting company. Their countrol panel is junk, their support is junk, their hosting is junk and as a registrar they control your domain not you. They have lied to me like no other company in the industry, they have cheated and lied to some of our clients so we made the decision to dump them forever and join the ranks of the many 1000's of GD sucks designers and developers. We will not take a new client if they have their domain and or hosting with GD unless they change registrars and hosting companies. That's one of the main reasons GD is taking a large hit on this deal...1000's of mad designers and developers that's had to deal with their lies, arrogance and controlling ways.
I had a friend send this to me today and all we could do was laugh because we had been there and done that with GD more than one time.
http://www.ripoffreport.com/directory/godaddy-com.aspx
I condone no one if they're happy with GD and I certainly hope the GD lovers will continue be a satisfied customer of them.
We're certainly happy with the honest registrars and hosting companies we use for our clients now.
Happy New Year everyone, may you have a blessed and safe one. -
Clicking on PSTN calls only when placing call on hold/taking off hold
Hey all,
We have an ongoing issue that we thought was isolated to a call flow going to our UCCX, but after extensive troubleshooting, have found that it is occuring when a PSTN caller is being placed on hold by or being taken off of hold (most noticeable when taking off of hold) either via an IP Phone or another system (Unity Connection when releasing to switch is transferring outside of Unity Connection, UCCX and it's hold/unhold steps). If we mirror this call flow, only calling from an IP Phone to Another and being placed on hold, or including Unity Connection or UCCX in the call flow, this doesn't occur. We've been working with TAC and our ITSP for the past couple weeks but haven't really made much traction. Wireshark traces on the CUBE, so far, only have the audible noise on the ITSP facing interface of the CUBE, which is what I would expect given that it's not heard internally.
Call flow:
PSTN Caller -> ITSP via SIP -> CUBE -> SIP -> CUCM -> IP Phone
CUCM Version 8.5.1.15900-4
CUBE Version 15.1(3)T2
MOH and CUBE have a G711 region relationship. Most IP Phones and systems have a G729 relationship, but as a test I put a phone in my G711 test region and the issue still occurs.
The traces show the call being setup with G729 (preference on my dial-peer), but the stream is all G711 (audio from my IP Phone and MOH). This was verified both in the trace and via the CUBE when doing a "show sip calls <my cell phone number>" that both legs were G711 and the issue was still occuring.
We regularly restart the IPVMS service as our MOH changes from time to time, but again this is only happening on PSTN calls, so my focus at this time has been the CUBE.
Has anyone run into this before? Perhaps it's a bug and I need to update my CUBE's firmware?Hi,
Did you enable “Play music on hold”?
If yes, you can try to disable it and then test again.
Please also check if there are any error message on FE Server when the issue happen.
Please also update to the latest version for Lync Server 2013 and have a test.
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support -
Hi all,
I am new to Application Servers, SIP servers PSTN gateway in general (so sorry if I make some nonsense question) and I am using OCMS in a standalone installation .
I managed to make IP to IP calls between two instance of Oracle Communicator but as my company owns a gateway we use to make PSTN calls from voip clients I was wondering if there is a way for OCMS to use this gateway.
thanks,
RobertoHi Roberto,
Depending on how your SIP-PSTN gateway is configured, you can most likely dial the sip address containing the phone number and the address of your PSTN gateway directly from Communicator: example: sip:18885556666@<ip-address-of-pstn-gateway.com>
If your company has an ENUM server, the Proxy/Registrar will automatically make a rule lookup there if an all digit phone number is passed from the Communicator. The ENUM server can map a phone number to a sip address (such as the one exposed by the PSTN gateway) using NAPTR records.
Good luck,
~Adam -
What is difference between Site template and web template
What is difference between Site template and web template
Both are almost same, are you referring to Site Definitions vs web template?, if so, refer to the following post
http://blogs.msdn.com/b/vesku/archive/2011/07/22/site-definitions-vs-web-templates.aspx.
--Cheers -
I have two IPhones and one IPad all with the same Apple ID. I can call facetime between the two IPhones, but I cannot call the IPAD from either of the IPhones and when I call my IPhone from the IPad it rings my wife's IPhone after briefly calling mine.
Each device needs a separate address. Use an email address (gmail.com) on the iPad.
Using FaceTime http://support.apple.com/kb/ht4319
Troubleshooting FaceTime http://support.apple.com/kb/TS3367
The Complete Guide to FaceTime + iMessage: Setup, Use, and Troubleshooting
http://tinyurl.com/a7odey8
Troubleshooting FaceTime and iMessage activation
http://support.apple.com/kb/TS4268
Using FaceTime and iMessage behind a firewall
http://support.apple.com/kb/HT4245
iOS: About Messages
http://support.apple.com/kb/HT3529
Set up iMessage
http://www.apple.com/ca/ios/messages/
Troubleshooting Messages
http://support.apple.com/kb/TS2755
Setting Up Multiple iOS Devices for iMessage and Facetime
http://macmost.com/setting-up-multiple-ios-devices-for-messages-and-facetime.htm l
FaceTime and iMessage not accepting Apple ID password
http://www.ilounge.com/index.php/articles/comments/facetime-and-imessage-not-acc epting-apple-id-password/
Unable to use FaceTime and iMessage with my apple ID
https://discussions.apple.com/thread/4649373?tstart=90
For non-Apple devices, check out the TextFree app https://itunes.apple.com/us/app/text-free-textfree-sms-real/id399355755?mt=8
How to Send SMS from iPad
http://www.iskysoft.com/apple-ipad/send-sms-from-ipad.html
Cheers, Tom -
Unable to pass traffic between sites
I've read through dozens of posts and so far have had no luck getting any of the suggestions to work - combined with many of these posts being multiple years old...so I'm going to try posting something current and see if I get anywhere.
