ATA 186 Reproducing DTMF tones from RFC2833 events
I am trying to get my ATA 186 (running version SCCP 3.x under CCM 5.04a) to reproduce received rtp-events. This is supporting a third-party IVR system that is connected via analog ports.
The IOS gateway that is sending the events is configured for dtmf-relay rtp-nte, and I can see the RTP events being received at the ATA. However, all I hear at the ATA is nothing or clicks.
I have the audiomode setting on the ATA set to 0x00150015 as is recommended in the ATA administrators guide.
Does anyone have DTMF working through the ATA?
Hi Kevin,
have you debugged the ATA with prserv.exe?
I had some issues in the past with a PESQ tester, and setting the audiomode to 0x00120012 helped for me.
hth
Sascha
....this is the output from prserv on a good call
|->CallState[0]=4 L=1 R=4016c6c FG=0 r=0,0
FG NewState[0]=4, L=1, R=4016c6c
|->CallInfo[0]
|->SetRingerMode[0]:3
<-|StOffHk[0]
|->SetRingerMode[0]:1
|->CallState[0]=1 L=1 R=4016c6c FG=0 r=4016c6c,0
|->SetRingerMode[0]:1
[0]StopTone
|->CallState[0]=5 L=1 R=4016c6c FG=0 r=4016c6c,0
|->CallInfo[0]
[0]StopTone
|->OpenRxChn[0]
[0]StopTone
<-|OpenRxChnAck[0]
Save Media[0]
|->StartMediaTx[0]:pt=0 ps=20 ss=0 ip=a63801f:19422
[0]MPT mode 0
[2]MPT mode 0
Enable LEC adapt [0]=1
[0]Enable encoder 0
[0]DPKT 1st: 3680 3520, pt 0
15948:00;2,0,0,0,
[0]DTMF 1 , insum:323505
[0]StopTone
[0]DTMF 2 , insum:285988
[0]DTMF 3 , insum:289227
[0]DTMF 4 , insum:287021
[0]DTMF 5 , insum:284108
[0]DTMF 6 , insum:286979
[0]DTMF 7 , insum:284993
[0]DTMF 8 , insum:302792
[0]DTMF 9 , insum:293475
[0]DTMF 0 , insum:293768
[0]DTMF * , insum:279360
[0]DTMF # , insum:301230
<-|EndCall[0]
<-|SK[0]:9, L=1, FGR=4016c6c
[0]StopTone
->CloseRxChn[0]
|->StopMediaTx[0]
|->CallState[0]=2 L=1 R=4016c6c FG=0 r=4016c6c,0
IDLE[0]
|->SetRingerMode[0]:1
15948:30;2,0,0,0,
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DTMF tones from CUCUM 9 thru H323 GW out SIP trunk not working
This is the setup. Currently in lab environment for a client, but needs to go into production
IP Phone -> CUCM 9 -> H323 GW -> SIP Trunk -> Proprietary device -> Analog phone
Calls complete both ways with no issues. Proprietary devices only uses G711ulaw, so I have configured a xcoder on the H323 GW to transcode to G729 across the WAN link (between the CUCM cluster and the H323 GW).
Pressing keys/sending DTMF tones from the IP phone are not heard in the analog phone
Running a debug voice ccpai inout at the H323 gateway shows me that the DTMF tones are being received the GW and are being sent along. See below:
Seaport#
Seaport#
Seaport#! Pressing digit "9" on VoIP phone
Seaport#
Seaport#
Seaport#
Seaport#
*Nov 5 15:41:57.637: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_begin:
Consume mask is not set. Relaying Digit 9 to dstCallId 0x49E
*Nov 5 15:41:57.637: //1181/00B99D4F0500/CCAPI/cc_relay_digit_begin_for_3way_conference:
Check DTMF relay digit begin for 3way conf
*Nov 5 15:41:57.713: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_end:
Consume mask is not set. Relaying Digit 9 to dstCallId 0x49E
*Nov 5 15:41:57.713: //1181/00B99D4F0500/CCAPI/cc_relay_digit_end_for_3way_conference:
Check DTMF relay digit end for 3way conf
Seaport#
Seaport#! Pressing digit "9" on VoIP phone " on VoIP phone 5" on VoIP phone
Seaport#
Seaport#
Seaport#
*Nov 5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_begin:
Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
*Nov 5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_relay_digit_begin_for_3way_conference:
Check DTMF relay digit begin for 3way conf
*Nov 5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_end:
Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
*Nov 5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_relay_digit_end_for_3way_conference:
Check DTMF relay digit end for 3way conf
Seaport#
Seaport#! Pressing digit " 5" on VoIP phone
Seaport#
Seaport#
Seaport#
*Nov 5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_begin:
Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
*Nov 5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_relay_digit_begin_for_3way_conference:
Check DTMF relay digit begin for 3way conf
*Nov 5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_end:
Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
*Nov 5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_relay_digit_end_for_3way_conference:
Check DTMF relay digit end for 3way conf
Seaport#
Seaport#
However, debug ccsip does not give me any indications that the DTMF tone is being sent out the SIP trunk. Debug ccsip all attached.
Relevant portions of the H323 configuration are below
voice service voip
no ip address trusted authenticate
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g729br8
interface Loopback0
ip address 172.16.88.254 255.255.255.255
ip pim sparse-dense-mode
h323-gateway voip interface
h323-gateway voip bind srcaddr 172.16.88.254
interface GigabitEthernet0/1
ip address 192.168.200.254 255.255.255.0
duplex auto
speed auto
interface Loopback0
ip address 172.16.88.254 255.255.255.255
ip pim sparse-dense-mode
h323-gateway voip interface
h323-gateway voip bind srcaddr 172.16.88.254
interface GigabitEthernet0/1 <- interface to proprietary device
ip address 192.168.200.254 255.255.255.0
duplex auto
speed auto
interface GigabitEthernet0/2 <-interface to Local LAN supporting IP Phones
ip address 10.10.10.254 255.255.255.0
duplex auto
speed auto
sccp local GigabitEthernet0/2
sccp ccm 10.10.10.254 identifier 1 priority 1 version 3.1
sccp ccm group 1
bind interface GigabitEthernet0/2
associate ccm 1 priority 1
associate profile 10 register xcoder_1
dspfarm profile 10 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 10
associate application SCCP
dial-peer voice 2 voip
description Default Incoming Dial Peer
incoming called-number .
voice-class codec 1
dtmf-relay h245-alphanumeric h245-signal rtp-nte
dial-peer voice 6 voip
destination-pattern 90052.. <- DN of analog phone
session protocol sipv2
session target ipv4:192.168.200.1 <- IP of proprietary device
codec g711ulaw
no vad
sip-ua
registrar ipv4:172.16.88.254 expires 3600
no transport tcp
telephony-service
sdspfarm units 4
sdspfarm transcode sessions 2
sdspfarm tag 1 xcoder_1
I also ran the debug voip rtp session named-event all but nothing was displayed when I pressed the digits on the IP Phone.
JeffPlease configure "dtmf-relay rtp-nte" command under SIP dial-peers.
Jorge Armijo
Please remember to rate helpful responses and identify helpful or correct answers. -
Cisco Unity 4.x not recognising DTMF tones from phones
I have cisco call manage 4.x and Unity 4.x . I have set up IVR but the IVR does not accept DTMF tones. Any suggestions anyone
Go to the Unity tools and you'll find the port status monitor there, open up all ports you have for inbound and then watch the output, if Unity is receiving the DTMFs you'll see the messages and the DTMF it received.
If you don't see any DTMFs messages then most lilely DTMFs are never reaching Unity and the problem is somewhere else.
HTH
java
If this helps, please rate
www.cisco.com/go/pdihelpdesk -
Having trouble calling my work voicemail from the iphone - it doesnt recognize the dtmf tones
i am having trouble calling my work voicemail frome the iphone - it does not recognize the DTMF tones from the dialpad. we have justgone to a new IP voice system at work
we use a remote dialling code to divert our office landlines to mobiles, we have had awful trouble with the iphones in doing this with the number stored as a contact, We also have trouble dialling the number direct from the keypad it still comes up with error and says the number has not been recognised.
I have managed to sort out how to remedy this for us on this end and i will give an example below, the 'p' represents a pause
(numbers changed for security)
original phonebook contact stored, please notice the pauses 'p' only the first single pause
01223495684p*65*4530*01223456456*07555456456#
the new number now stored in the phonebook with the extra pauses 'p'
01223495684pp*65p*4530p*01223456456p*07555456456# as you can see we dial a landline number and then give a pause for the next tone to cut in, then we enter the overall divert code, after this we enter the pin code followed by the landline number to divert and finally the mobile number that will be taking the calls and finishing off with hash.
all our other phones except the iphone will dial this number and work correctly with one pause at the beginning, I have now tried thios with two pauses at the beginning and one pause between each section and it now works fine,
This has cured our problem using the io5 iphones 4s. we could not dial these numbers with the iphones for nothing, all other phones work fine. When you have a bluetooth headset connected the problem is even worse.
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Hi folks,
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I tried updating to 15.1(4)M5 over the weekend, but while that seems to have helped the issue of the 2801 not sending some DTMF tones it's now occasionally sending double tones. Again, a packet capture shows the 2833 packets that I'd expect, but if I listen on the other end as I dial numbers I hear about 5% of the numbers dialed twice in short succession. -
!!cisco ATA 186, back to back.SOS
hi there, just enlighten me on cisco ATA 186 back to back config, my network is like:
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shukkyIf your ATA is running skinny right now then I think it is not possible (because no callmanager in the picture).
But if you load your ATA with a SIP image then in the SIP proxy filed, you can specify the IP address of the other SIP ATA and then you should be able to call from one ATA to another.
I haven't tried that with the ATA, but done with the 7940 phones and it worked.
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To reset the ATA for factory defaults
http://lists.digium.com/pipermail/asterisk-users/2003-August/017474.html
Hope this would help.
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I have a Linksys SPA 3102 as a Voip to PSTN gateway, this works great using other phones but not with my E61.
The problem seems to be that the DTMF tones are to long, when I send 1234, the SPA 3102 receives 111222333444!!!
Is there anyway to adjust the timing of E61 DTMF codes?I found another solution (thanks to hwittenb at Voxilla forum) which avoids me having to use DTMF to inititiate voip-to-pstn calls from Nokia phone to SPA3102. It's called a request uri and is described in the ATA admin guide section "How VoIP-To-PSTN Calls Work". It's a bit complicated to set up initially, but works in my setup....
Router set-up:
1. my SPA3102 is connected to Linksys router model WRT54GC set up with a dynamic dns (dyndns.org) address, so that I can always make direct IP calls to the SPA3102 even if my ISP changes my internet address
2. Router port forwarding is set to forward SIP UDP and RTP TCP ports to SPA3102. Check which are the correct port numbers to forward in SIP tab on the SPA3102
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2. SIP TAB: Note: I have not bothered with STUN server and the other settings for inbound IP dialling found in guides on the internet. This probably works because of Gizmo or Pfingo being used as the PSTN voip account.... e.g.: in the SIP tab, the NAT Support Parameters are factory defaults e.g.: all set to no
3. PSTN TAB: settings:
I use two accounts with the same service provider (either Pfingo or Gizmo) - one on the Nokia client, the other as the PSTN account in my SPA3102
SIP port: 5061
Dial Plan 1: (xx.)
VoIP-To-PSTN Gateway Enable: Yes
VoIP Caller Auth Method: None
One Stage Dialing: Yes
VoIP Caller ID Pattern: <user-ID of Nokia phone voip account>
VoIP Answer Delay: 0
Disconnect Tone: set correctly for my country
Note: Using this Caller ID pattern provides an authentication to prevent unknown caller IDs from placing PSTN calls through your SPA3102. You could leave this blank if you don't mind anyone using your voip-to-pstn. The caller ID pattern for a Gizmo account is not the number 1747... it is a word…
To dial a pstn number from Nokia client via voip-to-pstn through SPA3102:
1. store the phone number (pstn compatible format) you wish to dial into an internet telephone field, or SIP field, in a Nokia phonebook contact
<phone-number>@<dynamic-dns-address-for-router>:5061
2. with your Nokia voip account registered, then dial the contact sip uri address
When I dial, the voip call from Nokia client to SPA3102 is set up, the pstn FXO port is opened and the <phone-number> is dialled through the PSTN.... DTMF tones in call work fine with the established call. Good news!
Note: if PSTN line is already busy the call will not complete. Also my dial plan in the PSTN tab is actually a bit more complex than shown above to prevent IDD calls and premium rate numbers, again the call will not complete.
Message Edited by nicholso on 01-Sep-2009 09:12 AM
Message Edited by nicholso on 01-Sep-2009 09:15 AM -
I am trying to setup a cisco ATA 186 for a fax line with Cisco Call Manager 6.1.xxxx) I have a telephone hooked up to the Cisco ATA 186 and I see it register with the CCM. I have a telphone connected to port one and I get a dial tone, I can call into the phone from an outside line but it seems I cant dial to an outside number all I get is silence.
Could someone point me into a direction on troubleshooting this issue?
Thanks!Verify software versions first.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/compat/ccmcompmatr.html#wp50035
After that, you would really need to look at traces to find out why the audio isn't getting connected. Do phones on the same subnet have audio when you call in? Does this only happen on outside calls? If it only happens on outside calls, get a sniffer to see if you see the RTP traffic coming to and from the devices.
-Steven -
Issue on Cisco ATA 186 used for Fax
Hello everyone;
I have a cisco ATA 186 connected to an analog Fax on one side, and the other side is connected to the network containing a CUCM 8.6.
The problem is that when i try to send a fax throught it , half of the paper is transmitted or an usual occupation tone comes; while vocal communications are performed normaly.
Any idea of what could be the problem.
NB: the fax emission is between two distant sites relied by optical fiber.
Regards.Hello Frank;
Thank for the reply.
Here below more informations about the issue:
- It does work sometimes, but it did never worked properly.
- the problem is there for both sending and receiving.
- for the PSTN transport, i have no idea.
- I tried to change the ATA by another, still the same problem.
- ATAs work properly for voice communication.
- when i call this fax number internally the signal comes along.
- the problem is when faxing internally, it isn't used for external faxing.
- the call flow for internal faxing is like this:
Analog fax (in site A) > ATA (in site A) > access switch (in site A) > core switch (in site A) > cucm (in site A) > Core switch (in site A) > router (in site A) > optical fiber > router (in site B) > core switch (in site B) > cucm (in site B) > core switch (in site B) > voice gateway 248 (in site B) > Analog fax (in site B).
the problem occures when sending internal fax between these two sites A and B, when calling site B from site A, the ringing tone comes aloso the fax signal, but when i try the send the paper the stange tone come along and only half of it pass and it get to the site B as a blank paper or a half blank paper.
for any more explanation, don't hesitate to ask me.
thanks again for your time.
Regards. -
How can I tell what digits are being passed from a phone plugged into an ATA 186?
We have an ATA 186 with a credit card machine plugged into it that is failing to place calls. When I monitor the call with a butt set I hear dial tone, then 1 digit being dialed and then an error from callmanager basically telling me to check the number. The call never gets past that first digit. If I use a butt set to place a call everything works fine. I'm concerned the digit being passed is not the access code of 8, and I want to verify that. I have also included the 9001.log from the prsrev.exe I just don't know how to read it.Hi,
It look to me like the ATA isn't configured properly in CCM?
There's a lot of messages in the log stating "RegRej[1]:Error: DB Config", which I think means the device isn't registering properly.
Have you checked the ports are configured properly?
Here's a troubleshooting guide which may also help:
http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_configuration_example09186a0080094adf.shtml
Please rate if useful... -
What debug for seeing DTMF tones via CUBE -SIP?
On a 3845 running 12.4 20, the DTMF tones (for menu options) are not getting passed from the PSTN into the network after the call is established. The router is using CUBE software and passes the call to the PSTN via SIP trunking/SIP server. The call is established at G729. I'm using debug voip ccapi inout and debug sip messages, but I don't see the digits being passed.
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2. At G729, are the tones compressed, or are they out of band?HI aokanlawon ,
I finaly managed to get the debuf output from the "debug ccsip messages" and "debug voice ccapi inout" command.
Sorry for the long debug log, I don't have chance anywhere to upload the file. Also, I'm sending at the end, an output from the command "debug voip rtp session named-event"
Can You explain to me what's going on with the type of DTMF beeing transmitted?