Scenario:
Site A - Cisco ASA 5510 running 8.4(4)1 with two interface connections to a Cisco ME 6500 (which I do not manage), one for internet and one for a MPLS connection.
Site B – connecting to an unknown switch which is connected to the MPLS network.
Site C – Cisco ASA 5505 running 7.2(3) with one connection to an unknown switch (which I do not manage) for internet access.
Site A to Site B traffic flows between the two without issue.
Site A to Site C is a site-to-site VPN connection. Traffic flows between the two without issue.
The main issue I’m having is that Site B cannot talk to Site C and vice versa. Also my client VPN connections to Site A cannot get to Site B or Site C.
My first question is; is this even possible? (I sure expected it to be). And if so, what the heck am I doing wrong???
I’ve included a config from Site A which is where I’m guessing the problem is. Any insight is appreciated."I'm not following what you mean by that."
Your Site "A" and "B" connected through MPLS cloud and they are not connected through vpn-connection, right? I assume that your site "B" cannot communicate to site "C", therefore you must permit site-B's subnet traffic transit between site "A" and site "C" i.e. Site-B should have access to "C", right ?
"I may be misunderstanding, but isn't that what this is: "route MPLS 10.17.0.0 255.255.0.0 10.17.250.2 1"."
Great 10.17.0.0/16 route meant for site "B", that is fine, you wouldn't need an additional one.
"You completely lost me there :)"
I presume that your Site "B" and "C" does not have direct MPLS connection, therefore Site "A" becomes a transit path for site "B" and "C". You allow site-B's transit through the vpn-tunnel between site "A" and "C". Your site "C" assumes that subnet belong to site "B" is directly connected at site "A" but in reality it connects via a MPLS cloud and one last thing is that a route needed at site-B to push site-C's traffic to Site "A", a static route would do that.
As you would permit site-B's traffic to pass through vpn-tunnel site "A" and "C", in other words your "A" become a hub for traffic flowing between site "B" and "C".
"Should the route be applied to the inside or the outside interface?"
Outside. Your tunnel terminated on the outside interface, right? If so then it must point to outside's default-gateway address.
object network SiteB-network
subnet 10.17.2.0 255.255.255.0
this would allow you to access site-c subnet when you are remote-in to Site-A.
nat (outside,outside) source static VPN-pool VPN-pool destination static SiteC-network SiteC-network
this is to allow Site-B to access site-C subnet via the tunnel between site A and C.
nat (MPLS,outside) source static SiteB-network SiteB-network destination static SiteC-network SiteC-network
object network inside-network
subnet 192.168.1.0 255.255.255.0
nat (inside,outside) source static inside-network inside-network destination static SiteC-network SiteC-network
access-list outside_cryptomap extended permit ip object inside-network object SiteC-network
this is allow Site-B to access site-C subnet via the tunnel between site A and C.
access-list outside_cryptomap extended permit ip object SiteB-network object SiteC-network
Thanks
Rizwan Rafeek -
From monitoring report I can find complete peer to peer audio calls details but I would like to get a report of only PSTN calls. How can I get that report.
Hi,Askwizard,
Great you figure out it by yourself!
Yes,the audio call details provide different types of audio call details and you also can additiona call details report,check
http://blogs.technet.com/b/drrez/archive/2012/01/09/how-to-create-additional-lync-server-2010-call-details-reports-part-1.aspx for more details.
Regards,
Sharon
Sharon Shen
TechNet Community Support
Please remember to click “Mark as Answer” on the post that helps you, and to click “Unmark as Answer” if a marked post does not actually answer your question. -
Call transferring- Not transferring calls
I use call tranferring alot but my skype does not tranfer calls to my local mobile number.
Whenever skype is active, i get calls on skype. I started using the call transferring feature so that i can have my skype calls transferred on my local mobile numbers when internet connection is not available.
But my calls are not been tranferred now.
Why is this happening?Hi,
well, Tanner has a point, it must be solved processwise.
The company I used to work for had a customer with a large UCCE contact center and they asked for a report that would show this kind of suspicious activity. UCCE has actually a way to mark these calls, unfortunately, as far as I know, there is no such thing in UCCX. The rule was simple: you can be caught only once doing that.
G. -
How to i get my calls transferred to my voicemail when i am listening to my music on my iphone?
how can i get my calls transferred to my voicemail when i am listening to my meditation music on my iphone?
Put your phone in airplane mode, all calls will go straight to voicemail, yet you can still use the iPod function.
Maybe you are looking for
-
SEVERE ERROR: Starting system failed, rc=0
Hi Experts I am having problem in SRM 2.0 to SRM 5.5(NW 7) Upgrade. I have problem in startup of Shadow Instance. Here is log description. more DEVTRACE.LOG trc file: "dev_ms", trc level: 1, release: "700" [Thr 01] Wed Jan 9 17:56:
-
Why can't i use my i tunes gift card for apps on my mac
i redeemd my itunes gift card and in the app store it says i have the money available but when i go to download an app it keeps asking me for credit card info
-
Error when CRETING STOCK INITIAL UPLOAD USING MB1C
THE ERROR IS : The number range for document number assaignment has not been maintained in the year 2011 for the transaction event type WA. we are getting this error while creating stock initial upload using t.code MB1C. and another ERROR is : Number
-
Javax.print problem: data not of declared type
Hi all, I'm trying to print a simple text-only document with the below code but it keeps giving me an illegal argument exception. By the way, I've tried all the combinations of DocFlavor.*.* and SERVICE_FORMATTED.PRINTABLE and SERVICE_FORMATTED.PAGEA
-
I've had my mac for two-and-a-half years and never had a problem getting an application to quit. So, for anyone else who is having such a problem: Apple icon > Force Quit. This finally got the Bridge CS3 to go away.