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.02.19 17:18:31 =~=~=~=~=~=~=~=~=~=~=~=
Feb 19 17:21:03.254: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.115.5:5060;branch=z9hG4bK4cce15b002c71
From: "Caller" ;tag=659027~d01f1c4f-fe32-4ce8-bb98-06bd76f12320-33263464
To:
Date: Wed, 19 Feb 2014 16:21:03 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Cisco-Guid: 3528476416-0000065536-0000002046-0091423242
Session-Expires: 1800
P-Asserted-Identity: "Caller"
Remote-Party-ID: "Caller" ;party=calling;screen=yes;privacy=off
Contact:
Max-Forwards: 69
Content-Length: 0
Feb 19 17:21:03.258: //-1/D25047000000/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
cisco-username=+38923167XXX
----- ccCallInfo IE subfields -----
cisco-ani=sip:[email protected]
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=sip:[email protected]:5060
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
Feb 19 17:21:03.262: //-1/D25047000000/CCAPI/cc_api_call_setup_ind_common:
Interface=0x31340418, Call Info(
Calling Number=sip:[email protected],(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=sip:[email protected]:5060(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
Incoming Dial-peer=10, Progress Indication=NULL(0), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=176022
Feb 19 17:21:03.262: //-1/D25047000000/CCAPI/ccCheckClipClir:
In: Calling Number=sip:[email protected](TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
Feb 19 17:21:03.262: //-1/D25047000000/CCAPI/ccCheckClipClir:
Out: Calling Number=sip:[email protected](TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
Feb 19 17:21:03.262: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Feb 19 17:21:03.262: :cc_get_feature_vsa malloc success
Feb 19 17:21:03.262: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Feb 19 17:21:03.262: cc_get_feature_vsa count is 1
Feb 19 17:21:03.262: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Feb 19 17:21:03.262: :FEATURE_VSA attributes are: feature_name:0,feature_time:833881656,feature_id:160387
Feb 19 17:21:03.262: //176022/D25047000000/CCAPI/cc_api_call_setup_ind_common:
Set Up Event Sent;
Call Info(Calling Number=(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=(TON=Unknown, NPI=Unknown))
Feb 19 17:21:03.262: //176022/D25047000000/CCAPI/cc_process_call_setup_ind:
Event=0x31845110
Feb 19 17:21:03.262: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
Try with the demoted called number 0038923248XXX
Feb 19 17:21:03.262: //176022/D25047000000/CCAPI/ccCallSetContext:
Context=0x2B07195C
Feb 19 17:21:03.262: //176022/D25047000000/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 176022 with tag 10 to app "_ManagedAppProcess_Default"
Feb 19 17:21:03.262: //176022/D25047000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.115.5:5060;branch=z9hG4bK4cce15b002c71
From: "Caller" ;tag=659027~d01f1c4f-fe32-4ce8-bb98-06bd76f12320-33263464
To:
Date: Wed, 19 Feb 2014 16:21:03 gmt
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Feb 19 17:21:03.262: //176022/D25047000000/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
Feb 19 17:21:03.262: //-1/xxxxxxxxxxxx/CCAPI/ccGetMemPoolFromContainer:
mempool not found from usrContainer(31823634)
Feb 19 17:21:03.262: //-1/xxxxxxxxxxxx/CCAPI/ccCreateMemPoolInContainer:
Mempool(2AFD4B64) created in usrContainer(31823634)
Feb 19 17:21:03.266: //176022/D25047000000/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=TRUE, Mode=0,
Outgoing Dial-peer=20, Params=0x2B06551C, Progress Indication=NULL(0)
Feb 19 17:21:03.266: //176022/D25047000000/CCAPI/ccCheckClipClir:
In: Calling Number=sip:[email protected](TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
Feb 19 17:21:03.266: //176022/D25047000000/CCAPI/ccCheckClipClir:
Out: Calling Number=sip:[email protected](TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
Feb 19 17:21:03.266: //176022/D25047000000/CCAPI/ccCallSetupRequest:
Destination Pattern=.T, Called Number=0038923248XXX, Digit Strip=FALSE
Feb 19 17:21:03.266: //176022/D25047000000/CCAPI/ccCallSetupRequest:
Calling Number=sip:[email protected](TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=0038923248XXX(TON=Unknown, NPI=Unknown),
Redirect Number=, Display Info=Caller
Account Number=+38923167XXX, Final Destination Flag=TRUE,
Guid=D2504700-0001-0000-0000-07FE0573020A, Outgoing Dial-peer=20
Feb 19 17:21:03.266: //176022/D25047000000/CCAPI/cc_api_display_ie_subfields:
ccCallSetupRequest:
cisco-username=+38923167XXX
----- ccCallInfo IE subfields -----
cisco-ani=sip:[email protected]
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=0038923248XXX
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
Feb 19 17:21:03.266: //176022/D25047000000/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x31340418, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=sip:[email protected],(Calling Name=Caller)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=0038923248XXX(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=20, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
Feb 19 17:21:03.266: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Feb 19 17:21:03.266: :cc_get_feature_vsa malloc success
Feb 19 17:21:03.266: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Feb 19 17:21:03.266: cc_get_feature_vsa count is 2
Feb 19 17:21:03.266: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Feb 19 17:21:03.266: :FEATURE_VSA attributes are: feature_name:0,feature_time:833881880,feature_id:160388
Feb 19 17:21:03.266: //176023/D25047000000/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
Feb 19 17:21:03.266: //176023/D25047000000/CCAPI/ccCallSetContext:
Context=0x2B0654CC
Feb 19 17:21:03.266: //176022/D25047000000/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=20
Feb 19 17:21:03.266: //176023/D25047000000/CCAPI/cc_api_call_proceeding:
Interface=0x31340418, Progress Indication=NULL(0)
Feb 19 17:21:03.270: //176023/D25047000000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.126.98.37:5060;branch=z9hG4bK31F165A
Remote-Party-ID: "Caller" ;party=calling;screen=yes;privacy=off
From: sip:[email protected];tag=B8DD5C04-CE
To:
Date: Wed, 19 Feb 2014 16:21:03 gmt
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3528476416-0000065536-0000002046-0091423242
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1392826863
Contact:
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 68
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 331
v=0
o=CiscoSystemsSIP-GW-UserAgent 9057 1587 IN IP4 10.126.98.37
s=SIP Call
c=IN IP4 10.126.98.37
t=0 0
m=audio 19826 RTP/AVP 0 8 18 101 19
c=IN IP4 10.126.98.37
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
Feb 19 17:21:03.286: //176023/D25047000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.126.98.37:5060;branch=z9hG4bK31F165A
From: sip:[email protected];tag=B8DD5C04-CE
To:
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1392826863
Feb 19 17:21:03.362: //176023/D25047000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.126.98.37:5060;branch=z9hG4bK31F165A
From: sip:[email protected];tag=B8DD5C04-CE
To: ;tag=SDraqe898-2700717777
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1392826863
Content-Length: 0
Contact:
Server: Patton SN4940 1E30V 00A0BA097588 R6.T 2013-04-04_RFE2204 H323 RBS SIP M5T SIP Stack/4.1.12.18
Feb 19 17:21:03.362: //176023/D25047000000/CCAPI/cc_api_event_indication:
Event=97, Call Id=176023
Feb 19 17:21:03.362: //176023/D25047000000/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
Feb 19 17:21:03.362: //176023/D25047000000/CCAPI/cc_api_call_cut_progress:
Interface=0x31340418, Progress Indication=NULL(0), Signal Indication=NOT PRESENT(255),
Cause Value=0
Feb 19 17:21:03.362: //176023/D25047000000/CCAPI/cc_api_call_cut_progress:
Call Entry(Responsed=TRUE)
Feb 19 17:21:03.362: //176022/D25047000000/CCAPI/ccCallCutProgress:
Progress Indication=NULL(0), Signal Indication=NOT PRESENT(255), Cause Value=0
Voice Call Send Alert=FALSE, Call Entry(Alert Sent=FALSE)
Feb 19 17:21:03.362: //176022/D25047000000/CCAPI/ccCallCutProgress:
Call Entry(Responsed=TRUE)
Feb 19 17:21:03.362: //176022/D25047000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.2.115.5:5060;branch=z9hG4bK4cce15b002c71
From: "Caller" ;tag=659027~d01f1c4f-fe32-4ce8-bb98-06bd76f12320-33263464
To: ;tag=B8DD5C64-7FF
Date: Wed, 19 Feb 2014 16:21:03 gmt
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: ;party=called;screen=no;privacy=off
Contact:
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Feb 19 17:21:03.506: //176023/D25047000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.126.98.37:5060;branch=z9hG4bK31F165A
From: sip:[email protected];tag=B8DD5C04-CE
To: ;tag=SDraqe898-2700717777
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1392826863
Content-Length: 0
Contact:
Server: Patton SN4940 1E30V 00A0BA097588 R6.T 2013-04-04_RFE2204 H323 RBS SIP M5T SIP Stack/4.1.12.18
Feb 19 17:21:03.506: //176023/D25047000000/CCAPI/cc_api_call_alert:
Interface=0x31340418, Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)
Feb 19 17:21:03.506: //176023/D25047000000/CCAPI/cc_api_call_alert:
Call Entry(Retry Count=0, Responsed=TRUE)
Feb 19 17:21:03.506: //176022/D25047000000/CCAPI/ccCallAlert:
Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)
Feb 19 17:21:03.506: //176022/D25047000000/CCAPI/ccCallAlert:
Call Entry(Responsed=TRUE, Alert Sent=TRUE)
Feb 19 17:21:03.506: //176022/D25047000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.2.115.5:5060;branch=z9hG4bK4cce15b002c71
From: "Caller" ;tag=659027~d01f1c4f-fe32-4ce8-bb98-06bd76f12320-33263464
To: ;tag=B8DD5C64-7FF
Date: Wed, 19 Feb 2014 16:21:03 gmt
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: ;party=called;screen=no;privacy=off
Contact:
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Feb 19 17:21:03.730: //176023/D25047000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.126.98.37:5060;branch=z9hG4bK31F165A
From: sip:[email protected];tag=B8DD5C04-CE
To: ;tag=SDraqe898-2700717777
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1392826863
Content-Length: 274
Contact:
Content-Type: application/sdp
Supported: replaces
Server: Patton SN4940 1E30V 00A0BA097588 R6.T 2013-04-04_RFE2204 H323 RBS SIP M5T SIP Stack/4.1.12.18
v=0
o=MxSIP 0 18772 IN IP4 172.25.255.196
s=SIP Call
c=IN IP4 172.25.255.196
t=0 0
m=audio 20452 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=sendrecv
Feb 19 17:21:03.734: //176023/D25047000000/CCAPI/cc_api_caps_ind:
Destination Interface=0x0, Destination Call Id=-1, Source Call Id=176023,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=160, Signal Type=2)
Feb 19 17:21:03.734: //176023/D25047000000/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
Feb 19 17:21:03.734: //176022/D25047000000/CCAPI/cc_api_caps_ack:
Destination Interface=0x0, Destination Call Id=176023, Source Call Id=176022,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1348)
Feb 19 17:21:03.734: //176023/D25047000000/CCAPI/cc_api_event_indication:
Event=162, Call Id=176023
Feb 19 17:21:03.734: //176023/D25047000000/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
Feb 19 17:21:03.734: //176023/D25047000000/CCAPI/cc_api_call_connected:
Interface=0x31340418, Data Bitmask=0x1, Progress Indication=NULL(0),
Connection Handle=0
Feb 19 17:21:03.734: //176023/D25047000000/CCAPI/cc_api_call_connected:
Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0)
Feb 19 17:21:03.734: //176022/D25047000000/CCAPI/ccConferenceCreate:
(confID=0x2AFC22A8, callID1=0x2AF96, gcid=A945E877-98B811E3-897AFE89-917343ED, tag=0x0)
Feb 19 17:21:03.734: //176023/D25047000000/CCAPI/ccConferenceCreate:
(confID=0x2AFC22A8, callID2=0x2AF97, gcid=A945E877-98B811E3-897AFE89-917343ED, tag=0x0)
Feb 19 17:21:03.734: //176022/D25047000000/CCAPI/ccConferenceCreate:
Conference Id=0x2AFC22A8, Call Id1=176022, Call Id2=176023, Tag=0x0
Feb 19 17:21:03.734: //176022/D25047000000/CCAPI/ccConferenceCreate:
Feb 19 17:21:03.734: ccConferenceCreate: ret1=0, codecMask1=1, bytes1=160, negot1=0, dtmf1=6
ret2=0, codecMask2=1, bytes2=160, negot2=1, dtmf2=6,
tx_dynamic_pt1=0, rx_dynamic_pt1=0, codec_mode1=0, params_bitmap1 =0
tx_dynamic_pt2=0, rx_dynamic_pt2=0, codec_mode2=0, params_bitmap2 =0
Feb 19 17:21:03.734: //176022/D25047000000/CCAPI/ccConferenceCreate:
delay media to slow start case, codec negotation is not done
Feb 19 17:21:03.734: //176022/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
Feb 19 17:21:03.734: cc_api_get_xcode_stream : 4682
Feb 19 17:21:03.734: //176022/D25047000000/CCAPI/cc_api_bridge_done:
Conference Id=0xF2EA, Source Interface=0x31340418, Source Call Id=176022,
Destination Call Id=176023, Disposition=0x0, Tag=0x0
Feb 19 17:21:03.734: //176023/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
Feb 19 17:21:03.734: cc_api_get_xcode_stream : 4682
Feb 19 17:21:03.734: //176023/D25047000000/CCAPI/cc_api_bridge_done:
Conference Id=0xF2EA, Source Interface=0x31340418, Source Call Id=176023,
Destination Call Id=176022, Disposition=0x0, Tag=0x0
Feb 19 17:21:03.734: //176022/D25047000000/CCAPI/cc_generic_bridge_done:
Conference Id=0xF2EA, Source Interface=0x31340418, Source Call Id=176023,
Destination Call Id=176022, Disposition=0x0, Tag=0x0
Feb 19 17:21:03.734: //176022/D25047000000/CCAPI/ccConferenceCreate:
Call Entry(Conference Id=0xF2EA, Destination Call Id=176023)
Feb 19 17:21:03.734: //176023/D25047000000/CCAPI/ccConferenceCreate:
Call Entry(Conference Id=0xF2EA, Destination Call Id=176022)
Feb 19 17:21:03.734: //176022/D25047000000/CCAPI/cc_process_notify_bridge_done:
Conference Id=0xF2EA, Call Id1=176022, Call Id2=176023
Feb 19 17:21:03.738: //176023/D25047000000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.126.98.37:5060;branch=z9hG4bK31F171A4E
From: sip:[email protected];tag=B8DD5C04-CE
To: ;tag=SDraqe898-2700717777
Date: Wed, 19 Feb 2014 16:21:03 gmt
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Feb 19 17:21:03.738: //176022/D25047000000/CCAPI/ccCallConnect:
Progress Indication=NULL(0), Data Bitmask=0x1
Feb 19 17:21:03.738: //176022/D25047000000/CCAPI/ccCallConnect:
Call Entry(Connected=TRUE, Responsed=TRUE)
Feb 19 17:21:03.738: //176022/D25047000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.115.5:5060;branch=z9hG4bK4cce15b002c71
From: "Caller" ;tag=659027~d01f1c4f-fe32-4ce8-bb98-06bd76f12320-33263464
To: ;tag=B8DD5C64-7FF
Date: Wed, 19 Feb 2014 16:21:03 gmt
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: ;party=called;screen=no;privacy=off
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 268
v=0
o=CiscoSystemsSIP-GW-UserAgent 4524 2325 IN IP4 10.2.115.15
s=SIP Call
c=IN IP4 10.2.115.15
t=0 0
m=audio 28154 RTP/AVP 0 101 19
c=IN IP4 10.2.115.15
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Feb 19 17:21:03.750: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.115.5:5060;branch=z9hG4bK4cce29f57702
From: "Caller" ;tag=659027~d01f1c4f-fe32-4ce8-bb98-06bd76f12320-33263464
To: ;tag=B8DD5C64-7FF
Date: Wed, 19 Feb 2014 16:21:03 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 233
v=0
o=CiscoSystemsCCM-SIP 659027 1 IN IP4 10.2.115.5
s=SIP Call
c=IN IP4 10.2.52.17
b=TIAS:64000
b=AS:64
t=0 0
m=audio 16388 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Feb 19 17:21:03.754: //176022/D25047000000/CCAPI/cc_api_caps_ind:
Destination Interface=0x31340418, Destination Call Id=176023, Source Call Id=176022,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=160, Signal Type=2)
Feb 19 17:21:03.754: //176022/D25047000000/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
Feb 19 17:21:03.754: //176023/D25047000000/CCAPI/cc_api_caps_ack:
Destination Interface=0x31340418, Destination Call Id=176022, Source Call Id=176023,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1)
Feb 19 17:21:03.754: //176022/D25047000000/CCAPI/cc_api_event_indication:
Event=162, Call Id=176022
Feb 19 17:21:03.754: //176022/D25047000000/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
Feb 19 17:21:03.754: //176022/D25047000000/CCAPI/cc_api_caps_ind:
Destination Interface=0x31340418, Destination Call Id=176023, Source Call Id=176022,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=160, Signal Type=2)
Feb 19 17:21:03.754: //176022/D25047000000/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
Feb 19 17:21:03.754: //176023/D25047000000/CCAPI/cc_api_caps_ack:
Destination Interface=0x31340418, Destination Call Id=176022, Source Call Id=176023,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1)
Feb 19 17:21:14.410: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.115.5:5060;branch=z9hG4bK4cce43ae98389
From: "Caller" ;tag=659027~d01f1c4f-fe32-4ce8-bb98-06bd76f12320-33263464
To: ;tag=B8DD5C64-7FF
Date: Wed, 19 Feb 2014 16:21:03 GMT
Call-ID: [email protected]
User-Agent: Cisco-CUCM9.1
Max-Forwards: 70
P-Asserted-Identity: "Caller"
CSeq: 102 BYE
Reason: Q.850;cause=16
Content-Length: 0
Feb 19 17:21:14.410: //176023/D25047000000/CCAPI/ccGenerateToneInfo:
Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Sum Network, Params=0x0, Call Id=176023
Feb 19 17:21:14.410: //176022/D25047000000/CCAPI/cc_api_call_disconnected:
Cause Value=16, Interface=0x31340418, Call Id=176022
Feb 19 17:21:14.410: //176022/D25047000000/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=TRUE, Cause Value=16, Retry Count=0)
Feb 19 17:21:14.410: //176022/D25047000000/CCAPI/ccConferenceDestroy:
Conference Id=0xF2EA, Tag=0x0
Feb 19 17:21:14.410: //176022/D25047000000/CCAPI/cc_api_bridge_drop_done:
Conference Id=0xF2EA, Source Interface=0x31340418, Source Call Id=176022,
Destination Call Id=176023, Disposition=0x0, Tag=0x0
Feb 19 17:21:14.410: //176023/D25047000000/CCAPI/cc_api_bridge_drop_done:
Conference Id=0xF2EA, Source Interface=0x31340418, Source Call Id=176023,
Destination Call Id=176022, Disposition=0x0, Tag=0x0
Feb 19 17:21:14.410: //176022/D25047000000/CCAPI/cc_generic_bridge_done:
Conference Id=0xF2EA, Source Interface=0x31340418, Source Call Id=176023,
Destination Call Id=176022, Disposition=0x0, Tag=0x0
Feb 19 17:21:14.410: //176022/D25047000000/CCAPI/ccCallDisconnect:
Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=16)
Feb 19 17:21:14.414: //176022/D25047000000/CCAPI/ccCallDisconnect:
Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
Feb 19 17:21:14.414: //176023/D25047000000/CCAPI/ccCallDisconnect:
Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
Feb 19 17:21:14.414: //176023/D25047000000/CCAPI/ccCallDisconnect:
Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
Feb 19 17:21:14.414: //176022/D25047000000/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x31340418, Tag=0x0, Call Id=176022,
Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
Feb 19 17:21:14.414: //176022/D25047000000/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
Feb 19 17:21:14.414: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Feb 19 17:21:14.414: :cc_free_feature_vsa freeing 31B40630
Feb 19 17:21:14.414: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Feb 19 17:21:14.414: vsacount in free is 1
Feb 19 17:21:14.414: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.115.5:5060;branch=z9hG4bK4cce43ae98389
From: "Caller" ;tag=659027~d01f1c4f-fe32-4ce8-bb98-06bd76f12320-33263464
To: ;tag=B8DD5C64-7FF
Date: Wed, 19 Feb 2014 16:21:14 gmt
Call-ID: [email protected]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 BYE
Reason: Q.850;cause=16
Content-Length: 0
Feb 19 17:21:14.414: //176023/D25047000000/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.126.98.37:5060;branch=z9hG4bK31F187D0
From: sip:[email protected];tag=B8DD5C04-CE
To: ;tag=SDraqe898-2700717777
Date: Wed, 19 Feb 2014 16:21:03 gmt
Call-ID: [email protected]
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1392826874
CSeq: 102 BYE
Reason: Q.850;cause=16
Content-Length: 0
Feb 19 17:21:14.426: //176023/D25047000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.126.98.37:5060;branch=z9hG4bK31F187D0
From: sip:[email protected];tag=B8DD5C04-CE
To: ;tag=SDraqe898-2700717777
Call-ID: [email protected]
Timestamp: 1392826874
CSeq: 102 BYE
Server: Patton SN4940 1E30V 00A0BA097588 R6.T 2013-04-04_RFE2204 H323 RBS SIP M5T SIP Stack/4.1.12.18
Content-Length: 0
Feb 19 17:21:14.426: //176023/D25047000000/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x31340418, Tag=0x0, Call Id=176023,
Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
Feb 19 17:21:14.426: //176023/D25047000000/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
Feb 19 17:21:14.426: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Feb 19 17:21:14.426: :cc_free_feature_vsa freeing 31B40710
Feb 19 17:21:14.426: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Feb 19 17:21:14.426: vsacount in free is 0
Feb 19 17:21:14.426: //-1/xxxxxxxxxxxx/CCAPI/ccMemPoolTDFreeHelper:
data = 2B00079C
Feb 19 17:21:14.426: ccMemPoolTDFreeHelper:mem_mgr_mempool_free: mem_refcnt(2AFD4B64)=0 - mempool
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.02.19 17:18:31 =~=~=~=~=~=~=~=~=~=~=~=
Debug output from "debug voip rtp session named-event"
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.02.19 17:29:25 =~=~=~=~=~=~=~=~=~=~=~=
Feb 19 17:31:52.030: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0x79 timestamp 0x4BA0
Feb 19 17:31:52.030: << Pt:101 Evt:2 Pkt:0A 00 00
Feb 19 17:31:52.030: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0x79 timestamp 0x4BA0
Feb 19 17:31:52.030: Pt:101 Evt:2 Pkt:0A 00 00 >>
Feb 19 17:31:52.042: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0x7B timestamp 0x4BA0
Feb 19 17:31:52.042: << Pt:101 Evt:2 Pkt:0A 00 A0
Feb 19 17:31:52.042: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0x7B timestamp 0x4BA0
Feb 19 17:31:52.042: Pt:101 Evt:2 Pkt:0A 00 A0 >>
Feb 19 17:31:52.050: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0x7C timestamp 0x4BA0
Feb 19 17:31:52.050: << Pt:101 Evt:2 Pkt:0A 01 40
Feb 19 17:31:52.050: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0x7C timestamp 0x4BA0
Feb 19 17:31:52.050: Pt:101 Evt:2 Pkt:0A 01 40 >>
Feb 19 17:31:52.062: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0x7E timestamp 0x4BA0
Feb 19 17:31:52.062: << Pt:101 Evt:2 Pkt:0A 01 E0
Feb 19 17:31:52.062: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0x7E timestamp 0x4BA0
Feb 19 17:31:52.062: Pt:101 Evt:2 Pkt:0A 01 E0 >>
Feb 19 17:31:52.070: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0x7F timestamp 0x4BA0
Feb 19 17:31:52.070: << Pt:101 Evt:2 Pkt:0A 02 80
Feb 19 17:31:52.070: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0x7F timestamp 0x4BA0
Feb 19 17:31:52.070: Pt:101 Evt:2 Pkt:0A 02 80 >>
Feb 19 17:31:52.082: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0x81 timestamp 0x4BA0
Feb 19 17:31:52.082: << Pt:101 Evt:2 Pkt:0A 03 20
Feb 19 17:31:52.082: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0x81 timestamp 0x4BA0
Feb 19 17:31:52.082: Pt:101 Evt:2 Pkt:0A 03 20 >>
Feb 19 17:31:52.090: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0x82 timestamp 0x4BA0
Feb 19 17:31:52.090: << Pt:101 Evt:2 Pkt:0A 03 C0
Feb 19 17:31:52.090: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0x82 timestamp 0x4BA0
Feb 19 17:31:52.090: Pt:101 Evt:2 Pkt:0A 03 C0 >>
Feb 19 17:31:52.102: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0x84 timestamp 0x4BA0
Feb 19 17:31:52.102: << Pt:101 Evt:2 Pkt:0A 04 60
Feb 19 17:31:52.102: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0x84 timestamp 0x4BA0
Feb 19 17:31:52.102: Pt:101 Evt:2 Pkt:0A 04 60 >>
Feb 19 17:31:52.110: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0x85 timestamp 0x4BA0
Feb 19 17:31:52.110: << Pt:101 Evt:2 Pkt:0A 05 00
Feb 19 17:31:52.110: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0x85 timestamp 0x4BA0
Feb 19 17:31:52.110: Pt:101 Evt:2 Pkt:0A 05 00 >>
Feb 19 17:31:52.122: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0x87 timestamp 0x4BA0
Feb 19 17:31:52.122: << Pt:101 Evt:2 Pkt:0A 05 A0
Feb 19 17:31:52.122: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0x87 timestamp 0x4BA0
Feb 19 17:31:52.122: Pt:101 Evt:2 Pkt:0A 05 A0 >>
Feb 19 17:31:52.130: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0x88 timestamp 0x4BA0
Feb 19 17:31:52.130: << Pt:101 Evt:2 Pkt:0A 06 40
Feb 19 17:31:52.130: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0x88 timestamp 0x4BA0
Feb 19 17:31:52.130: Pt:101 Evt:2 Pkt:0A 06 40 >>
Feb 19 17:31:52.142: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0x8A timestamp 0x4BA0
Feb 19 17:31:52.142: << Pt:101 Evt:2 Pkt:0A 06 E0
Feb 19 17:31:52.142: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0x8A timestamp 0x4BA0
Feb 19 17:31:52.142: Pt:101 Evt:2 Pkt:0A 06 E0 >>
Feb 19 17:31:52.150: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0x8B timestamp 0x4BA0
Feb 19 17:31:52.150: << Pt:101 Evt:2 Pkt:0A 07 80
Feb 19 17:31:52.150: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0x8B timestamp 0x4BA0
Feb 19 17:31:52.150: Pt:101 Evt:2 Pkt:0A 07 80 >>
Feb 19 17:31:52.162: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0x8D timestamp 0x4BA0
Feb 19 17:31:52.162: << Pt:101 Evt:2 Pkt:0A 08 20
Feb 19 17:31:52.162: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0x8D timestamp 0x4BA0
Feb 19 17:31:52.162: Pt:101 Evt:2 Pkt:0A 08 20 >>
Feb 19 17:31:52.170: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0x8E timestamp 0x4BA0
Feb 19 17:31:52.170: << Pt:101 Evt:2 Pkt:0A 08 C0
Feb 19 17:31:52.170: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0x8E timestamp 0x4BA0
Feb 19 17:31:52.170: Pt:101 Evt:2 Pkt:0A 08 C0 >>
Feb 19 17:31:52.182: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0x90 timestamp 0x4BA0
Feb 19 17:31:52.182: << Pt:101 Evt:2 Pkt:0A 09 60
Feb 19 17:31:52.182: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0x90 timestamp 0x4BA0
Feb 19 17:31:52.182: Pt:101 Evt:2 Pkt:0A 09 60 >>
Feb 19 17:31:52.190: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0x91 timestamp 0x4BA0
Feb 19 17:31:52.190: << Pt:101 Evt:2 Pkt:0A 0A 00
Feb 19 17:31:52.190: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0x91 timestamp 0x4BA0
Feb 19 17:31:52.190: Pt:101 Evt:2 Pkt:0A 0A 00 >>
Feb 19 17:31:52.202: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0x93 timestamp 0x4BA0
Feb 19 17:31:52.202: << Pt:101 Evt:2 Pkt:8A 0A A0
Feb 19 17:31:52.202: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0x93 timestamp 0x4BA0
Feb 19 17:31:52.202: Pt:101 Evt:2 Pkt:8A 0A A0 >>
Feb 19 17:31:52.202: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0x94 timestamp 0x4BA0
Feb 19 17:31:52.202: << Pt:101 Evt:2 Pkt:8A 0A A0
Feb 19 17:31:52.202: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0x94 timestamp 0x4BA0
Feb 19 17:31:52.202: Pt:101 Evt:2 Pkt:8A 0A A0 >>
Feb 19 17:31:52.218: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0x95 timestamp 0x4BA0
Feb 19 17:31:52.218: << Pt:101 Evt:2 Pkt:8A 0A A0
Feb 19 17:31:52.218: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0x95 timestamp 0x4BA0
Feb 19 17:31:52.218: Pt:101 Evt:2 Pkt:8A 0A A0 >>
Feb 19 17:31:52.542: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xA6 timestamp 0x5B40
Feb 19 17:31:52.542: << Pt:101 Evt:3 Pkt:0A 00 00
Feb 19 17:31:52.542: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xA6 timestamp 0x5B40
Feb 19 17:31:52.542: Pt:101 Evt:3 Pkt:0A 00 00 >>
Feb 19 17:31:52.550: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xA7 timestamp 0x5B40
Feb 19 17:31:52.550: << Pt:101 Evt:3 Pkt:0A 00 A0
Feb 19 17:31:52.550: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xA7 timestamp 0x5B40
Feb 19 17:31:52.550: Pt:101 Evt:3 Pkt:0A 00 A0 >>
Feb 19 17:31:52.562: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xA9 timestamp 0x5B40
Feb 19 17:31:52.562: << Pt:101 Evt:3 Pkt:0A 01 40
Feb 19 17:31:52.562: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xA9 timestamp 0x5B40
Feb 19 17:31:52.562: Pt:101 Evt:3 Pkt:0A 01 40 >>
Feb 19 17:31:52.570: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xAA timestamp 0x5B40
Feb 19 17:31:52.570: << Pt:101 Evt:3 Pkt:0A 01 E0
Feb 19 17:31:52.570: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xAA timestamp 0x5B40
Feb 19 17:31:52.570: Pt:101 Evt:3 Pkt:0A 01 E0 >>
Feb 19 17:31:52.582: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xAC timestamp 0x5B40
Feb 19 17:31:52.582: << Pt:101 Evt:3 Pkt:0A 02 80
Feb 19 17:31:52.582: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xAC timestamp 0x5B40
Feb 19 17:31:52.582: Pt:101 Evt:3 Pkt:0A 02 80 >>
Feb 19 17:31:52.590: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xAD timestamp 0x5B40
Feb 19 17:31:52.590: << Pt:101 Evt:3 Pkt:0A 03 20
Feb 19 17:31:52.590: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xAD timestamp 0x5B40
Feb 19 17:31:52.590: Pt:101 Evt:3 Pkt:0A 03 20 >>
Feb 19 17:31:52.602: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xAF timestamp 0x5B40
Feb 19 17:31:52.602: << Pt:101 Evt:3 Pkt:0A 03 C0
Feb 19 17:31:52.602: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xAF timestamp 0x5B40
Feb 19 17:31:52.602: Pt:101 Evt:3 Pkt:0A 03 C0 >>
Feb 19 17:31:52.610: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xB0 timestamp 0x5B40
Feb 19 17:31:52.610: << Pt:101 Evt:3 Pkt:0A 04 60
Feb 19 17:31:52.610: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xB0 timestamp 0x5B40
Feb 19 17:31:52.610: Pt:101 Evt:3 Pkt:0A 04 60 >>
Feb 19 17:31:52.622: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xB2 timestamp 0x5B40
Feb 19 17:31:52.622: << Pt:101 Evt:3 Pkt:0A 05 00
Feb 19 17:31:52.622: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xB2 timestamp 0x5B40
Feb 19 17:31:52.622: Pt:101 Evt:3 Pkt:0A 05 00 >>
Feb 19 17:31:52.630: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xB3 timestamp 0x5B40
Feb 19 17:31:52.630: << Pt:101 Evt:3 Pkt:0A 05 A0
Feb 19 17:31:52.630: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xB3 timestamp 0x5B40
Feb 19 17:31:52.630: Pt:101 Evt:3 Pkt:0A 05 A0 >>
Feb 19 17:31:52.642: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xB5 timestamp 0x5B40
Feb 19 17:31:52.642: << Pt:101 Evt:3 Pkt:0A 06 40
Feb 19 17:31:52.642: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xB5 timestamp 0x5B40
Feb 19 17:31:52.642: Pt:101 Evt:3 Pkt:0A 06 40 >>
Feb 19 17:31:52.650: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xB6 timestamp 0x5B40
Feb 19 17:31:52.650: << Pt:101 Evt:3 Pkt:0A 06 E0
Feb 19 17:31:52.650: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xB6 timestamp 0x5B40
Feb 19 17:31:52.650: Pt:101 Evt:3 Pkt:0A 06 E0 >>
Feb 19 17:31:52.662: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xB8 timestamp 0x5B40
Feb 19 17:31:52.662: << Pt:101 Evt:3 Pkt:8A 07 80
Feb 19 17:31:52.662: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xB8 timestamp 0x5B40
Feb 19 17:31:52.662: Pt:101 Evt:3 Pkt:8A 07 80 >>
Feb 19 17:31:52.662: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xB9 timestamp 0x5B40
Feb 19 17:31:52.662: << Pt:101 Evt:3 Pkt:8A 07 80
Feb 19 17:31:52.662: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xB9 timestamp 0x5B40
Feb 19 17:31:52.662: Pt:101 Evt:3 Pkt:8A 07 80 >>
Feb 19 17:31:52.678: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xBA timestamp 0x5B40
Feb 19 17:31:52.678: << Pt:101 Evt:3 Pkt:8A 07 80
Feb 19 17:31:52.678: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xBA timestamp 0x5B40
Feb 19 17:31:52.678: Pt:101 Evt:3 Pkt:8A 07 80 >>
Feb 19 17:31:52.970: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xC9 timestamp 0x67C0
Feb 19 17:31:52.970: << Pt:101 Evt:5 Pkt:0A 00 00
Feb 19 17:31:52.970: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xC9 timestamp 0x67C0
Feb 19 17:31:52.970: Pt:101 Evt:5 Pkt:0A 00 00 >>
Feb 19 17:31:52.982: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xCA timestamp 0x67C0
Feb 19 17:31:52.982: << Pt:101 Evt:5 Pkt:0A 00 A0
Feb 19 17:31:52.982: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xCA timestamp 0x67C0
Feb 19 17:31:52.982: Pt:101 Evt:5 Pkt:0A 00 A0 >>
Feb 19 17:31:52.990: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xCC timestamp 0x67C0
Feb 19 17:31:52.990: << Pt:101 Evt:5 Pkt:0A 01 40
Feb 19 17:31:52.990: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xCC timestamp 0x67C0
Feb 19 17:31:52.990: Pt:101 Evt:5 Pkt:0A 01 40 >>
Feb 19 17:31:53.002: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xCD timestamp 0x67C0
Feb 19 17:31:53.002: << Pt:101 Evt:5 Pkt:0A 01 E0
Feb 19 17:31:53.002: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xCD timestamp 0x67C0
Feb 19 17:31:53.002: Pt:101 Evt:5 Pkt:0A 01 E0 >>
Feb 19 17:31:53.010: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xCF timestamp 0x67C0
Feb 19 17:31:53.010: << Pt:101 Evt:5 Pkt:0A 02 80
Feb 19 17:31:53.010: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xCF timestamp 0x67C0
Feb 19 17:31:53.010: Pt:101 Evt:5 Pkt:0A 02 80 >>
Feb 19 17:31:53.022: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xD0 timestamp 0x67C0
Feb 19 17:31:53.022: << Pt:101 Evt:5 Pkt:0A 03 20
Feb 19 17:31:53.022: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xD0 timestamp 0x67C0
Feb 19 17:31:53.022: Pt:101 Evt:5 Pkt:0A 03 20 >>
Feb 19 17:31:53.030: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xD2 timestamp 0x67C0
Feb 19 17:31:53.030: << Pt:101 Evt:5 Pkt:0A 03 C0
Feb 19 17:31:53.030: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xD2 timestamp 0x67C0
Feb 19 17:31:53.030: Pt:101 Evt:5 Pkt:0A 03 C0 >>
Feb 19 17:31:53.042: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xD3 timestamp 0x67C0
Feb 19 17:31:53.042: << Pt:101 Evt:5 Pkt:0A 04 60
Feb 19 17:31:53.042: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xD3 timestamp 0x67C0
Feb 19 17:31:53.042: Pt:101 Evt:5 Pkt:0A 04 60 >>
Feb 19 17:31:53.050: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xD5 timestamp 0x67C0
Feb 19 17:31:53.050: << Pt:101 Evt:5 Pkt:0A 05 00
Feb 19 17:31:53.050: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xD5 timestamp 0x67C0
Feb 19 17:31:53.050: Pt:101 Evt:5 Pkt:0A 05 00 >>
Feb 19 17:31:53.062: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xD6 timestamp 0x67C0
Feb 19 17:31:53.062: << Pt:101 Evt:5 Pkt:0A 05 A0
Feb 19 17:31:53.062: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xD6 timestamp 0x67C0
Feb 19 17:31:53.062: Pt:101 Evt:5 Pkt:0A 05 A0 >>
Feb 19 17:31:53.070: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xD8 timestamp 0x67C0
Feb 19 17:31:53.070: << Pt:101 Evt:5 Pkt:0A 06 40
Feb 19 17:31:53.070: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xD8 timestamp 0x67C0
Feb 19 17:31:53.070: Pt:101 Evt:5 Pkt:0A 06 40 >>
Feb 19 17:31:53.082: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xD9 timestamp 0x67C0
Feb 19 17:31:53.082: << Pt:101 Evt:5 Pkt:0A 06 E0
Feb 19 17:31:53.082: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xD9 timestamp 0x67C0
Feb 19 17:31:53.082: Pt:101 Evt:5 Pkt:0A 06 E0 >>
Feb 19 17:31:53.090: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xDB timestamp 0x67C0
Feb 19 17:31:53.090: << Pt:101 Evt:5 Pkt:0A 07 80
Feb 19 17:31:53.090: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xDB timestamp 0x67C0
Feb 19 17:31:53.090: Pt:101 Evt:5 Pkt:0A 07 80 >>
Feb 19 17:31:53.102: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xDC timestamp 0x67C0
Feb 19 17:31:53.102: << Pt:101 Evt:5 Pkt:8A 08 20
Feb 19 17:31:53.102: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xDC timestamp 0x67C0
Feb 19 17:31:53.102: Pt:101 Evt:5 Pkt:8A 08 20 >>
Feb 19 17:31:53.102: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xDD timestamp 0x67C0
Feb 19 17:31:53.102: << Pt:101 Evt:5 Pkt:8A 08 20
Feb 19 17:31:53.102: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xDD timestamp 0x67C0
Feb 19 17:31:53.102: Pt:101 Evt:5 Pkt:8A 08 20 >>
Feb 19 17:31:53.118: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xDE timestamp 0x67C0
Feb 19 17:31:53.118: << Pt:101 Evt:5 Pkt:8A 08 20
Feb 19 17:31:53.118: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xDE timestamp 0x67C0
Feb 19 17:31:53.118: Pt:101 Evt:5 Pkt:8A 08 20 >>
Feb 19 17:31:53.270: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xE7 timestamp 0x7080
Feb 19 17:31:53.270: << Pt:101 Evt:2 Pkt:0A 00 00
Feb 19 17:31:53.270: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xE7 timestamp 0x7080
Feb 19 17:31:53.270: Pt:101 Evt:2 Pkt:0A 00 00 >>
Feb 19 17:31:53.282: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xE8 timestamp 0x7080
Feb 19 17:31:53.282: << Pt:101 Evt:2 Pkt:0A 00 A0
Feb 19 17:31:53.282: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xE8 timestamp 0x7080
Feb 19 17:31:53.282: Pt:101 Evt:2 Pkt:0A 00 A0 >>
Feb 19 17:31:53.290: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xEA timestamp 0x7080
Feb 19 17:31:53.290: << Pt:101 Evt:2 Pkt:0A 01 40
Feb 19 17:31:53.290: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xEA timestamp 0x7080
Feb 19 17:31:53.290: Pt:101 Evt:2 Pkt:0A 01 40 >>
Feb 19 17:31:53.302: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xEB timestamp 0x7080
Feb 19 17:31:53.302: << Pt:101 Evt:2 Pkt:0A 01 E0
Feb 19 17:31:53.302: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xEB timestamp 0x7080
Feb 19 17:31:53.302: Pt:101 Evt:2 Pkt:0A 01 E0 >>
Feb 19 17:31:53.310: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xED timestamp 0x7080
Feb 19 17:31:53.310: << Pt:101 Evt:2 Pkt:0A 02 80
Feb 19 17:31:53.310: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xED timestamp 0x7080
Feb 19 17:31:53.310: Pt:101 Evt:2 Pkt:0A 02 80 >>
Feb 19 17:31:53.322: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xEE timestamp 0x7080
Feb 19 17:31:53.322: << Pt:101 Evt:2 Pkt:0A 03 20
Feb 19 17:31:53.322: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xEE timestamp 0x7080
Feb 19 17:31:53.322: Pt:101 Evt:2 Pkt:0A 03 20 >>
Feb 19 17:31:53.330: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xF0 timestamp 0x7080
Feb 19 17:31:53.330: << Pt:101 Evt:2 Pkt:0A 03 C0
Feb 19 17:31:53.330: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xF0 timestamp 0x7080
Feb 19 17:31:53.330: Pt:101 Evt:2 Pkt:0A 03 C0 >>
Feb 19 17:31:53.342: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xF1 timestamp 0x7080
Feb 19 17:31:53.342: << Pt:101 Evt:2 Pkt:0A 04 60
Feb 19 17:31:53.342: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xF1 timestamp 0x7080
Feb 19 17:31:53.342: Pt:101 Evt:2 Pkt:0A 04 60 >>
Feb 19 17:31:53.350: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xF3 timestamp 0x7080
Feb 19 17:31:53.350: << Pt:101 Evt:2 Pkt:0A 05 00
Feb 19 17:31:53.350: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xF3 timestamp 0x7080
Feb 19 17:31:53.350: Pt:101 Evt:2 Pkt:0A 05 00 >>
Feb 19 17:31:53.362: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xF4 timestamp 0x7080
Feb 19 17:31:53.362: << Pt:101 Evt:2 Pkt:0A 05 A0
Feb 19 17:31:53.362: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xF4 timestamp 0x7080
Feb 19 17:31:53.362: Pt:101 Evt:2 Pkt:0A 05 A0 >>
Feb 19 17:31:53.370: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xF6 timestamp 0x7080
Feb 19 17:31:53.370: << Pt:101 Evt:2 Pkt:0A 06 40
Feb 19 17:31:53.370: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xF6 timestamp 0x7080
Feb 19 17:31:53.370: Pt:101 Evt:2 Pkt:0A 06 40 >>
Feb 19 17:31:53.382: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xF7 timestamp 0x7080
Feb 19 17:31:53.382: << Pt:101 Evt:2 Pkt:0A 06 E0
Feb 19 17:31:53.382: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xF7 timestamp 0x7080
Feb 19 17:31:53.382: Pt:101 Evt:2 Pkt:0A 06 E0 >>
Feb 19 17:31:53.390: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xF9 timestamp 0x7080
Feb 19 17:31:53.390: << Pt:101 Evt:2 Pkt:0A 07 80
Feb 19 17:31:53.390: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xF9 timestamp 0x7080
Feb 19 17:31:53.390: Pt:101 Evt:2 Pkt:0A 07 80 >>
Feb 19 17:31:53.402: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xFA timestamp 0x7080
Feb 19 17:31:53.402: << Pt:101 Evt:2 Pkt:0A 08 20
Feb 19 17:31:53.402: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xFA timestamp 0x7080
Feb 19 17:31:53.402: Pt:101 Evt:2 Pkt:0A 08 20 >>
Feb 19 17:31:53.410: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xFC timestamp 0x7080
Feb 19 17:31:53.410: << Pt:101 Evt:2 Pkt:0A 08 C0
Feb 19 17:31:53.410: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xFC timestamp 0x7080
Feb 19 17:31:53.410: Pt:101 Evt:2 Pkt:0A 08 C0 >>
Feb 19 17:31:53.422: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xFD timestamp 0x7080
Feb 19 17:31:53.422: << Pt:101 Evt:2 Pkt:8A 09 60
Feb 19 17:31:53.422: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xFD timestamp 0x7080
Feb 19 17:31:53.422: Pt:101 Evt:2 Pkt:8A 09 60 >>
Feb 19 17:31:53.422: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xFE timestamp 0x7080
Feb 19 17:31:53.422: << Pt:101 Evt:2 Pkt:8A 09 60
Feb 19 17:31:53.422: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xFE timestamp 0x7080
Feb 19 17:31:53.422: Pt:101 Evt:2 Pkt:8A 09 60 >>
Feb 19 17:31:53.438: s=VoIP d=DSP payload 0x65 ssrc 0x58F0F00A sequence 0xFF timestamp 0x7080
Feb 19 17:31:53.438: << Pt:101 Evt:2 Pkt:8A 09 60
Feb 19 17:31:53.438: s=DSP d=VoIP payload 0x65 ssrc 0x58F0F00A sequence 0xFF timestamp 0x7080
Feb 19 17:31:53.438: Pt:101 Evt:2 Pkt:8A 09 60 >> -
How do I move all clips from an event into the edit line...
If I have an event of tons of clips that comprise an hour of footage, how can I select all of those clips at once from the event library and transfer them to the edit line above?
There is no Select All feature, that I can see, and it takes forever to highlight each chunk of clips, one at a time, and drag them up.
I'd like to drag the whole 1 hour's worth of clips in one event in my library up to the edit line at once.
Thanks!Hi Rich,
I am trying to somehow "select all" clips, (not just all shots within a clip), but all 100 or so slips from an entire tape, from the first clip, to the last clip, and everything in between, and move them all up to the edit pane at once.
As it is now, I have to click on a thumbnail, select the entire clip, and move one clip at a time up to the edit pane.
I want to move all the clips up at once. -
IPhone will not send stored DTMF tones to toll free numbers?
Has anybody else noticed this? When I program my iPhone to connect
to a teleconferencing service (ATT's ironically), the DTMF tones are not
tansmitted. If I change the dial in number to a "caller paid" area code,
the tones are transmitted fine. Probably something to make it more difficult
for consumers to use cheaper international calling accounts.
For example a number such as 866 NNN NNNN , XXXXXXXX will not transmist the numbers after the pause, but change the are code to 865, and it does.I'm having the same issue. We use a calling card to make international calls and could really benefit from the ability to send 15 digits at a time from the address book. I do it on the RAZR, albeit ungracefully, though it's a smooth operation on my SE W810.
Any thoughts on how this might work (if it does) on the iPhone would be much appreciated.
Regards,
va -
How to copy a clip from one event to another?
Hi all... I need some help please.
I have several events with multiple clips in them.
I want to create a new event. I would like to COPY multiple different clips to this new event. I don't want to move clips from the events they are in now.
I want to do this because I want to put together a movie with several clips that I want to consolidte from several events. But I don't want to move the clips, I want to copy them into a new event. So yes, basically have duplicate clips, but in two different events.
Is this possible? Please tell me it is
Thanks.Thanks AppleMan... was hoping you would reply.
Here is my situation. I have an external hard drive with several years of events (2005-2011). I want to take several selected clips from those events to put on my internal hard drive. The clips I want to use for this video (a surprise video for bday too) I want to bring to a Newly created Single event on the hard drive. This way I have all clips I want to use to make this video in one location. I don't want to move them from the EHD.
This way I can work on the video without that EHD (on my MBP).
After I am done making the video, I'll likely delete this event with the duplicates. So it will take up hard drive space for only a few months. And it won't be tons of clips... just selected ones.
Thank you!!
p.s. Wait, if I move them from an EHD to a new single event on IHD, will it move or copy them? Maybe it does copy them rather than move them? I don't know.
The other issue is I have all clips from 2012 on my IHD. I would like to copy some of those clips to this new event also. Again, so all clips being used in the video montage are in a single location. This will be for ease of making video since I will have all clips being used in one location/event.
Thanks again AppleMan. -
I am trying to configure a particular SNMP community string on many ATAs for a Cisco 'Know the Network' project. On a Cisco ATA 186, you cannot telnet or SSH to it and the only way to remotely change the parameters are via the web GUI, which doesn't have the SNMP field available.
Is this possible without getting physical console access to the ATA?No.
An ASA can, as you noted, restrict source and destination IP and port. To do what you are asking, one would need to prevent a string within the payload from being transmitted (or only accept certain strings).
You should just put the access-list on the destination device(s) restricting what host(s) are allowed snmp rw (as you alluded to). That's a very common implementation straight out of the textbook.
Maybe you are looking for
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Anyway to transfer files between iPod & Mac without using iTunes or WiFi?
I need a way to transfer files to and from my iPod touch without having to sync to iTunes, and without using WiFi. Something like Bluetooth would be ideal, except it only seems to work with other iOS 5 devices for some reason I've seen various apps t
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Pop up window doesnt closing during continuous run
i have developed pop up window for my application... im getting the pop up window to condition in my application... im facing the problem in closing the the pop up window during continuous run, wen i click ok its not closing in continuous run... can
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How do we customize BSEG-XINVE for posting invoices on Polish Company Cod ?
When posting an invoice in Spain or France, there is a the flag BSEG-XINVE that you can tick. BSEG-INVE = Indicator that you are dealing with a vendor invoice for capital goods. I would like to use this field in other countries (Poland and Czech Repu
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How to fill the blanks after field content?
Hi, I have a problem about ABAP programming. As follow: data: txt char(20) type c. txt = 'text_vol '. Now I excute these codes in SAP abap workbench. The result is: txt = 'text_vol'. The blanks follow the 'tex_vol' is condense by system. How can I pe
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Are European quotation marks (not smart quotes) available? They look like: << and >>. They don't seem available among special characters, even Unicode.